=== release 1.20.1 ===

2022-03-14 11:33:33 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.20.1

2022-03-14 11:33:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	  Update ChangeLogs for 1.20.1

2022-02-22 10:13:28 +0100  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: fix rtp_source_get_nack_deadlines doc
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1950>

2022-01-14 23:42:27 -0600  Tim Mooney <Tim.Mooney@ndsu.edu>

	* sys/v4l2/ext/types-compat.h:
	  v4l2: include <sys/ioccom.h> on Illumos
	  Needed for _IOR/_IORW
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1947>

2022-03-03 13:47:05 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Fix race condition between qbuf and pool streamoff
	  There is a chance that pool->buffers[index] sets BUFFER_STATE_QUEUED, but
	  it has not been queued yet which makes pool->buffers[index] still NULL.
	  At this time, if pool_streamff release all buffers with BUFFER_STATE_QUEUED
	  state regardless of whether the buffer is NULL or not, it will cause segfault.
	  To fix this, also check buffer when streamoff release buffer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1948>

2022-03-11 10:32:42 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/deinterlace/tvtime/scalerbob.c:
	  deinterlace: scalerbob: Reduce latency to 0
	  We only need the current field, just like `linear`.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1944>

2022-03-12 17:13:48 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/yadif.c:
	  yadif: Fix CHECK macro for YUY2 format
	  Used to make comb artifacts for videotestsrc pattern=ball for YUY2
	  format only (not AYUV).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1943>

2022-03-10 18:43:45 +0900  Sangchul Lee <sc11.lee@samsung.com>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpvp8depay: Fix crash when making 'GstRTPPacketLost' custom event
	  This patch fixes a seg.fault in gst_structure_new() with warnings as below.
	  GLib-GObject-WARNING **:
	  ../gobject/gtype.c:4330: type id '0' is invalid
	  GLib-GObject-WARNING **:
	  can't peek value table for type '<invalid>' which is not currently referenced
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1920>

2022-03-09 12:17:11 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/gstsouploader.c:
	  soup: Load the runtime library, not the development library
	  libsoup-2.4.so / libsoup-3.0.so are symlinks installed by development
	  packages, they are not available at runtime.
	  Also eliminate G_MODULE_SUFFIX since it's not useful for us, and is
	  actually incorrect on macOS anyway.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1071
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1907>

2021-02-08 21:40:19 +0100  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxreceive.h:
	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	  rtprtx: don't access type-system per buffer
	  When doing only a single stream of audio/video this hardly matters,
	  but when doing many at the same time, the fact that you have to get
	  a hold of the glib global type-system lock every time you process a buffer,
	  means that there is a limit to how many streams you can process in
	  parallel.
	  Luckily the fix is very simple, by doing a cast rather than a full
	  type-check.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1890>

2020-02-10 14:37:30 +0100  Havard Graff <havard@pexip.com>

	* tests/check/elements/rtprtx.c:
	  rtprtx: signed/unsigned and style fixes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1888>

2022-03-04 15:36:20 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Prevent race between _set_method and latency query
	  It's possible that the method is being manipulated while downstream
	  queries our latency, leading to crashes.
	  Prevent that from happening.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1859>

2022-03-03 23:04:36 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/gstsouploader.c:
	  soup: Fix static build with MSVC
	  ../ext/soup/gstsouploader.c(818): error C4098: '_soup_session_send_async': 'void' function returning a value
	  It's technically a false warning, but that's how MSVC works, so fix
	  it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1846>

2022-03-03 00:37:57 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/meson.build:
	  soup: Fix pkgconfig generation and documentation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1846>

2022-03-02 23:22:39 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/meson.build:
	  soup: Fix static build when default_library=both
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1007
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1846>

2022-03-02 23:11:09 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/meson.build:
	  soup: Don't error out in static build unless option is enabled
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1846>

2022-02-27 15:45:01 +0100  Philippe Normand <philn@igalia.com>

	* ext/soup/gstsouploader.c:
	  soup: Lookup libsoup dylib files on Apple platforms
	  Fixes #1007
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1846>

2022-03-01 20:59:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Handle pixel-aspect-ratio caps field correctly when checking caps equality
	  Not having this field is equivalent with it being 1/1 so consider
	  it like that. The generic caps functions are not aware of these
	  semantics and would consider the caps different, causing a negotiation
	  failure when caps are changing from caps with to caps without or the
	  other way around.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1833>

2022-03-01 20:56:43 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Handle multiview-mode/flags caps fields correctly when checking caps equality
	  Not having these fields is equivalent with them being mono/0 so consider
	  them like that. The generic caps functions are not aware of these
	  semantics and would consider the caps different, causing a negotiation
	  failure when caps are changing from caps with to caps without or the
	  other way around.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1833>

2022-02-23 11:10:11 +0100  Sebastian Fricke <sebastian.fricke@collabora.com>

	* README.md:
	  Maintain build instructions at a single location
	  Do not maintain similar build instructions within each gst-plugins-*
	  subproject and the subproject/gstreamer subproject. Use the build
	  instructions from the mono-repository and link to them via hyperlink.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1829>

2022-03-01 00:12:56 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/x86/yadif.asm:
	  yadif.asm: Fix improper usage of LOAD macro
	  LOAD macro relies in m7 being zero for interleaving purposes. Using LOAD
	  on the m7 register makes it interleave with its new content instead of
	  with 0.
	  The effect of this bug was bobbing on some static lines that appeared
	  over fast-moving content.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1823>

2022-03-01 00:12:33 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/x86/yadif.asm:
	  yadif.asm: Typo fixes in comments
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1823>

2022-02-28 20:39:11 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/yadif.c:
	  yadif: Fix bug in C implementation of CHECK
	  It was different compared to the corresponding part in both ffmpeg and
	  the asm implementation. Fixing this makes videotestsrc pattern=spokes
	  not jump at all when not using the asm optimisations.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1823>

2021-12-06 12:49:18 -0500  Joseph Donofry <rubberduckie3554@gmail.com>

	* sys/osxaudio/gstosxaudiodeviceprovider.c:
	  osxaudiosrc: Support a device as both input and output
	  osxaudiodeviceprovider now probes devices more than once to determine
	  if the device can function as both an input AND and output device.
	  Previously, if the device provider detected that a device had any output
	  capabilities, it was treated solely as an Audio/Sink.  This causes issues
	  that have both input and output capabilities (for example, USB interfaces
	  for professional audio have both input and output channels).  Such devices
	  are now listed as both an Audio/Sink as well as an Audio/Source.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1809>

2022-02-24 20:28:23 +0530  Sanchayan Maity <sanchayan@asymptotic.io>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpldacpay.c:
	* gst/rtp/gstrtpldacpay.h:
	  rtp: ldac: Set frame count information in payload
	  The RTP payload seems to be required as it carries the frame count
	  information. Also, gst_rtp_base_payload_allocate_output_buffer had
	  the second argument incorrect.
	  Strangely some devices like Shanling MP4 and Sony XM3 would still
	  work without this while some like the Sony XM4 do not.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1804>

2022-02-25 12:44:26 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: greedyh: Stop adding 2 to cur_field_idx
	  Just a simplification.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1799>

2022-02-24 17:36:40 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: greedyh: Use _plane in _packed, fix planar formats
	  This greatly reduces code duplication. It also exposed the cause for
	  planar formats not being properly deinterlaced:
	  The planar path was missing the initial offset adjustment that the
	  packed path did to `L2` and `L2P` in the case of an even field, which
	  caused it to select the wrong weave lines every other field.
	  Add those offsets in `_plane`.
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1047
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1799>

2022-02-25 12:39:31 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: greedyh: Rename _planar_plane to _plane
	  As well as `i` to `plane`.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1799>

2022-02-25 12:36:17 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: greedyh: Move code from _planar into _planar_plane
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1799>

2022-02-25 12:30:21 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: greedyh: Move _planar_plane upwards
	  In preparation of refactoring. No functional change.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1799>

2022-02-18 15:23:13 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Emit a warning when no codec data found
	  It is bad if an mkv file does not have codec data for the ProRes
	  variant, so we should emit a warning. ffmpeg does the same thing.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1766>

2022-02-21 13:27:06 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtp/gstrtpulpfecenc.c:
	  ulpfecenc: slightly safer dispose impl
	  Technically dispose can be called more than once (even if gstelement is
	  not actually set up to do that) so need to protect against that.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1767>

2022-02-21 13:24:07 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtp/gstrtpulpfecenc.c:
	  ulpfecenc: fix unmatched free() call
	  One must always match a g_slice_new with a g_slice_free and a g_new with
	  a g_free.  This was not the case for the internal ctx struct.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1767>

2021-11-09 17:37:24 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtp/gstrtpulpfecenc.c:
	  rtpulpfecenc: add some debug logging
	  Like, what configuration we are using or whether a fec packet is
	  generated.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1767>

2022-02-02 12:49:29 +0100  Rouven Czerwinski <rouven@czerwinskis.de>

	* sys/v4l2/gstv4l2tuner.c:
	  gstv4l2tuner: return NULL if no norm set
	  If the video4linux device supports norms but has no norm set, norm is
	  returned as an uninitialized variable after the ioctl call, leading to
	  gst_v4l2_tuner_get_norm_by_std_id() returning a random norm from the
	  supported norms. Catch this case and instead return NULL to indicate
	  that no norm is setup.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1758>

2022-02-17 17:36:22 +0100  Sebastian Wick <sebastian.wick@redhat.com>

	* gst/matroska/matroska-demux.c:
	  matroska: default prores fourcc apcn
	  If there is no codec private data for prores it should default to Apple
	  ProRes 422 Standard Definition (apcn). Can be tested with
	  strobe_scientist.mkv from
	  https://developers.google.com/media/vp9/hdr-encoding
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1753>

2021-04-08 12:18:09 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't post an error message if pushing a sample failed with FLUSHING
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1732>

2022-02-11 23:55:57 +0100  Marek Vasut <marex@denx.de>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Pull row_stride from GST_VIDEO_FRAME_PLANE_STRIDE()
	  The libjpeg-turbo internal state might not be correctly initialized for
	  the first frame in a stream, pull the frame stride from gstreamer frame
	  metadata instead, which is correct even for the first frame, and which
	  makes this code consistent with the surrounding lines.
	  Fixes: e6d83d8f96 ("jpegdec: Support libjpeg-turbo colorspace conversion")
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1716>

2022-02-11 23:44:24 +0100  Marek Vasut <marex@denx.de>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Call gst_jpeg_turbo_parse_ext_fmt_convert() before jpeg_start_decompress()
	  It is imperative that the libjpeg-turbo state is properly initialized
	  before jpeg_start_decompress() is called. Make sure cinfo.out_color_space
	  and cinfo.raw_data_out are set to their final values matching their peer
	  caps before calling jpeg_start_decompress().
	  Fixes: e6d83d8f96 ("jpegdec: Support libjpeg-turbo colorspace conversion")
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1716>

2022-02-11 23:44:20 +0100  Marek Vasut <marex@denx.de>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Factor out gst_jpeg_turbo_parse_ext_fmt_convert()
	  Pull out peer caps checking code into gst_jpeg_turbo_parse_ext_fmt_convert().
	  This code is used by libjpeg-turbo extras to determine whether peer is capable
	  of handling buffers into which libjpeg-turbo can directly decode data. This
	  kind of check must be performed before jpeg_start_decompress() is called in
	  gst_jpeg_dec_prepare_decode() as well as in gst_jpeg_dec_negotiate(), hence
	  the common code.
	  This commit does modify the code a little to make it easier to call from both
	  call sites without much duplication, hence the extra `if (*clrspc)` test.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1716>

2022-02-11 23:29:27 +0100  Marek Vasut <marex@denx.de>

	* ext/jpeg/gstjpegdec.c:
	  Revert "jpegdec: only allow conversions from RGB"
	  This reverts commit 2aa2477208c029b0e1b8232d69f4f99a3bf1d473.
	  The commit is completely wrong, libjpeg-turbo is perfectly capable
	  of decoding I420 (YUV) to RGB. The test case provided alongside the
	  aforementioned commit passes without this revert because it decodes
	  image of JCS_YCrCb color space, so the new `if (clrspc == JCS_RGB)`
	  condition is false on that image, and the libjpeg-turbo decoding
	  does not get used. The real bug is hidden by that commit.
	  The real problem is in the call order of gst_jpeg_dec_prepare_decode()
	  and gst_jpeg_dec_negotiate(). The gst_jpeg_dec_prepare_decode() calls
	  jpeg_start_decompress() which sets up internal state of the libjpeg,
	  however, neither cinfo.out_color_space nor cinfo.raw_data_out are
	  set correctly yet. Those two are set up in gst_jpeg_dec_negotiate()
	  which is called a bit later. Therefore, the real fix is the set up
	  cinfo.out_color_space and cinfo.raw_data_out before calling
	  jpeg_start_decompress(). This is however a separate patch.
	  Fixes: 2aa2477208 ("jpegdec: only allow conversions from RGB")
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1716>

2022-02-11 21:35:54 +0100  Heiko Becker <heirecka@exherbo.org>

	* ext/lame/meson.build:
	  meson: Don't build lame plugin with -Dlame=disabled
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1720>

2022-02-01 14:28:24 +0100  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: Fix rotation not being applied when paused
	  The video wouldn't be redrawn immediately when a rotation was applied
	  but the pipeline was paused, as no new buffers were scheduled to be
	  displayed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1710>

2022-02-01 14:26:02 +0100  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	  gtk: Add a way to queue redrawing the base GTK widget
	  This will be used to request a redraw of the GTK widget should the
	  display be changed using properties not directly handled by the base GTK
	  widget, but by one of its descendants.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1710>

2022-02-07 16:17:28 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* meson.build:
	  Back to development

=== release 1.20.0 ===

2022-02-03 19:53:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* README:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.20.0

2022-02-03 19:53:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	  Update ChangeLogs for 1.20.0

2022-02-01 05:07:04 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* docs/meson.build:
	* sys/osxvideo/osxvideosink.m:
	  docs: Add objc and objcpp files to hotdoc gst_c_sources
	  Hotdoc should be able to extract and parse comments out of these. Just
	  need to be careful to only add the glob in directories that actually
	  contain *.m (objc) and *.mm (objcpp) files.
	  Also fix some doc comments and remove redundant ones.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1614>

2022-01-30 19:06:29 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxvideo/osxvideosink.m:
	  docs: Rename "OS X" to "macOS" in some documentation
	  No one uses the term "Mac OS X" anymore, it's "macOS". "OS X" is even
	  worse, because people will usually start the search with "mac".
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1607>

2022-01-30 03:56:14 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* docs/gst_plugins_cache.json:
	* sys/osxaudio/gstosxaudioelement.c:
	  osxaudio: Document GstOsxAudioElement interface
	  This is listed as a public interface implemented by osxaudio, so we
	  need to mark it as a plugin API so that it's listed in the
	  documentation correctly.
	  This is an ancient symbol, so add it to the symbol index too.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1601>

2022-01-28 23:15:28 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* docs/gst_plugins_cache.json:
	  applemedia: Document vtenc / vtdec elements
	  Also preserve-alpha property should only be exposed on the
	  vtenc_prores element since h264 does not support transparency.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/issues/94
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1601>

2022-01-29 12:56:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouputils.c:
	  soup: Don't store a strong reference to the logging object
	  Otherwise this causes a reference cycle between the session, the logger
	  and the logging object (i.e. the sink element or session wrapper).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1603>

2022-01-29 12:55:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Always abort the session once its last user is gone
	  And wait until there are no pending GSources on the main context anymore
	  afterwards.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1603>

2022-01-29 12:54:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Post context message after setting up the context from the source's thread
	  This simplifies the code and especially the locking a bit, and makes
	  sure we only export the session after it is fully set up.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1603>

=== release 1.19.90 ===

2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.19.90

2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	  Update ChangeLogs for 1.19.90

2022-01-28 15:47:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpsink: Protect against spurious wakeups during startup
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1594>

2022-01-28 15:33:04 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouputils.c:
	* ext/soup/gstsouputils.h:
	  souphttpsrc: Don't use the source element after setup from the session thread
	  The source element might be gone already if the session is shared with
	  other source elements.
	  As a consequence, do all logging via the session object instead of using
	  the source element.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1594>

2022-01-28 15:31:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't abort all pending operations on the session if shutting down a source with a shared session
	  Only do it for a non-shared session. Other sources would otherwise get
	  their requests cancelled unexpectedly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1594>

2022-01-28 15:31:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't set boolean to FALSE right after checking that it is FALSE
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1594>

2022-01-28 15:30:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: soup_session_new_with_options() can't fail with NULL
	  So don't check for it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1594>

2022-01-27 16:28:48 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/autodetect/gstautodetect.h:
	  autodetect: fix debug init category
	  Since the split of elements, the debug category
	  was default for autodetect.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1590>

2022-01-21 16:09:30 +0100  Daniel Kolesa <dkolesa@igalia.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	* ext/soup/gstsouploader.c:
	* ext/soup/gstsouploader.h:
	  soup: move libsoup session into its own thread
	  Starting with libsoup3, there is no attempt to handle thread safety
	  inside the library, and it was never considered fully safe before
	  either. Therefore, move all session handling into its own thread.
	  The libsoup thread has its own context and main loop. When some
	  request is made or a response needs to be read, an idle source
	  is created to issue that; the gstreamer thread issuing that waits
	  for that to be complete. There is a per-src condition variable to
	  deal with that.
	  Since the thread/loop needs to be longer-lived than the soup
	  session itself, a wrapper object is provided to contain them. The
	  soup session only has a single reference, owned by the wrapper
	  object.
	  It is no longer possible to force an external session, since this
	  does not seem to be used anywhere within gstreamer and would be
	  tricky to implement; this is because one would not have to provide
	  just a session, but also the complete thread arrangement made in
	  the same way as the system currently does internally, in order to
	  be safe.
	  Messages are still built gstreamer-side. It is safe to do so until
	  the message is sent on the session. Headers are also processed on
	  the gstreamer side, which should likewise be safe.
	  All requests as well as reads on the libsoup thread are issued
	  asynchronously. That allows libsoup to schedule things with as
	  little blocking as possible, and means that concurrent access
	  to the session is possible, when sharing the session.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/947
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1555>

2022-01-27 01:38:39 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/de.po:
	* po/fr.po:
	* po/ro.po:
	  gst-plugins-good: update translations
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1579>

2022-01-26 11:22:31 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix critical while serializing timeout element message
	  The "cause" field wasn't registered as a GEnumValue, so do that.
	  Fixes this critical in gst_structure_to_string():
	  `gst_value_serialize: assertion 'G_IS_VALUE (value)' failed`
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1573>

2022-01-24 10:45:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Warn when calculating the next fragment time in timecode mode fails
	  But only if timecode mode is enabled as it will fail all the time
	  otherwise.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1557>

2022-01-25 15:05:47 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tests/examples/qt/qmlsink-multisink/videoitem/videoitem.cpp:
	  qt: Retain compatibility with Qt 5.9
	  QSharedPointer.get() was added in Qt 5.11, and it does the same thing
	  as .data()
	  https://doc.qt.io/qt-5/qsharedpointer.html#get
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/867
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1565>

2022-01-21 14:13:12 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/types-compat.h:
	* sys/v4l2/ext/v4l2-common.h:
	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	  video4linux2: Sync kernel headers against 5.16.0
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/965>

2022-01-17 11:44:47 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videdec: Fix race condition between drain and state changes
	  This is due to an unsafe usage of the pad task. We didn't ensure proper
	  ownership of the task. That race involved the task being released too early,
	  and was detected, luckily, by the glib mutex implementationt that
	  reported the mutex being disposed while being locked.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1478>

2022-01-16 14:41:41 +0000  Philippe Normand <philn@igalia.com>

	* ext/soup/gstsoup.c:
	* ext/soup/gstsouploader.c:
	* ext/soup/gstsouploader.h:
	* ext/soup/gstsouputils.h:
	* ext/soup/meson.build:
	  soup: Reintroduce compile-time libsoup dependency for static builds
	  On Android (especially) and for static builds in general it is safer to link
	  against libsoup and have the dynamic custom loading disabled. For those cases we
	  can safely assume the application will use either libsoup2 or libsoup3 and not
	  both.
	  Fixes #939
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1536>

2022-01-16 14:40:11 +0000  Philippe Normand <philn@igalia.com>

	* ext/soup/gstsoupelement.c:
	  soup: Initialize debug category before use
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1536>

2022-01-16 14:39:42 +0000  Philippe Normand <philn@igalia.com>

	* ext/soup/gstsouploader.c:
	  soup: Fix return types for a couple VTable functions
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1536>

2022-01-14 19:53:29 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/cairo/meson.build:
	  meson: Build cairo subproject when unavailable on the system
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/952
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1523>

2022-01-20 17:59:30 +0100  Aleksandar Topic <aleks@qtec.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Fix example launch line format issue
	  The currently shown example launch line will not run, because it
	  cannot handle png images.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1547>

2021-05-07 18:22:06 +0200  Bastien Nocera <hadess@hadess.net>

	* docs/gst_plugins_cache.json:
	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkbasesink.h:
	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	  gtk: Add "video-aspect-ratio-override" property
	  Allow front-ends to override the pixel aspect ratio found inside the
	  video file itself, or most likely, missing from the video file.
	  This is a long-standing feature of totem.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1137>

2021-10-13 15:33:14 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Remove _update_par() forward declaration
	  No functional changes.
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1137/diffs#note_1102782
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1137>

2021-10-13 15:38:52 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Mark properties as changeable in the PLAYING state
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1137#note_1102789
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1137>

2021-05-11 11:28:57 +0200  Bastien Nocera <hadess@hadess.net>

	* docs/gst_plugins_cache.json:
	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Fix "pixel-aspect-ratio" property range
	  Fix the arguments passed to gst_param_spec_fraction in the incorrect
	  order.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1137>

2021-05-07 18:19:55 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Rename display pixel aspect ratio related constants
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1137>

2021-05-07 18:16:30 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Make "pixel-aspect-ratio" changes immediate
	  Schedule a resize when the display's pixel aspect ratio has changed, if
	  one isn't already scheduled.
	  Closes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/883
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1137>

2022-01-05 02:07:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* docs/meson.build:
	* meson.build:
	  meson: Add explicit check: kwarg to all run_command() calls
	  This is required since Meson 0.61.0, and causes a warning to be
	  emitted otherwise:
	  https://github.com/mesonbuild/meson/commit/2c079d855ed87488bdcc6c5c06f59abdb9b85b6c
	  https://github.com/mesonbuild/meson/issues/9300
	  This exposed a bunch of places where we had broken run_command()
	  calls, unnecessary run_command() calls, and places where check: true
	  should be used.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>

2021-12-30 16:31:33 +0000  Heinrich Kruger <heinrich.kruger@sohonet.com>

	* gst/rtp/gstrtphdrext-colorspace.c:
	  rtp-hdrext-colorspace: Fix color range encoding
	  The color space RTP header extension encodes color range as specified in
	  https://www.webmproject.org/docs/container/#Range. In other words:
	  0: Unspecified,
	  1: Broadcast Range,
	  2: Full range,
	  3: Defined by matrix coefficients and transfer characteristic.
	  This does not match the values of GstVideoColorRange, so it is not
	  correct to just write the colorimetry.range value to the header
	  extension.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1482>

2021-12-24 14:52:38 +0900  Jeongki Kim <jeongki.kim@jeongki.kim>

	* gst/rtp/gstrtpg726depay.c:
	  rtpg726depay: fix endian conversion
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1469>

2021-12-23 14:29:55 +0000  Corentin Damman <c.damman@intopix.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix typo in tests
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1466>

2021-12-20 09:28:50 +0000  Florian Zwoch <fzwoch@gmail.com>

	* ext/aalib/gstaatv.c:
	  aatv: Fixes for rain-mode
	  Some rain-mode properties tried to read float from int value.
	  Initialize rain after setting rain-mode. Rain was non-functional if
	  width/height were left at default values.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1459>

2021-12-07 23:48:39 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpreddec.h:
	  reddec: implement support for the BUNDLE case
	  When multiple streams are bundled together, there may be more
	  than one red payload type to handle.
	  In addition, as the red decoder works by filling in gaps in
	  the seqnums, there needs to be one rtp_history queue per sequence
	  domain.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>

2021-12-07 23:43:21 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: add new request-fec-decoder-full signal for BUNDLE
	  When multiple streams are bundled together, the application needs
	  to know about the payload type in order to instantiate the appropriate
	  FEC decoder.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>

2021-12-03 02:52:06 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtpredenc.c:
	* gst/rtp/gstrtpredenc.h:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/rtp/gstrtpulpfecenc.h:
	* tests/check/elements/rtpred.c:
	  rtp/redenc|ulpfecenc: add support for TWCC
	  In redenc, when input buffers have a header for the TWCC extension,
	  we now add one to our wrapper buffers.
	  In ulpfecenc we add one in that case to our protection buffers.
	  This makes TWCC functional when UlpRed is used in webrtcbin.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1414>

2021-12-02 13:32:33 +0000  Thibault Saunier <tsaunier@igalia.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Handle 'profile' field
	  In order to allow "level-asymmetry-allowed" we now handle a new
	  "profile" field, which as the same semantics as the "profile" field in
	  H.264 stream so that we can force payloaded stream to have the right
	  format when using the `gst_sdp_media_get_caps_from_media` to set caps
	  filter after the payloader. This allows a simple negotiation in standard
	  RTP negotiation based on SDPs (like webrtc) for that particular case,
	  closely respecting the specs.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1410>

2021-12-10 16:08:27 +0100  Jakub Adam <jakub.adam@collabora.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: avoid blocking wait for X events
	  XNextEvent() blocks indefinitely in absence of X11 events, which can
	  prevent the pipeline from stopping.
	  This can cause problems when ximagesrc is used in "remote desktop"
	  scenarios and the GStreamer application itself, through which the user
	  is viewing and controlling the machine, is the only source of input
	  events.
	  Replace the call with non-blocking XCheckTypedEvent().
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1438>

2021-12-10 17:36:30 +0100  Célestin Marot <marotcelestin@gmail.com>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: fix caps leak
	  since `gst_caps_replace()` and `gst_pad_set_caps()` both ref the caps and neither of them takes the ownership of the caps -> it must be unreffed in `gst_multi_file_src_set_property()`
	  to test the leak (on Unix): `echo coucou > /tmp/file.txt && GST_TRACERS=leaks GST_DEBUG="GST_TRACER:7" gst-launch-1.0 multifilesrc location=/tmp/file.txt caps='txt' ! fakesink`
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1436>

2021-12-06 15:37:06 +0100  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: only allow conversions from RGB
	  libjpeg-turbo only supports converting from RGB to other RGB formats.
	  Fix runtime error when trying to convert from a YUV format for example.
	  Fix #916
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1421>

2021-11-29 15:35:36 +0100  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: handle 0x0 sizes gracefully
	  Also ignore 0x0 sizes in the fallback case and assume the size can be
	  anything between 1x1 and MAXxMAX.
	  This fixes the case where a width=0, height=0 caps are created. Whith
	  this patch the caps will contain width=[1,MAX], height=[1,MAX].
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1396>

2021-11-23 20:54:57 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	* tests/check/elements/rtpfunnel.c:
	  rtpfunnel: fix extmap handling on accept-caps
	  Follow-up on 97d83056b315c56834eaa6776ae4c6a0848b5ef9, only check
	  for intersection with the current srccaps when checking if a sinkpad
	  can accept caps.
	  I must have been lucky in my firefox testing then, and always entered
	  the code path with audio getting negotiated first, thus not failing
	  the is_subset check when srccaps had been negotiated as
	  application/x-rtp, and an accept-caps query was made for the video
	  caps with a defined extmap.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1384>

2021-11-12 13:54:59 +0100  Jean Felder <jean.felder@gmail.com>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: Map GST_TAG_MUSICBRAINZ_RELEASETRACKID
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1342>

2021-11-12 13:51:56 +0100  Jean Felder <jean.felder@gmail.com>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: Map GST_TAG_MUSICBRAINZ_RELEASEGROUPID
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1342>

2021-11-18 16:27:17 +0000  Tobias Reineke <tobi@g3th.net>

	* ext/shout2/gstshout2.c:
	  shout2: Add compatibility for libshout >= 2.4.2
	  In libshout >=2.4.2 shout_open() can return SHOUTERR_RETRY in addition
	  to SHOUTERR_BUSY.
	  The nonblocking example in libshout fixes the problem in a similar
	  way, as mentioned by the author in this issue:
	  https://gitlab.xiph.org/xiph/icecast-libshout/-/issues/2316
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/848
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1363>

2021-11-15 14:39:14 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  qmlglsink: fix another resize case
	  If qt asks us to redraw before we have both set a buffer and caps we
	  would attempt to use the new caps with the old buffer which could result
	  in bad things happening.
	  Only update caps from new_caps once the buffer has actually been set.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1346>

2021-11-01 10:08:32 +0700  Trung Do <trung1.dothanh@toshiba.co.jp>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Update fmt if padded height is greater than fmt height
	  If padded height is greater, buffer bytesused could be larger than plane length,
	  and cause VIDIOC_QBUF failure.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1355>

2021-11-09 19:41:14 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	* tests/check/elements/rtpfunnel.c:
	  rtpfunnel: don't enforce twcc during upstream negotiation
	  A previous patch has caused rtpfunnel to output twcc-related
	  information downstream, however this leaked into upstream
	  negotiation (through funnel->srccaps), causing payloader to
	  negotiate twcc caps even when not prompted to do so by the user.
	  Fix this by only enforcing that upstream sends us application/x-rtp
	  caps as was the case originally.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>

2021-10-30 01:18:18 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/rtptwcc.c:
	  rtptwcc: complete bufferlist fix
	  When dealing with bufferlists, we need to store one "SentPacket"
	  structure per buffer, not one per buffer list!
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>

2021-11-12 17:59:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Log cslg_shift that was determined
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-12 13:00:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use a composition time offset of 0 for "no decode samples" for the time being
	  This needs codec-specific handling, but using 0 instead of G_MININT32 at
	  least gives somewhat reasonable behaviour.
	  See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/883
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-12 12:46:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Always check ctts for unreasonably large offsets
	  If this happens then ignore the whole ctts. Previously we only did this
	  if the PTS/DTS shift was determined from the ctts instead of the cslg.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-12 12:39:17 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux: Dump composition time offsets in trun as signed integers
	  Just like we do for ctts without regard of the version of the box.
	  Huge offsets are interpreted as negative offsets by qtdemux so this
	  works.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-12 11:36:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.h:
	  qtdemux: Add a comment why only positive cslg shifts are considered
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-02 18:38:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Only adjust segment.stop by cslg_shift if stop is not -1
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-02 18:29:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Handle negative composition offsets in the trun box the same way as for non-fragmented streams
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-02 17:43:17 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Parse ctts version
	  Negative composition time offsets are only allowed with version 1 of the
	  box, however we parse it as a signed value also for version 0 boxes as
	  unfortunately there are such files out there and it's unlikely to have
	  (valid) huge composition offsets.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-02 17:41:01 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux_dump.c:
	  qtdemux: Add support for version 1 cslg boxes
	  They use 64 bit fields instead of 32 bit.
	  Also parse offset as a signed integer (in both versions) and clamp it to
	  a positive value as negative values don't really interest us here.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-11-02 17:54:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't free cslg data that we don't own on corrupt files
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>

2021-05-07 10:44:15 +0200  Rafał Dzięgiel <rafostar.github@gmail.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-read-common.c:
	  matroska: Ref index table when updating track info
	  Track index table array was being lost during track info update.
	  Ref it over to updated info, so it can be used for finding
	  nearest seek points.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1203>

2021-11-12 11:28:23 +0100  Rafał Dzięgiel <rafostar.github@gmail.com>

	* gst/matroska/ebml-read.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-read-common.c:
	  matroska: Use g_array_unref everywhere
	  Instead of using g_array_free which is not thread safe use g_array_unref instead
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1203>

2021-11-11 19:11:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/cairo/gstcairooverlay.c:
	* gst/equalizer/gstiirequalizernbands.c:
	  docs: fix unnecessary ampersand, < and > escaping in code blocks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1340>

2021-11-12 03:17:44 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
	  st2022-1-fecdec: fix packet trimming
	  g_sequence_remove_range's end iter is exclusive, so if one
	  wants to remove that item as well, it should be called with
	  the next iter.
	  This could in theory fix an issue where:
	  * The sequence isn't entirely trimmed, with an old item lingering
	  * Following FEC packets are immediately discarded because they
	  arrived later than corresponding media packets, long enough for
	  seqnums to wrap around
	  * We now try to reconstruct a media packet with a completely obsolete
	  FEC packet, chaos ensues.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1341>

2021-11-11 15:49:19 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  qmlsink: support caps changes better
	  We need to hold onto the last buffer until the next buffer arrives.
	  Before, if a caps change comes we would remove the currently rendering
	  buffer.  if Qt asks use to render something, we would render the dummy
	  black texture.
	  Fixes a period of black output when upstream is e.g. changing resolution
	  as in hls adaptive bitrate scenarios.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1338>

2021-09-27 16:52:22 +0100  James Cowgill <james.cowgill@blaize.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: Record buffer states in pool to fix dequeue race
	  The `gst_v4l2_buffer_pool_dqbuf` function contains this ominous comment:
	  /* get our GstBuffer with that index from the pool, if the buffer was
	  * outstanding we have a serious problem.
	  */
	  outbuf = pool->buffers[group->buffer.index];
	  Unfortunately it is common for buffers in _output_ buffer pools to be
	  both queued and outstanding at the same time. This can happen if the
	  upstream element keeps a reference to the buffer, or in an encoder
	  element itself when it keeps a reference to the input buffer for each
	  frame.
	  Since the current code doesn't handle this case properly we can end up
	  with crashes in other elements such as:
	  (gst-launch-1.0:32559): CRITICAL **: 17:33:35.740: gst_video_frame_map_id: assertion 'GST_IS_BUFFER (buffer)' failed
	  and:
	  (gst-launch-1.0:231): GStreamer-CRITICAL **: 00:16:20.882: write map requested on non-writable buffer
	  Both these crashes are caused by a race condition related to releasing
	  the same buffer twice from two different threads. If a buffer is queued
	  and outstanding this situation is possible:
	  **Thread 1**
	  - Calls `gst_buffer_unref` decrementing the reference count to zero.
	  - The core GstBufferPool object marks the buffer non-outstanding.
	  - Calls the V4L2 release buffer function.
	  - If the buffer is _not_ queued:
	  - Release it back to the free pool (containing non-queued buffers).
	  **Thread 2**
	  - Dequeues the queued output buffer.
	  - Marks the buffer as not queued.
	  - If the buffer is _not_ outstanding:
	  - Calls the V4L2 release buffer function.
	  - Release it back to the free pool (containing non-queued buffers).
	  If both of these threads run at exactly the same time there is a small
	  window where the buffer is marked both not outstanding and not queued
	  but before it has been released. In this case the buffer will be freed
	  twice causing the above crashes.
	  Unfortunately the variable recording whether a buffer is outstanding is
	  part of the core `GstBuffer` object and is managed by `GstBufferPool` so
	  it's not as straightforward as adding a mutex. Instead we can fix this
	  by additionally recording the buffer state in `GstV4l2BufferPool`, and
	  handle "internal" and "external" buffer release separately so we can
	  detect when a buffer becomes not outstanding.
	  In the new solution:
	  - The "external" buffer pool release and the "dqbuf" functions
	  atomically update the buffer state and determine if a buffer is still
	  queued or outstanding.
	  - Subsequent code and a new
	  `gst_v4l2_buffer_pool_complete_release_buffer` function can proceed to
	  release (or not) a buffer knowing that it's not racing with another
	  thread.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1010>

2021-11-09 15:10:06 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: separate out the two fec decoder locations
	  The pipeline flow for receiving looks like this:
	  rtpsession ! rtpssrcdemux ! session_fec_decoder ! rtpjitterbuffer ! \
	  rtpptdemux ! stream_fec_decoder ! ...
	  There are two places where a fec decoder could be placed.
	  1. As requested from the 'request-fec-decoder' signal: after rtpptdemux
	  for each ssrc/pt produced
	  2. after rtpssrcdemux but before rtpjitterbuffer: added for the
	  rtpst2022-1-fecenc/dec elements,
	  However, there was some cross-contamination of the elements involved and
	  the request-fec-decoder signal was also being used to request the fec
	  decoder for the session_fec_decoder which would then be cached and
	  re-used for subsequent fec decoder requests.  This would cause the same
	  element to be attempted to be linked to multiple elements in different
	  places in the pipeline.  This would fail and cause all kinds of havoc
	  usually resulting in a not-linked error being returned upstream and an
	  error message being posted by the source.
	  Fix by not using the request-fec-decoder signal for requesting the
	  session_fec_decoder and instead solely rely on the added properties for
	  that case.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1300>

2021-11-09 21:20:19 +0800  Zhao, Gang <gang.zhao.42@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix can not demux Opus track made by qtmux
	  Opus stream info is read from dOps box [1]. The offset of dOps box in Opus box is different in mp4a version 1 and 0 [2]. Calculate the offset of dOps box according to mp4a version.
	  [1] https://opus-codec.org/docs/opus_in_isobmff.html
	  [2] subprojects/gst-plugins-good/gst/isomp4/atoms.c:sample_entry_mp4a_copy_data:2146
	  Fixed: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/918
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1329>

2021-11-09 13:36:28 +0800  Haihua Hu <jared.hu@nxp.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: set video alignment of video meta
	  need apply video alignment info on video meta, downstream
	  element can get buffer alignment from video meta
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1326>

2021-11-06 16:48:20 +0800  Zhao, Gang <gang.zhao.42@gmail.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: Skip check of key frame when open GOP
	  Valid stream [1] might has no key frame.
	  Fixed: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/890
	  [1] https://storage.googleapis.com/downloads.webmproject.org/test_data/libvpx/vp90-2-16-intra-only.webm
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1316>

2021-10-05 18:07:06 +0100  Joe Todd <joextodd@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudio: Increase max channels to 64
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1059>

2021-11-05 15:17:20 +0530  Sanchayan Maity <sanchayan@asymptotic.io>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Only set caps if they do not match current caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1311>

2021-11-04 16:04:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Support bools for setting extra-controls
	  They're just mapped to 0 / 1 and can already be set as integers, but
	  being able to set them as booleans seems useful.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1308>

2021-11-03 18:44:03 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* meson.build:
	  Back to development

=== release 1.19.3 ===

2021-11-03 15:43:36 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.19.3

2021-11-03 15:43:32 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	  Update ChangeLogs for 1.19.3

2021-11-02 16:46:08 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: Make minimum distance between keyframes in next-file=key-frame mode configurable
	  Previously this was hardcoded to 10s, which is not necessarily the
	  desired behaviour.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1292>

2021-11-01 16:32:10 +0100  Erlend Eriksen <erlend_ne@hotmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix deadlock in gst_qt_mux_prepare_moov_recovery
	  Regression from 5766731bd4200c3a374522a749386f740347661a
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1288>

2021-10-30 16:22:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	  video: Fix order of new video formats
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1282>

2021-10-30 00:58:55 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtphdrext-colorspace.c:
	  Couple more g_memdup() -> g_memdup2() fixes
	  Fixes deprecation warnings with newer GLib versions.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1279>

2021-10-23 00:44:57 +0200  Piotrek Brzeziński <piotr@centricular.com>

	* docs/gst_plugins_cache.json:
	  video-format: Add support for ARGB64 LE/BE and similar variants
	  Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1247>

2021-10-29 03:46:25 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/rtptwcc.c:
	  rtptwcc: don't assume all PacketInfo->data are buffers
	  They can also be buffer lists
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1268>

2021-10-25 11:37:45 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: require matching GStreamer dep versions for unstable development releases
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>

2021-10-28 21:37:47 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix text and closed-caption handling.
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182
	  broke text and and closed caption extraction when introducing WebVTT
	  support, by making the output buffers not have timestamps any more.
	  Fix that by making the process functions copy buffer metadata
	  when generating new output buffers.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1262>

2021-10-25 01:02:07 +0100  Tim-Philipp Müller <tim@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/ky.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  gst-plugins-good: update translations
	  Fixes #656
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1240>

2021-10-23 14:38:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add pasp box to the list of known boxes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1235>

2021-01-20 11:53:51 +0100  Rafał Dzięgiel <rafostar.github@gmail.com>

	* gst/matroska/matroska-read-common.c:
	  matroska: Set image/attachment structure mimetype
	  Set structure mimetype to fix data detection by mimetype in other plugins.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1205>

2021-01-20 11:33:39 +0100  Rafał Dzięgiel <rafostar.github@gmail.com>

	* gst/matroska/matroska-read-common.c:
	  matroska: Treat non-image structure as attachment
	  Otherwise each structure is named as GstTagImageInfo even if
	  it does not contain any images which is misleading.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1205>

2021-10-21 20:32:05 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix leak of pad when a fec encoder and aux sender a created
	  The ghost sink pad retrieved by rtpbin from the aux sender was not freed
	  when there was a previous element (fec encoder) in the chain.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1222>

2021-10-18 15:47:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/meson.build:
	  meson: update for meson.build_root() and .build_source() deprecation
	  -> use meson.project_build_root() or .global_build_root() instead.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>

2021-10-18 00:40:14 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/meson.build:
	* meson.build:
	* tests/check/meson.build:
	  meson: update for dep.get_pkgconfig_variable() deprecation
	  ... in favour of dep.get_variable('foo', ..) which in some
	  cases allows for further cleanups in future since we can
	  extract variables from pkg-config dependencies as well as
	  internal dependencies using this mechanism.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>

2021-10-18 00:03:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: clean up conditional paths after version bump
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>

2021-09-15 14:19:06 -0400  Olivier Crête <olivier.crete@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtphdrext-clientaudiolevel.c:
	* gst/rtpmanager/gstrtphdrext-clientaudiolevel.h:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/meson.build:
	* tests/check/elements/rtphdrextclientaudiolevel.c:
	* tests/check/meson.build:
	  rtphdrext-clientaudiolevel: Rename RFC 6464 element
	  Multiplying elements named after RFC numbers is confusing,
	  so let's give them meaningful names.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1125>

2021-06-05 03:13:52 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/meson.build:
	* gst/isomp4/qtdemux-webvtt.c:
	* gst/isomp4/qtdemux-webvtt.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add support for wvtt (WebVTT) subtitles.
	  WebVTT in ISO MP4 is specified in ISO 14496-30,
	  and needed for DASH support. It's stored in an
	  mp4 specific format. To handle it compatibly,
	  the wvtt boxes are converted back into WebVTT text
	  and pushed as application/x-subtitle-vtt
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>

2021-08-27 22:45:18 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: Split buffer process functions.
	  Split the different handling for closed captions, VobSub subpicture
	  and timed text samples into separate simplified process functions.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>

2021-08-27 21:58:25 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  isomp4: Use a function pointer for buffer splitting.
	  Swap the `need_process` boolean check on qtdemux streams
	  for a direct function pointer to the splitting function,
	  so we can stop adding extra cases to the single growing
	  `gst_qtdemux_process_buffer()` function.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>

2021-10-14 18:38:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: bump meson requirement to >= 0.59
	  For monorepo build and ugly/bad, for advanced feature
	  option API like get_option('xyz').required(..) which
	  we use in combination with the 'gpl' option.
	  For rest of modules for consistency (people will likely
	  use newer features based on the top-level requirement).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>

2021-10-05 12:28:22 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Allow more fields changes renegotiating h264
	  And consider interlace-mode=progress as equivalent to the field not
	  being specified.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1062>

2021-10-05 10:50:32 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Ignore some fields when renegotiating
	  Those values are never used in the muxer so we should not fail the
	  negotiation if those are changing.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1062>

2021-10-17 01:59:35 -0300  Thibault Saunier <tsaunier@igalia.com>

	* ext/lame/meson.build:
	  lame:meson: Avoid using fallback when we could use system install
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1176>

2021-10-05 11:38:33 -0300  Thibault Saunier <tsaunier@igalia.com>

	* ext/lame/gstlamemp3enc.h:
	* ext/lame/meson.build:
	  meson: Support building lame as subproject
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1068>

2021-10-12 15:52:48 -0300  Thibault Saunier <tsaunier@igalia.com>

	* docs/meson.build:
	  meson: Streamline the way we detect when to build documentation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>

2021-10-13 14:42:15 +0100  Philippe Normand <philn@igalia.com>

	* ext/soup/stub/soup.h:
	  soup-stub: Gate G_URI_FLAGS_SCHEME_NORMALIZE behind glib version check
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1138>

2021-10-13 10:38:07 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gtkgstglwidget.c:
	  gtkglsink: Avoid assertion when applying "auto" rotation method
	  Guard against the orientation not coming from an inexistant tag, nor
	  from the application (rotation set to "auto") which caused an assertion.
	  When the application requests the auto rotation method, make sure it is
	  resolved to a rotation that's applicable.
	  ERROR:gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstglwidget.c:745:gtk_gst_gl_widget_set_rotate_method: code should not be reached
	  Fixes: 103ceb853a5f7dade07a1ac4aa517e9df6ed5ded
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1134>

2021-10-04 09:31:02 +0100  Philippe Normand <philn@igalia.com>

	* ext/soup/gstsoup.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpclientsink.h:
	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	* ext/soup/gstsouploader.c:
	* ext/soup/gstsouploader.h:
	* ext/soup/gstsouputils.c:
	* ext/soup/gstsouputils.h:
	* ext/soup/meson.build:
	* ext/soup/stub/soup.h:
	* meson.build:
	* tests/check/elements/souphttpsrc.c:
	* tests/check/meson.build:
	  soup: Runtime compatibility support for libsoup2 and libsoup3
	  The src and sink elements no longer link against libsoup. It is now loaded at
	  runtime. If any version is resident already, it is used. Otherwise we first try
	  to load libsoup3 and if it's not found we fallback to libsoup2.
	  For the unit-tests, we now build one version of the test unit file per libsoup
	  version found. So if both libsoup2 and libsoup3 are available on the host, the
	  CI will cover them both.
	  Based on initial patch by Daniel Kolesa <dkolesa@igalia.com> and
	  Patrick Griffis <pgriffis@igalia.com>.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1044>

2021-09-10 18:03:55 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtopuspay: Set marker bit inside RTP packet too
	  At the end of a talk spurt, not only set the marker flag on the
	  GstBuffer, but also set the bit inside the RTP header as recommended
	  by the RFC.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1124>

2021-10-11 14:27:24 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gstgtkbasesink.c:
	  gtksink: Fetch the default "widget" value in the docs
	  There's really no interesting "widget" value that could be shown in the
	  docs, so use the GST_PARAM_DOC_SHOW_DEFAULT flag to avoid showing
	  another value.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1088>

2021-10-11 10:22:41 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkglsink.c:
	  gtksink: Avoid errors fetching widget property
	  Avoid errors when fetching the "widget" property and GTK initialisation
	  fails, such as when running in a non-graphical environment.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1088>

2021-05-04 13:27:30 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkglsink.c:
	  gtksink: Return reference to GtkWidget in the acquire function
	  This should ensure thread safety.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1088>

2020-09-03 18:27:19 -0400  Olivier Crête <olivier.crete@collabora.com>

	* docs/gst_plugins_cache.json:
	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkbasesink.h:
	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstglwidget.h:
	  gtkglsink: Add rotate-method property
	  This mostly just takes code out of glimagesink and applies it here.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1088>

2021-10-12 19:01:37 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't assert on the input side if no GOP is available when shutting down
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1119>

2021-10-12 11:43:16 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/meson.build:
	  libsoup: Bump to 2.74 to fix gssapi build failure on macOS
	  See: https://gitlab.gnome.org/GNOME/libsoup/-/merge_requests/255
	  We also need to disable sysprof by default because it only builds on
	  native Linux. If someone really wants it, they can enable the option
	  on the command-line by passing -Dlibsoup:sysprof=enabled
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1114>

2020-06-05 16:20:18 +0200  Marek Vasut <marex@denx.de>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: Support libjpeg-turbo colorspace conversion
	  The libjpeg-turbo has a built-in support for performing colorspace
	  conversion. The performance of this conversion is much better than
	  doing the same separately using videoconvert. Implement support for
	  this conversion to RGBx/xRGB/BGRx/xBGR formats. Other formats can
	  be easily added later.
	  - The decoding of various pixel formats can be tested and compared to
	  non-libjpeg-turbo decoding as follows:
	  for gfmt in {RGB,BGR}{,x} x{RGB,BGR} ; do
	  echo "$gfmt"
	  gst-launch-1.0 -q \
	  videotestsrc pattern=colors ! \
	  video/x-raw,format=${gfmt} ! \
	  fakesink dump=true | \
	  head -n 200 | tail -n 1
	  gst-launch-1.0 -q --gst-plugin-path=build/ext/jpeg/ \
	  videotestsrc pattern=colors ! \
	  video/x-raw,format=${gfmt} ! \
	  jpegenc ! \
	  jpegdec ! \
	  video/x-raw,format=${gfmt} ! \
	  fakesink dump=true | \
	  head -n 200 | tail -n 1
	  done
	  Result looks as follows, i.e. comparable:
	  RGB
	  00000c70 (0x7f7736fbdd10): 05 33 19 05 33 26 05 33 33 05 33 40 05 33 4c 05  .3..3&.33.3@.3L.
	  00000c70 (0x7f389e8f7d10): 05 32 17 04 32 22 04 32 31 04 32 3e 04 32 4a 04  .2..2".21.2>.2J.
	  RGBx
	  00000c70 (0x7f79efd0ad10): cc 07 22 ff d9 07 22 ff e6 07 22 ff f3 07 22 ff  .."..."..."...".
	  00000c70 (0x7fb6989f3d10): cd 06 22 00 d9 06 22 00 e6 06 22 00 f4 06 22 00  .."..."..."...".
	  BGR
	  00000c70 (0x7fa0a6c42d10): 05 0c 33 05 19 33 05 26 33 05 33 33 05 40 33 05  ..3..3.&3.33.@3.
	  00000c70 (0x7fc74165fd10): 05 08 32 04 17 32 04 22 32 04 31 32 04 3e 32 04  ..2..2."2.12.>2.
	  BGRx
	  00000c70 (0x7fbf399f1d10): 22 07 cc ff 22 07 d9 ff 22 07 e6 ff 22 07 f3 ff  "..."..."..."...
	  00000c70 (0x7f50e3d1cd10): 22 06 cd 00 22 06 d9 00 22 06 e6 00 22 06 f4 00  "..."..."..."...
	  xRGB
	  00000c70 (0x7f0b950a2d10): ff cc 07 22 ff d9 07 22 ff e6 07 22 ff f3 07 22  ..."..."..."..."
	  00000c70 (0x7f4416b8dd10): 00 cd 06 22 00 d9 06 22 00 e6 06 22 00 f4 06 22  ..."..."..."..."
	  xBGR
	  00000c70 (0x7f237d74dd10): ff 22 07 cc ff 22 07 d9 ff 22 07 e6 ff 22 07 f3  ."..."..."..."..
	  00000c70 (0x7f095547dd10): 00 22 06 cd 00 22 06 d9 00 22 06 e6 00 22 06 f4  ."..."..."..."..
	  ^^          ^^          ^^          ^^
	  Notice how the alpha channel is set to arbitrary value in case of the
	  libjpeg-turbo decoding into RGBx/BGRx/xRGB/xBGR pixel formats. This is
	  documented in libjpeg-turbo README.md as follows:
	  "
	  When using the RGBX, BGRX, XBGR, and XRGB colorspaces during decompression, the
	  X byte is undefined, and in order to ensure the best performance, libjpeg-turbo
	  can set that byte to whatever value it wishes.
	  "
	  - The interlaced num_fields=2 mjpeg stream can be generated and
	  tested as follows (this does require mjpegtools):
	  $ gst-launch-1.0 videotestsrc num-buffers=10 ! jpegenc ! multifilesink location=in%04d.jpg
	  $ jpeg2yuv -f 25 -I t -L 0 -j in%04d.jpg | yuv2lav -f avi -o result.avi
	  ...
	  $ gst-launch-1.0 --gst-plugin-path=build/ext/jpeg/ filesrc location=result.avi ! \
	  avidemux ! jpegdec ! video/x-raw,format=RGBx ! videoconvert ! autovideosink
	  Signed-off-by: Marek Vasut <marex@denx.de>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1091>

2021-09-21 13:37:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Keep track of the pending input GOPs in a queue
	  This cleans up input GOP handling and makes it possible to handle more
	  than 2 pending GOPs, which could happen before if keyframes are arriving
	  with too short of a distance between them.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>

2021-09-16 19:36:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Handle frame reordering due to B frames better
	  Instead of assuming that the PTS of a keyframe is the lowest PTS of a
	  GOP, wait until the DTS has passed this PTS and take the minimum PTS up
	  to that point. That way the minimum PTS of a GOP can be determined, at
	  least for closed GOP streams. Open GOP streams still can't be handled
	  properly.
	  By knowing the minimum PTS of each GOP, keyframes can be requested at
	  the correct time relative to the GOP (and thus fragment) start and
	  fragment overflow calculations can calculate the correct durations of
	  the GOPs.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>

2021-09-16 14:07:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Reset timecodes when resetting the sink
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>

2021-09-16 13:11:20 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Use GST_CLOCK_STIME_NONE instead of 0 to initialize max out running time
	  Otherwise streams with only DTS would misbehave while it is negative.
	  For outputting everything immediately at EOS, use G_MAXINT64 instead
	  which is bigger-or-equal to any other running time.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>

2021-09-16 13:02:29 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Only update max in/out running time if it's actually bigger
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>

2021-09-16 12:53:12 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Only count keyframes for the reference context, consistently
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>

2021-09-24 13:38:39 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	  rtphdrext: Pass just the attributes to the subclass
	  Since the base class now does the parsing, there is no need
	  to reproduce that code in all the subclasses, just pass the attributes
	  which are the only relevant bit anyway.
	  Also, only store the direction if the subclass accepted the caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>

2021-09-24 12:52:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	  rtphdrext-rfc6464: Use helper function to set caps field
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>

2021-09-23 16:01:40 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtphdrext-colorspace.c:
	* gst/rtpmanager/gstrtphdrext-twcc.c:
	  rtphdrext: Set caps without attributes as the default
	  Most subclasses just use the simple function, so just let the base class
	  do it. It makes less code in subclasses.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>

2021-09-23 15:36:00 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtphdrext-colorspace.c:
	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	* gst/rtpmanager/gstrtphdrext-twcc.c:
	  rtphdrext: Put simple caps generation as the base class default
	  Instead of having a helper function that gets called by almost every
	  subclass, just let the base class set the caps fields automatically.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>

2021-09-28 10:11:15 +1000  Brad Hards <bradh@frogmouth.net>

	* README:
	* RELEASE:
	  doc: update IRC links to OFTC
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/945>

2021-09-26 01:07:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* meson.build:
	  Back to development
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>

=== release 1.19.2 ===

2021-09-23 01:33:39 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.19.2

2021-09-22 14:03:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph263pdepay.c:
	  rtph263pdepay: flag keyframes on output buffers
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1091>

2021-08-18 19:47:40 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtphdrext-twcc.c:
	  rtphdrhext-twcc: Return failure on map failure
	  This feels like exactly like a case that should fail.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>

2021-08-18 19:46:25 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtphdrext-colorspace.c:
	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	* gst/rtpmanager/gstrtphdrext-twcc.c:
	  rtphdrext: Update write() API to return a signed value
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>

2021-09-14 17:26:27 +0900  Seungha Yang <seungha@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Try to build AAC codec-data whenever it's possible
	  AAC codec_data is a just collection of AAC profile, samplerate, and
	  channels. We can know samplerate and channels from parsed
	  SampleEntry data. Although the AAC profile is unknown there,
	  let's assume it as AAC-LC like we've been doing for the version 1
	  atom.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1082>

2021-09-07 22:23:01 +0200  Vivienne Watermeier <vwatermeier@igalia.com>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flv: fix seqnum handling for seeks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1078>

2021-01-18 16:06:27 +1100  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: also allow muxing different h264/5 profiles/levels/etc
	  All of that is advertised through the codec_data itself so can change
	  just fine within isomp4.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1071>

2021-09-11 09:24:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroska: Add support for muxing/demuxing ffv1
	  Previously only demuxing when stored via the RIFF/AVI mapping was
	  supported.
	  See https://github.com/FFmpeg/FFV1/blob/master/ffv1.md#matroska-file-format
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/923
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1080>

2021-09-12 12:18:32 +0100  Philippe Normand <philn@igalia.com>

	* docs/gst_plugins_cache.json:
	  docs: Update cache
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1081>

2021-08-03 19:12:11 +0900  Seungha Yang <seungha@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Fix crash when interlaced field height is not DCT block size aligned
	  In case of interlaced JPEG file, we are doubling stride.
	  The scratch scan line should take account of it as well.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1042>

2021-09-02 08:38:54 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Also detect 0x000001 as H264 byte-stream start code in codec_data
	  This works around some AVI files storing byte-stream data in the
	  codec_data. The previous workaround was only checking for
	  0x00000001 (4 bytes) instead of 0x000001 (3 bytes).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1072>

2021-08-31 11:05:16 +0100  Philippe Normand <philn@igalia.com>

	* ext/qt/qtitem.cc:
	  qt: Fix build for Qt 5.9
	  The QQuickItem::size() method was introduced in 5.10, so use direct width() and
	  height() access instead.
	  Fixes #908
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1069>

2021-08-19 10:32:27 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtphdrextrfc6464.c:
	  rtphdrext-rfc6464: Add test for inserting in payloader using the API
	  This makes it clearer how to use the plugin in an API driven application.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>

2021-08-18 19:36:07 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	  rtphdrext-rfc6464: Put max level if the audio is beyond it
	  Otherwise, it just fails to add the extension, which makes no
	  sense. And our level element produces levels higher than 127 in some
	  cases.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>

2021-08-18 19:35:36 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	  rtphdrext-rfc6464: Add example pipeline
	  This makes it a bit easier to understand how to use it in an application.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>

2021-08-18 19:07:18 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtphdrextrfc6464.c:
	  rtphdrext-rfc6464: Add test for inserting it based on caps
	  Tests adding the extension based on the caps.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>

2021-08-27 14:32:45 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Force stream-start push when re-using EOS'd streams
	  When re-using streams, we *do* need to push a `stream-start` event downstream if
	  we previously were EOS'd. Failure to do that would never remove the EOS status
	  on all downstream elements and cause weird issues.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1067>

2021-08-27 02:05:45 -0400  Brad Smith <brad@comstyle.com>

	* gst/deinterlace/meson.build:
	  deinterlace: Use proper ASM output format for *BSD OS
	  FreeBSD/NetBSD/OpenBSD amd64 use the ELF binary format.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1066>

2021-08-27 13:30:57 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  qmlgl: don't critical on input events before input format has been set
	  Accessing the unset GstVideoInfo would result in criticals
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1065>

2021-08-25 11:53:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  docs: Add `Since` marker to "twcc-feedback-interval" property
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-08-25 10:33:24 +0200  Havard Graff <havard@pexip.com>

	* docs/gst_plugins_cache.json:
	  docs: update with "twcc-feedback-interval"
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-04-13 16:19:22 +0200  Tulio Beloqui <tulio.beloqui@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.h:
	* gst/rtpmanager/rtptwcc.c:
	* tests/check/elements/rtpsession.c:
	  rtptwcc: changes to use rtp buffer arrival time and current time.
	  For TWCC we are more interested to track the arrival time (receive side)
	  and the current time (sender side) of the buffers rather than the
	  running time.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-03-26 11:57:42 +0100  Knut Inge Hvidsten <knut.hvidsten@pexip.com>

	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtptwcc.c:
	* gst/rtpmanager/rtptwcc.h:
	  rtptwcc: add payloadtype to RTPTWCCPacket
	  The consumer of the stats can then separate between different media-types,
	  and do individual stats for each of them.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-03-19 18:19:43 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtptwcc.c:
	  rtptwcc: make enabling TWCC sticky
	  Meaning that if a caps comes along that does NOT have TWCC in it,
	  this does not turn of TWCC for the rest, as this is in fact
	  completely allowed. (To have some payload-types not containing TWCC
	  seqnums).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-02-23 09:44:05 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtptwcc.c:
	* gst/rtpmanager/rtptwcc.h:
	  rtptwcc: move TWCC-logic over to the TWCC-manager
	  Prevent cluttering up the rtpsession, and keeping things localized.
	  Also write TWCC-seqnums for *all* streams in the session if configured by
	  caps.
	  A while back WebRTC was not doing TWCC for audio, basically breaking the
	  whole idea of a "transport-wide seqnuencenumber" applying for all bundled
	  streams. However, they have since fixed this, and now it no longers
	  makes sense to be able to single out certain payloadtypes for
	  use with TWCC, rather just including them all.
	  This also makes using RTX, RED, FEC etc much simpler, as it will apply
	  to them all as they enter the rtpsession.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-02-23 09:50:04 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtptwcc.c:
	  rtptwcc: fix warning
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-02-11 15:17:16 +0100  Tulio Beloqui <tulio.beloqui@pexip.com>

	* gst/rtpmanager/rtptwcc.c:
	* tests/check/elements/rtpsession.c:
	  rtptwcc: fixes and optimizations around run-length chunks
	  Co-authored-by: Havard Graff <havard.graff@gmail.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2020-12-18 14:01:23 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtptwcc.c:
	* tests/check/elements/rtpsession.c:
	  rtptwcc: fix seqnum-wrap
	  Using the proper API to do this is obviously an improvement, and
	  adding a test for the case of a packet-loss when the seqnum wrap
	  is also a good idea.
	  Co-authored-by: Tulio Beloqui <tulio.beloqui@pexip.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2020-12-18 13:06:35 +0100  Tulio Beloqui <tulio.beloqui@pexip.com>

	* gst/rtpmanager/rtptwcc.c:
	* tests/check/elements/rtpsession.c:
	  rtptwcc: fixed feedback packet count overflow that allowed late packets to be processed
	  Co-authored-by: Havard Graff <havard.graff@gmail.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2020-12-16 16:31:18 +0100  Tulio Beloqui <tulio.beloqui@pexip.com>

	* gst/rtpmanager/rtptwcc.c:
	* tests/check/elements/rtpsession.c:
	  rtptwcc: fixed parsing of old sequence number
	  Co-authored-by: Havard Graff <havard.graff@gmail.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2020-12-16 16:16:09 +0100  Tulio Beloqui <tulio.beloqui@pexip.com>

	* gst/rtpmanager/rtptwcc.c:
	* tests/check/elements/rtpsession.c:
	  rtptwcc: fixed guint8 overflow of feedback packet count
	  Co-authored-by: Havard Graff <havard.graff@gmail.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2020-11-19 23:50:23 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtptwcc.c:
	* gst/rtpmanager/rtptwcc.h:
	* tests/check/elements/rtpsession.c:
	  rtptwcc: add feedback-interval
	  To allow RTCP TWCC reports to be scheduled on a timer instead of per
	  marker-bit.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-08-20 11:54:01 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/rtptwcc.c:
	* gst/rtpmanager/rtptwcc.h:
	  rtptwcc: remove _set_send_packet_ts
	  Not in use.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2020-11-17 00:45:02 +0100  Havard Graff <havard@pexip.com>

	* tests/check/elements/rtpsession.c:
	  rtptwcc: make twcc-tests more deterministic
	  They were a bit racy.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>

2021-08-24 13:28:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add depth for ProRes 4:4:4:4 variants if available
	  Might be 24bpp in case an alpha channel is coded but
	  the image is always opaque.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1061>

2021-08-22 23:16:26 +0000  Ruslan Khamidullin <ruslank@borisfx.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: for Apple ProRes, allow overriding pixel bit depth for 4:4:4:4 variants
	  e.g. when exporting an opaque image, yet with alpha channel.
	  Apple ProRes certification requires that, when a ProRes-writing
	  application *knows* that the entire frame is opaque, the application
	  writes only RGB without alpha even when the clip is RGBA. For that,
	  this tiny change allows the app to override pixel depth when writing ProRes.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1061>

2019-05-22 11:16:56 +0200  Havard Graff <havard.graff@gmail.com>

	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxdec.h:
	  vpxdec: Fix direct rendering, avoid holding write access
	  When a buffer is pushed downstream, we should try not to hold the
	  buffer mapped with write access. Doing so would often lead to
	  an unneccesary memcpy later.
	  For instance, gst_buffer_make_writable() in
	  gst_video_decoder_finish_frame() will cause a memcpy because of
	  _memory_get_exclusive_reference().
	  We know that we can perform a two-step remap when using system
	  memory, as this will not cause the location of the memory to
	  change.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/812>

2021-08-19 16:26:17 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4/mux: add a function for seeking to a specific output byte position
	  We do it enough times that this makes sense.  Also add a debug log line
	  for the seek position requested.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>

2021-08-19 16:02:47 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/atoms.c:
	  isomp4/mux: don't overwrite with a bigger moov when fragmenting
	  When outputting fragmented mp4, with a seekable downstream, we rewrite
	  the moov to maybe add a duration to the mvex.  If we start by not
	  writing the initial moov->mvex->mhed duration and then overwrite with a
	  moov containing mhed atom, the moov's will have different sizes and
	  could overwrite subsequent data and result in an unplayable file.
	  e.g. The initial moov would be of size 842 and the final moov would have
	  a size of 862.
	  Fix by always pushing out the mhed duration in the moov when
	  fragmenting.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/898
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>

2021-01-15 20:53:27 +1100  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: actually make streamable fallback work
	  We weren't setting the fragment_mode field anymore now that the
	  implementation doesn't change based on the value of the streamable
	  property.  This lead to invalid files.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>

2021-01-15 20:54:56 +1100  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: fix trun data offset handling
	  The trun offset was missing a calculation for one of the box type
	  headers.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/866
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>

2020-10-15 00:28:36 +1100  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/gstqtmux.c:
	  isomp4/mux: fixes for fragmented mp4 output
	  Various buffer offset calculations were not quite correct in all cases.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/866
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>

2021-08-10 00:53:57 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/matroska/matroska-mux.c:
	  matroska-mux: support H264 avc3 / H265 hev1
	  The matroska codec specs is unfortunately vague on the subject,
	  stating for H264:
	  AVC/H.264 stored as described in [@!ISO.14496-15]
	  and for H265:
	  HEVC/H.265 stored as described in [@!ISO.14496-15]
	  This spec however specifies multiple stream formats, our
	  implementation has opted for interpreting this as avc1 / hvc1,
	  both of which disallow in-band SPS.
	  Most decoders however will support in-band SPS / PPS, and
	  this commit gives the option to explicitly mux in avc3 / hev1,
	  which allows changing stream parameters on the fly, that is
	  useful for smart encoding for example.
	  When either of these stream formats are picked as the input,
	  changes in codec_data / tier / level / profile do not cause
	  renegotiation failure, a warning is logged however as it isn't
	  clear how compliant such a stream is.
	  The stream-format field is correctly ordered in the template
	  caps to avoid selecting potentially non-compliant options on
	  automatic negotiation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>

2021-08-10 00:51:36 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4/qtmux: allow renegotiating when tier / level / profile change
	  Those are carried either in codec_data or in-band SPS (for avc3),
	  and it is OK for those to change, though decoders obviously need
	  to support it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>

2021-08-06 23:36:48 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  isomp4/qtmux: accept video/x-h264, stream-format=avc3
	  The main difference between avc1 and avc3 is that avc3 is allowed
	  to contain in-band SPS / PPS. In practice decoders will always use
	  in-band parameter sets anyway, but it is cleaner to explicitly
	  advertise it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>

2021-08-06 22:59:23 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4/qtmux: make sure to switch to next chunk on new caps
	  For example, with single video sink pad, and new codec_data is
	  received, current_chunk_offset must be reset to -1 for the
	  aggregate loop to open a new chunk.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>

2021-08-06 22:55:32 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/atoms.c:
	  isomp4/atoms: fix multiple stsd entries
	  stsd entries are serialized in reverse order (starting from
	  g_list_last()), and must be prepended to the entry list for their
	  index to be correct when referenced from stsc entries.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>

2021-08-12 11:03:58 -0400  Arun Raghavan <arun@asymptotic.io>

	* docs/gst_plugins_cache.json:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroska-mux: Add a timestamp-offset property
	  Adds a user-controllable timestamp offset to clusters and blocks. This
	  should be useful if we want to have timestamps that have significance
	  outside of the current file (for example, we might set the offset to the
	  wallclock when the file is being created, or some other common base, if
	  we want to correlate streams across multiple files).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1051>

2021-07-15 12:02:40 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* gst/matroska/matroska-demux.c:
	  matroska: demux: update stream_start_time
	  The stream_start_time can be less than the first detected.
	  In case of B-Frame based media, the first frame PTS might be
	  greater than the next one.
	  Need to keep the segment.start if a seek has been performed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1030>

2021-08-17 16:08:33 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/matroska/matroska-demux.c:
	  mastrokademux: Remove redundant assignment
	  The segment.position is unconditionnaly set few lines below.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1030>

2021-08-17 16:49:47 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst/videocrop/gstvideocrop.c:
	* tests/icles/videocrop-test.c:
	  videocrop: Fix icles tests.
	  Internally videcrop can call gst_video_crop_set_info() with NULL as in
	  caps. Then critical messages are raised when the in caps are
	  processed.
	  To fix this the in caps are checked, and if they are present, its
	  capsfeature is extracted, otherwise, the previous raw caps detection
	  remains as before.
	  Also the videocrop-test removes the format field in the structure
	  because now its always passed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1056>

2021-01-08 17:34:02 +0100  Jakub Adam <jakub.adam@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpelements.h:
	* gst/rtp/gstrtphdrext-colorspace.c:
	* gst/rtp/gstrtphdrext-colorspace.h:
	* gst/rtp/meson.build:
	* tests/check/elements/rtphdrext-colorspace.c:
	* tests/check/meson.build:
	  rtp: Color Space header extension
	  Implements WebRTC header extension defined in
	  http://www.webrtc.org/experiments/rtp-hdrext/color-space.
	  It uses RTP header to communicate color space information and optionally
	  also metadata that is needed in order to properly render a high dynamic
	  range (HDR) video stream.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/853>

2021-08-09 10:46:30 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add protection when set decoder capture fps accroding to output fps
	  Some v4l2 drivers don't have the capacity to change framerate. There is
	  chance to make decoder capture fps to be 0/0 if numerator and denominator
	  returned by G_PARM ioctl are both 0. It causes critical warning
	  "passed '0' as denominator for `GstFraction'".
	  In order to fix this, add protection when set decoder capture fps according
	  to output fps.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1048>

2021-06-01 15:33:01 +0200  Per Förlin <perfn@axis.com>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add support to ignore x-server HEADER reply
	  When connecting to an RTSP server in tunnled mode (HTTP) the server
	  usually replies with a x-server header. This contains the address
	  of the intended streaming server. However some servers return an
	  "invalid" address. Here follows two examples when it might happen.
	  1. A server use Apache combined with a separate RTSP process to handle
	  Https request on port 443. In this case Apache handle TLS and
	  connects to the local RTSP server, which results in a local
	  address 127.0.0.1 or ::1 in the x-server reply. This address is
	  returned to the actual RTSP client in the x-server header.
	  The client will receive this address and try to  connect to it
	  and fail.
	  2. The client use a ipv6 link local address with a specified scope id
	  fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via Http on port 80.
	  The RTSP server receives the connection and returns the address
	  in the x-server header. The client will receive this address and
	  try to connect to it "as is" without the scope id and fail.
	  In the case of streaming data from RTSP servers like 1. and 2. it's
	  useful to have the option to simply ignore the x-server header reply
	  and continue using the original address.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1007>

2021-08-04 12:33:06 +0700  Dmitry Shusharin <pmdvsh@gmail.com>

	* ext/qt/qtitem.cc:
	  gstqmlgl: fix indent
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>

2021-07-30 16:52:23 +0700  Dmitry Shusharin <pmdvsh@gmail.com>

	* ext/qt/gstqsgtexture.cc:
	* ext/qt/gstqsgtexture.h:
	  gstqmlgl: wrap raw GstGLContext into GWeakRef
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>

2021-07-30 16:32:13 +0700  Dmitry Shusharin <pmdvsh@gmail.com>

	* tests/examples/qt/meson.build:
	* tests/examples/qt/qmlsink-multisink/main.cpp:
	* tests/examples/qt/qmlsink-multisink/main.qml:
	* tests/examples/qt/qmlsink-multisink/meson.build:
	* tests/examples/qt/qmlsink-multisink/qmlsink-multi.qrc:
	* tests/examples/qt/qmlsink-multisink/videoitem/VideoItem.qml:
	* tests/examples/qt/qmlsink-multisink/videoitem/videoitem.cpp:
	* tests/examples/qt/qmlsink-multisink/videoitem/videoitem.h:
	  gstqmlgl: add multisink test application
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>

2021-07-30 17:21:46 +0700  Dmitry Shusharin <pmdvsh@gmail.com>

	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  gstqmlgl: refactoring: rename ambiguous variables, clean up unused and duplicated ones
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>

2021-07-30 17:20:59 +0700  Dmitry Shusharin <pmdvsh@gmail.com>

	* ext/qt/gstqtglutility.cc:
	  gstqmlgl: rework WGL-specific context init code
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>

2021-07-30 17:20:49 +0700  Dmitry Shusharin <pmdvsh@gmail.com>

	* ext/qt/gstqtglutility.cc:
	  gstqmlgl: retrieve correct device bound to current GL context (+ minor code cleanup)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>

2021-07-30 17:20:25 +0700  Dmitry Shusharin <pmdvsh@gmail.com>

	* ext/qt/gstqsgtexture.cc:
	  gstqmlgl: correct validation for Qt GL context
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>

2021-07-30 17:20:07 +0700  Dmitry Shusharin <pmdvsh@gmail.com>

	* ext/qt/gstqtglutility.h:
	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	* ext/qt/qtwindow.cc:
	* ext/qt/qtwindow.h:
	  gstqmlgl: create helper QRunnable-based class for render jobs
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>

2021-08-06 16:25:02 +0200  Tulio Beloqui <tulio@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fixed stall on gap when using rtx
	  Co-authored-by: Håvard Graff <havard@pexip.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1055>

2021-08-13 19:32:53 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/flv/gstindex.c:
	  flv: use g_memdup2() as g_memdup() is deprecated
	  g_memdup() is deprecated since GLib 2.68 and we want to avoid
	  deprecation warnings with recent versions of GLib.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1052>

2021-08-15 12:26:38 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Always use the content decoder but set `Accept-Encoding: identity` if no compression should be used
	  Some servers respond with gzip-encoded responses regardless of whether
	  the request allowed it to be used in the response. By always having the
	  content decoder enabled, these invalid responses can be decoded
	  correctly while for well-behaving servers the `compress` property
	  selects between allowing compressed responses or not.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/833
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1053>

2021-08-12 22:57:01 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qt: always update the sink_retrieved flag when the sink retrieves
	  Fixes a case where adding a qmlgloverlay element after an existing
	  qmlglsink elements was already in the pipeline would create an entirely
	  separate GstGLDisplay pointing to the same underlying display resource.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1050>

2021-08-11 14:52:52 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstaspectratiocrop.h:
	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	* gst/videocrop/gstvideocropelement.c:
	* gst/videocrop/gstvideocropelements.h:
	* gst/videocrop/gstvideocropplugin.c:
	* gst/videocrop/meson.build:
	  videocrop: Resurrect logging category.
	  Fix for a regression from commit 8f1384c9. That commit moved the debug
	  category definition, as static, into a gstvideocropelement.c, but that
	  category was used as default, in gstvideocrop.c, so it was never used
	  at logging, so the debug selector never showed the logs for
	  videocrop.
	  This patch move back the category definition into gstvideocrop.c and
	  leaving the function videocrop_element_init() as a noop.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1049>

2021-07-31 23:14:34 +0900  Seungha Yang <seungha@centricular.com>

	* ext/jpeg/meson.build:
	  jpeg: Add support for meson fallback
	  Allow building jpeg plugin by using meson fallback
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1041>

2021-07-27 10:43:21 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Keep decoder capture fps same as output fps if it's not set
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1035>

2021-07-27 18:33:18 +0900  Seungha Yang <seungha@centricular.com>

	* docs/gst_plugins_cache.json:
	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudioclient.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: Add port-names property to select ports explicitly
	  By this new property, user can select physical port to connect,
	  and element will pick requested port instead of random ones.
	  User should provide full port name including "client_name:" prefix.
	  An example is
	  jackaudiosrc port-names="system:capture_1,system:capture_3" ! ...
	  jackaudiosink port-names="system:playback_2"
	  In addition to "port-names" property, a new connect type "explicit"
	  is added so that element can post error message if requested
	  "port-names" contains invalid port(s).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1037>

2021-07-23 11:04:00 +0200  Kai Uwe Broulik <kai_uwe.broulik@mbition.io>

	* ext/qt/gstqsgtexture.cc:
	* ext/qt/gstqtsink.cc:
	  qt: Support RGB format
	  In GstQSGTexture::hasAlphaChannel return value based on
	  whether the video format has alpha channel.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1040>

2021-07-26 20:14:32 +0900  Seungha Yang <seungha@centricular.com>

	* docs/gst_plugins_cache.json:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: Add low-latency property for automatic latency-optimized setting
	  Similar to wasapi/wasapi2 plugins on Windows, adding low-latency
	  option so that jack element can optimize GstAudioRingBufferSpec
	  setting for low latency.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1034>

2021-07-26 19:55:25 +0900  Seungha Yang <seungha@centricular.com>

	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudioclient.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackringbuffer.h:
	  jack: Remove trailing whitespace
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1034>

2021-07-27 17:58:15 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* docs/gst_plugins_cache.json:
	* gst/videocrop/gstvideocrop-private.h:
	  videocrop: Resurrect any caps feature negotiation.
	  Commit e31cbce4 brought a regression to negotiate featured caps. But
	  only by removing the entry in the caps template. This commit brings it
	  back.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1039>

2021-07-26 18:43:04 +0900  Seungha Yang <seungha@centricular.com>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: Fix assertion fail when device supports only mono channel
	  MAX should be larger than MIN for GST_TYPE_INT_RANGE.
	  GStreamer-CRITICAL **: 18:26:27.912:
	  gst_value_collect_int_range: assertion 'collect_values[0].v_int < collect_values[1].v_int' failed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1033>

2021-07-21 20:14:46 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtsrc.cc:
	* ext/qt/qtwindow.cc:
	* ext/qt/qtwindow.h:
	  qmlglsrc: fix operation without any qmlglsink
	  E.g. a pipeline like qmlglsrc ! gldownload ! ... would currently fail to
	  run because the OpenGL context are not created in the correct order.
	  The QtWindow also needs to know the OpenGL context used by downstream
	  elements in order to set optimize for the correct GstGLSyncMeta for
	  synchonisation purposes.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1036>

2021-07-26 17:55:24 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix some reference leaks in error cases.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1023>

2021-07-08 00:12:52 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* tests/check/elements/splitmuxsink.c:
	  splitmuxsink: Prevent hang going back to NULL after failures
	  Prevent a condition where splitmuxsink won't go back to NULL state
	  after a child element fails to change state by making sure that
	  a READY->READY state change doesn't fail, and by returning
	  GST_FLOW_ERROR or GST_FLOW_FLUSHING upstream to shut down streaming
	  as quickly as possible.
	  This can happen after (for example) setting an invalid filename
	  on the sink element. In that case, the READY->PAUSED transition
	  fails, but with internal elements still in the NULL state. Trying
	  to set splitmuxsink back to NULL then ends up trying to bring
	  those NULL elements up to READY with a READY->READY transition,
	  (which fails, prevent splitmuxsink from getting to NULL)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1023>

2021-07-13 01:27:45 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: reduce noise when gst_pad_set_caps fails
	  It may be that downstream is simply flushing, in which case logging
	  an error is misleading.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1029>

2021-07-08 02:22:20 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: always use factory property when set
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1024>

2021-06-21 13:47:50 +0200  Yacine Bandou <yacine.bandou@softathome.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: No need for new "application/x-cbcs" caps
	  Instead of using the new "application/x-cbcs" caps, we are just adding
	  a new structure field "ciphe-mode", to indicate which encryption scheme
	  is used: "cenc", "cbcs", "cbc1" or "cens".
	  Similarly for the protection metadata, we add the "cipher-mode" field
	  to specify the encryption mode with which the buffers are encrypted.
	  "cenc": AES-CTR (no pattern)
	  "cbc1": AES-CBC (no pattern)
	  "cens": AES-CTR (pattern specified)
	  "cbcs": AES-CBC (pattern specified, using a constant IV)
	  Currently only "cenc" and "cbcs" are supported.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1013>

2021-07-05 16:12:57 +0100  Philippe Normand <philn@igalia.com>

	* ext/qt/qtitem.h:
	  qt: Fix clang build
	  The updatePaintNode method is part of the QQuickItem class interface, so needs
	  to be flagged as overriding the default implementation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/567>

2020-04-15 10:38:04 +0100  Philippe Normand <philn@igalia.com>

	* ext/qt/gstqtsink.cc:
	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  qt: Add navigation events support
	  Currently handles only mouse events.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/567>

2020-04-15 10:33:22 +0100  Philippe Normand <philn@igalia.com>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Scroll events dispatch support
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/567>

2021-07-02 01:41:05 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Always write a tags element into seekhead
	  If there are only stream tags, we still want to write the
	  tags entry into the seekhead, so that tags can be found
	  quickly in the player.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/905
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1020>

2021-06-30 23:52:26 +0900  Seungha Yang <seungha@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't need to update track per GstCaps if it's not changed
	  Skip GstQTMuxPad::set_caps() call for duplicated caps.
	  All the processing done in set_caps() method for duplicated caps
	  are redundant.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1019>

2021-07-01 13:18:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Remove pads and reset the element also in READY->NULL
	  Mostly for completeness.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1018>

2021-07-01 13:18:09 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: Remove pads also in PAUSED->READY
	  They're based on per-stream information and that should be reset
	  whenever going to READY state.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1018>

2021-02-16 16:39:34 +0100  Jakub Adam <jakub.adam@collabora.com>

	* docs/gst_plugins_cache.json:
	  docs: update plugins cache for vp9enc
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>

2021-04-09 19:22:29 +0200  Jakub Adam <jakub.adam@collabora.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	* ext/vpx/gstvpxenums.h:
	* ext/vpx/meson.build:
	  vpx: add enum for adaptive quantization modes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>

2021-02-16 13:28:00 +0100  Jakub Adam <jakub.adam@collabora.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	  vp9enc: expose frame-parallel-decoding property
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>

2021-02-16 12:57:55 +0100  Jakub Adam <jakub.adam@collabora.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	  vp9enc: expose aq-mode property
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>

2021-06-26 20:00:03 +0900  Seungha Yang <seungha@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Fix broken SO_SNDBUF get/set on Windows
	  SO_SNDBUF has been undefined on Windows because of missing WinSock2.h
	  include. And don't use native socket functions (e.g., setsockopt())
	  if code is expected to be built on Windows. We don't link ws2_32.lib
	  for this plugin.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1016>

2021-06-24 14:57:14 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtphdrext-twcc.c:
	  rtpmanager: Access GstRTPHdrExt fields through accessor
	  This way, the implementation can be private.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1017>

2021-06-22 17:19:19 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Refuse seeks in BYTES format
	  If downstream tries to seek in BYTES format, don't pass that through
	  to upstream. The byte positions downstream requests won't make any
	  sense in the muxed stream. There might be other formats we want to
	  pass through to upstream, but BYTES is not one of them. If we get a
	  seeking query about BYTES format, refuse that too.
	  This fixes a situation where we're playing a fragmented mp4 over http
	  and qtdemux refuses the initial seek (in TIME format), but then
	  h264parse/baseparse send a seek in BYTES format and everything falls
	  apart.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1014>

2021-06-16 16:30:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: update codec_data in caps regardless of format
	  Updating of codec_data in the caps is important to propagate changes
	  in sps/pps/vps via NALs. Without this, downstream does not renegotiate
	  when upstream changes resolution.
	  The comment referring to rtph264pay is from 2015 and is out of date.
	  rtph264pay stopped doing that in 2017 with commit
	  dabeed52a995d27e16eba9e4617e61eb0bcd44c4
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1011>

2021-06-04 13:56:05 +0300  Jordan Petridis <jordan@centricular.com>

	* docs/gst_plugins_cache.json:
	  doc: update gst_plugins_cache.json
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1006>

2021-06-03 20:33:45 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  qtitem: don't potentially leak a large number of buffers
	  The only other place where these queued buffers are removed, is in
	  setCaps() but that is not called at all on shutdown so this list of
	  buffers could not be removed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1004>

2021-05-28 09:54:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* docs/gst_plugins_cache.json:
	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Remove arbitrary encoding size limitation
	  The encoder is happy to encode with sizes less then 16x16, so remove this
	  arbitrary limitation. This also fixes the fact the sink and src template caps
	  disagree.
	  Fixes #888
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/998>

2021-05-23 15:42:38 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: use g_memdup2() as g_memdup() is deprecated
	  - atom nodes/bytereader sizes are already checked
	  - palettes: are fixed/known size
	  g_memdup() is deprecated since GLib 2.68 and we want to avoid
	  deprecation warnings with recent versions of GLib.
	  Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>

2021-05-23 01:28:11 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/ebml-read.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-read-common.c:
	  matroskademux: use g_memdup2() as g_memdup() is deprecated
	  - ebml-read: add some sanity checks when going from 64-bit
	  to 32-bit length
	  - matroska-ids: codec_data_size has been checked via
	  gst_ebml_read_binary(), is existing allocation.
	  - matroska-demux: alloc size is from existing allocations
	  g_memdup() is deprecated since GLib 2.68 and we want to avoid
	  deprecation warnings with recent versions of GLib.
	  Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>

2021-05-22 19:39:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/libpng/gstpngdec.c:
	* ext/vpx/gstvpxenc.c:
	* ext/wavpack/gstwavpackenc.c:
	* meson.build:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtph264.c:
	* tests/check/elements/rtph265.c:
	* tests/check/elements/rtpopus.c:
	* tests/check/elements/rtpvp8.c:
	* tests/check/elements/rtpvp9.c:
	* tests/icles/gdkpixbufoverlay-test.c:
	  Use g_memdup2() where available and add fallback for older GLib versions
	  - png: alloc size variable is a png type that's always 32-bit
	  - vpx: alloc size based on existing allocation
	  - wavpack: alloc size based on existing allocation
	  - icles: gdkpixbufoverlay: trusted and hard-coded input data
	  - rtp tests: rtp-payloading, vp8, vp9, h264, h265: trusted and/or static input data
	  g_memdup() is deprecated since GLib 2.68 and we want to avoid
	  deprecation warnings with recent versions of GLib.
	  Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>

2021-06-01 15:28:36 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* meson.build:
	  Back to development

=== release 1.19.1 ===

2021-06-01 00:11:44 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* README:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.19.1

2021-05-29 12:54:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: fix image corruption when compiled with MSVC on Windows
	  On Windows with MSVC, jpeg_header_size would end up 2 bytes larger
	  than it should be. This then leads to the first 2 bytes of the
	  actual jpeg image data to be dropped, because we think those
	  belong to the header, which results in an undecodable image when
	  reconstructed in the depayloader.
	  What happens is that when the compiler evaluates
	  jpeg_header_size = mem.offset + read_u16_and_inc_offset_by_2(&mem);
	  it actually uses the mem.offset value after it has been increased
	  by the function call on the right hand size of the equation.
	  From section 6.5 of the C99 spec:
	  3. The grouping of operators and operands is indicated by the syntax [74].
	  Except as specified later (for the function-call (), &&, ||, ?:, and
	  comma operators), the order of evaluation of subexpressions and the
	  order in which side effects take place are both unspecified.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/889
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/999>

2021-05-25 16:19:20 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Set default latency if the frame duration is invalid
	  If the duration of the v4l2object is invalid, use default 25fps instead.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/996>

2021-05-26 00:23:56 +0900  Seungha Yang <seungha@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Drop "field-order" field while transforming caps
	  Like other basetransform subclasses are doing, drop field
	  which can be converted by deinterlace.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>

2021-05-25 20:10:34 +0900  Seungha Yang <seungha@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Drop field-order field if outputting progressive
	  Progressive with field-order doesn't make sense
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>

2021-05-21 14:19:29 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* tests/check/elements/rtpssrcdemux.c:
	  rtpssrcdemux: fix "data flow before segment event" crash
	  This crash could happen at any time a RTP and RTCP buffer arrived
	  simultaneously in ssrcdemux.
	  The problem was that sticky-event arriving while the rtp and rtcp pads
	  were being set up could arrive just too late to be included in the initial
	  forwarding.
	  The fix checks if the stickies have been sent on the srcpad about to be
	  pushed on, and if not sends them. It also blocks any stickes from
	  being forwarded *prior* to this happening, to avoid them arriving on
	  the srcpad multiple times.
	  Since the test loops 1000 times, this will make running under valgrind
	  take forever, so use the RUNNING_ON_VALGRIND variable to detect we
	  are running under valgrind, and reduce the loop-count to 2 in that case.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>

2021-05-21 18:45:17 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: refactor destruction of GstRtpSsrcDemuxPads
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>

2021-05-21 18:30:28 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	  rtpssrcdemux: make naming consistent
	  Use plural for GstRtpSsrcDemuxPads, since it contains two pads, and
	  use the variable-name 'dpads' everywhere.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>

2021-05-23 15:14:11 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: use g_strndup() for copying text data
	  So we don't rely on NUL terminators inside the data.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>

2021-05-23 13:29:07 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: clean up adtl/note/labl chunk parsing
	  We were passing the size of the adtl chunk to the note/labl
	  sub-chunk parsing function, which means we may memdup lots of
	  data after the chunk string's NUL terminator that doesn't
	  really belong to it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>

2021-05-23 13:24:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: guard against overflow when comparing chunk sizes
	  Could be rewritten as lsize > (size - 8) a well, but the
	  extra check seems clearer. Doesn't look like it was problematic,
	  lsize wasn't actually used when parsing the sub-chunks.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>

2021-05-21 13:31:12 -0300  Daniel Almeida <daniel.almeida@collabora.com>

	* docs/gst_plugins_cache.json:
	  doc: update gst_plugins_cache.json
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/991>

2021-05-05 13:20:04 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix decoder glitches with H264 content
	  To avoid decoder starvation causing glitches on screen,
	  the demuxer shall clip only when the buffer is a key frame
	  and the lace time is greater than the stop time.
	  Fixes gst-editing-services#128
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/973>

2021-05-11 20:41:38 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtoverlay.cc:
	  qml: don't use buffers that have invalid contents
	  If the GL context is not shareable, ignore it.
	  A future change may also not output the relevant output either.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/983>

2021-05-11 20:38:52 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqsgtexture.cc:
	  qml: also use the dummy texture when no buffer has been set
	  Fixes corrupted texture output when changing OpenGL display/contexts.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/983>

2021-05-11 17:20:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* docs/gst_plugins_cache.json:
	  doc: Update cache for RGBP format addition
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>

2021-04-23 14:37:46 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Advertise codec-alpha in caps
	  This will be used to select the appropriate decoders. We also only attach the
	  GstVideoCodecAlphaMeta if the AlphaMode element is set, this is to stay on the
	  safe side and mimic what browsers (verified in Firefox and Chromium code) do.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>

2021-03-22 16:58:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Store alpha stream in VideoCodecAlphaMeta
	  This generalize the feature over using mini object quark data. If
	  that feature was Matroska specifc, using the new CustomMeta would have
	  been enough and arguably cleaner then QData, though it seems that
	  similar technique is use with AV1 Image Format (AVIF).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>

2016-12-03 14:27:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: extract VP8 alpha from BlockAdditionals
	  And put it on buffers as qdata (which is easier in this
	  case than a private custom meta because it can be picked
	  up easily in other modules).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>

2021-05-03 17:39:05 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.cc:
	* ext/qt/gstqtglutility.h:
	* ext/qt/gstqtoverlay.cc:
	* ext/qt/qtitem.cc:
	* ext/qt/qtwindow.cc:
	  qt: return a different GstGLDisplay object when the first sink requests
	  This allows the 'replace-gstreamer-opengl-context' context machinery to
	  correctly replace the OpenGL context used by the pipeline when the first
	  qmlglsink is added to the pipeline.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/982>

2021-05-07 11:16:47 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Plug leaks of saddr in error cases
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>

2021-05-07 11:16:21 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Whitespace
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>

2021-05-07 00:43:44 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Plug a method subobject leak
	  Changing the method would leak the previous method.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/976>

2021-05-06 15:04:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/gstvp9enc.c:
	  vp9enc: Add color range support
	  When setting the colorspace, we now clear the range to reduced range,
	  the default, and then we also set the range so the VP9 encoder encodes
	  the right information in the bitstream.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>

2021-05-06 14:51:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvpxenc.c:
	  vp9enc: Move colorspace configuration in VP9 enc
	  This is not supported by VP8 and was causing a warning.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>

2021-05-06 14:48:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvpxenc.c:
	* ext/vpx/gstvpxenc.h:
	  vpxdenc: Add a GstVideoCodecState to configure_encoder virtual
	  This will be needed to configure the VP9 specific colorimetry, which is
	  currently configured for VP8 casing warning.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>

2021-05-05 16:48:10 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Remove coordinates double-translation
	  Remove our own translation in the mouse event capture code, as that
	  translation will be done through the navigation interface.
	  Tested by resizing the window created by:
	  gst-launch-1.0 -v videotestsrc ! navigationtest ! glupload ! glcolorconvert ! tee name=t  ! gtkglsink
	  and checking that the cursor follows the mouse as expected.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>

2021-05-05 14:28:15 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gstgtkbasesink.c:
	  gtk: Translate navigation events coordinates
	  If the application passed down some pointer coordinates, translate those
	  from display coordinates to stream coordinates, so things work as
	  expected even if the video is resized.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>

2021-05-05 14:24:31 +0200  Bastien Nocera <hadess@hadess.net>

	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	  gtk: Export _display_size_to_stream_size()
	  Export _display_size_to_stream_size() so that GstNavigation implementors
	  can translate from display coordinates to stream coordinates before
	  pushing the events upstream to the DVD source, for example.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>

2018-02-26 17:26:41 +0100  David Fernandez <d.fernandezlop@gmail.com>

	* docs/gst_plugins_cache.json:
	* gst/matroska/matroska-mux.c:
	  matroska-mux: Change accepted caps width and height from [16, MAX] to [1, MAX]
	  There are cases where the video size might be less than 16x16.
	  This change allows the Matroska muxer to accept this cases.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/539>

2021-04-20 22:08:23 +0200  François Laignel <fengalin@free.fr>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtsp/gstrtspsrc.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/flvmux.c:
	* tests/check/elements/interleave.c:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rtpbin.c:
	* tests/check/elements/rtpcollision.c:
	* tests/check/elements/rtpmux.c:
	* tests/check/elements/splitmuxsink.c:
	* tests/check/elements/videomixer.c:
	* tests/examples/rtp/client-PCMA.c:
	* tests/examples/rtp/server-alsasrc-PCMA.c:
	  Use gst_element_request_pad_simple
	  Instead of the deprecated gst_element_get_request_pad.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/958>

2021-04-30 08:12:47 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	  qtmux: Make sure to write 64-bit STCO table when needed.
	  qtmux attempts to choose between writing a 32-bit stco chunk offset table
	  when it can, but switch to a 64-bit co64 table when file offsets go over
	  4GB.
	  This patch fixes a problem where the atom handling code was checking
	  mdat-relative offsets instead of the final file offset (computed by
	  adding the mdat position plus the mdat-relative offset) - leading to
	  problems where files with a size between 4GB and 4GB+offset-of-the-mdat
	  would write incorrect STCO tables with some samples having truncated
	  32-bit offsets.
	  Smaller files write STCO correctly, larger files would switch to
	  co64 and also output correctly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/970>

2021-04-22 15:01:32 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Add interlace-mode back to caps for camera
	  skip_try_fmt_probes is set to TRUE for v4l2src to skip interlace-mode and
	  colorimetry when probe caps. gst_v4l2_object_set_format_full() will add
	  colorimetry back to caps when iterating over the negotiated caps. There is
	  one case that v4l2src is first in preview state then starts recording.
	  v4l2src caps will change with an additional interlace-mode structure after
	  renegotiation, then v4l2src needs to reset. But this camera driver can't
	  orphan buffer pool, it causes require buffer failed as streaming is still
	  in active state.
	  To fix this, also need to add interlace-mode back to caps for camera to
	  avoid reset.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/961>

2021-04-02 18:41:28 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtpopuspay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtpopuspay: set MARKER flag
	  Set MARKER flag on first buffer after DTX.
	  According to RFC 3551 section 4.1 the marker bit needs to be set on
	  "the first packet after a silence period during which packets have
	  not been transmitted contiguously".
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>

2021-03-31 11:18:30 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtpopuspay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtpopuspay: add DTX support
	  If enabled, the payloader won't transmit empty frames.
	  Can be tested using:
	  opusenc dtx=true bitrate-type=vbr ! rtpopuspay dtx=true
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>

2021-04-24 11:15:50 -0400  Doug Nazar <nazard@nazar.ca>

	* ext/taglib/gstid3v2mux.cc:
	  taglib: Update createFrame() to non-deprecated version.
	  ID3v2::FrameFactory::createFrame() versions not taking a Header have
	  been deprecated since v1.5 (Jan 2008).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/965>

2021-04-25 02:16:45 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix divide-by-zero
	  The estimated packet-duration can sometimes end up as zero, and dividing
	  by that is never a good idea...
	  The test reproduces the scenario, and the fix is easy.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/966>

2020-06-02 19:38:33 +0200  Havard Graff <havard@pexip.com>

	  rtpjitterbuffer: clean up and improve missing packets handling
	  * Try to make variable and function names more clear.
	  * Add plenty of comments describing the logic step-by-step.
	  * Improve the logging around this, making the logs easier to read and
	  understand when debugging these issues.
	  * Revise the logic of packets that are actually beyond saving in doing
	  the following:
	  1. Do an optimistic estimation of which packets can still arrive.
	  2. Based on this, find which packets (and duration) are now hopelessly
	  lost.
	  3. Issue an immediate lost-event for the hopelessly lost and then add
	  lost/rtx timers for the ones we still hope to save, meaning that if
	  they are to arrive, they will not be discarded.
	  * Revise the use of rtx-delay:
	  Earlier the rtx-delay would vary, depending on the pts of the latest
	  packet and the estimated pts of the packet it being issued a RTX for,
	  but now that we aim to estimate the PTS of the missing packet accurately,
	  the RTX delay should remain the same for all packets.
	  Meaning: If the packet have a PTS of X, the delay in asked for a RTX
	  for this packet is always a constant X + delay, not a variable one.
	  * Finally ensure that the chaotic "check-for-stall" tests uses timestamps
	  that starts from 0 to make them easier to debug.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/952>

2021-04-23 12:07:52 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	  level: make properties thread-safe
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/962>

2021-04-22 14:11:09 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/level/gstlevel.c:
	  level: disable passthrough when audio-level-meta is enabled
	  Ensure we receive a writable buffer to add the meta.
	  Fix #878
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/962>

2021-04-23 08:28:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Don't pass a non-GObject pointer to GST_DEBUG_OBJECT and similar
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/964>

2021-04-22 08:57:23 +0200  Edward Hervey <edward@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Avoid generation of invalid timestamps
	  When updating timestamps and timer timeouts with a new offset, make sure that
	  the resulting value is valid (and not a negative (signed) value which ends up in
	  a massive (unsigned) value).
	  Fixes #571
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/960>

2021-04-21 18:41:08 +0100  Philippe Normand <philn@igalia.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Fix glib warning emitted when attribute query fails
	  The v4l2object is not a GstObject. Logging has to go through its dbg_obj
	  specially meant for this.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/959>

2021-03-25 13:20:38 +0100  VaL Doroshchuk <valentyn.doroshchuk@qt.io>

	* ext/qt/gstqtoverlay.cc:
	* tests/examples/qt/qmloverlay/overlay.py:
	  qmloverlay: Use first found GstGLVideoItem as widget property
	  GstGLVideoItem is required to render input video in the overlay's qml.
	  And currently qmlgloverlay requires to set this GstGLVideoItem to its widget property.
	  Instead of fetching GstGLVideoItem from the overlay's root object (root-item prop),
	  and setting it back as a widget (widget prop),
	  proposing to use found GstGLVideoItem in the current object hierarchy (passed in qml-scene) by default.
	  Also useful in Python, which solves the issue when casting gpointer <=> QQuickItem* is required.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/919>

2021-04-19 16:39:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2.c:
	  v4l2: fix debug category initialisation again
	  Would spew warnings on the rpi4 when calling into
	  gst_v4l2_object_get_codec_caps() from the probe_and_register()
	  function since the v4l2_debug category initialisation would
	  only be done later as part of the element/device provider
	  registration.
	  Also log things in the probe function to the v4l2 category
	  instead of the default category while we're at it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/956>

2021-04-19 01:29:33 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix race saving seek event seqnum.
	  We need to save the seek seqnum before the flush stop event
	  since that will start the basesrc task which may send the segment
	  event before we're ready.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/954>

2021-03-31 10:52:14 +0200  Marco Felsch <m.felsch@pengutronix.de>

	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  qmlglsink: allow to set force-aspect-ratio property
	  Add the forceAspectRatio Q_PROPERTY to allow changing the aspect ratio
	  from QML code as well.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/937>

2021-04-19 11:14:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: fix spurious SOURCE_CHANGED error-level log messages
	  They're harmless, and some drivers at least return EINVAL
	  instead of ENOTTY for unsupported events (here: uvcvideo).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/955>

2021-04-14 16:32:06 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: remove use of packed struct for payload
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/950>

2021-04-14 11:13:45 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/dtmf/gstdtmfcommon.h:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: convert to bit accessors
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/949>

2021-04-13 09:23:12 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Remove some dead code
	  stop is not used after this point, nor do we create a new segment
	  here since 84725d62b57bc74ce34abde755f35bf8f948f94d
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>

2021-04-10 02:53:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Do not overwrite the known duration after a seek
	  This breaks the duration query and also the seeking query.
	  Broke in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>

2021-04-10 04:40:46 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Just assign the segment instead of memcpy
	  Assignments copy by value, we don't need to memcpy...
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>

2021-04-13 11:30:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Check srcresult before waiting on the condition variable too
	  It might've been set to FLUSHING between the last check and the waiting,
	  and in that case we'd be waiting here forever now.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/944>

2021-04-12 23:15:17 -0400  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/rtpsession.c:
	  rtp: fix test_twcc_header_and_run to support big endian.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>

2021-04-12 23:13:15 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/rtpmanager/rtptwcc.c:
	  rtp: fix rtptwcc to support big endian.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>

2021-04-12 21:59:45 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	  rtp: fix rtphdrextrfc6464 to support big endian.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>

2021-04-12 21:36:58 -0400  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/alpha.c:
	  tests: Fix alpha test on big endian machines.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/943>

2021-03-19 02:51:20 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Protect against writing absurd sample durations
	  If the input DTS goes backward or is missing, the calculated
	  sample duration goes negative and wraps around to a very big
	  number. In that case, just write a sample with a duration of
	  0 and hope the problem is transient.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/914>

2021-04-10 03:09:44 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: De-dup seek event seqnums to avoid multiple seeks
	  Seek events are sent upstream on each sink, so if we receive multiple
	  seeks with the same seqnum, we must only perform one seek, not N seeks
	  where N = the number of sinks in the pipeline connected to rtspsrc.
	  This is the same thing done by demuxers like qtdemux or matrsokademux.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/938>

2021-04-10 01:55:28 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Using multicast UDP has no relation to seekability
	  The transport has no relation to whether a media can be seeked. The
	  range response having a duration is the correct thing to check for.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/939>

2021-04-10 01:54:48 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Add more logging for range parsing and seekable
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/939>

2021-04-10 14:47:23 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	  videocrop: Update documentation cache
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>

2021-04-07 21:57:11 +0200  Markus Ebner <info@ebner-markus.de>

	* gst/videocrop/gstvideocrop-private.h:
	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	  videocrop: Add support for GBR* video formats
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>

2021-04-07 18:54:49 +0200  Markus Ebner <info@ebner-markus.de>

	* gst/videocrop/gstvideocrop-private.h:
	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	  videocrop: Added support for planar pixel formats > 8bits
	  - Added support for planar pixel formats with depths greater than 8bits
	  to transform_planar implementation
	  - Added a whole lot of new pixel formats to the support-list
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>

2021-04-07 17:52:34 +0200  Markus Ebner <info@ebner-markus.de>

	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop-private.h:
	* gst/videocrop/gstvideocrop.c:
	  videocrop: Move supported format list into private header
	  - Moved declaration of supported pixel formats to private header, which
	  can be shared between videocrop and aspectvideocrop
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>

2021-04-06 17:02:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: More logging when calculating rfc7273 timestamps
	  This code can be fragile, since it is very exacting in the timestamps
	  that it will accept. Add more logging so it's easier to debug issues
	  and figure out whether it's a bug in the calculation or something
	  wrong in the incoming buffers.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/934>

2021-04-08 13:29:10 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	  rtp: missing debug init after element splitting
	  - h264depay
	  - h265depay
	  - sv3vdepay
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/936>

2020-03-30 09:29:07 +0200  Michal Dzik <michal.dzik@streamunlimited.com>

	* gst/rtp/gstrtpsbcpay.c:
	  rtp: rename gst_rtp_sbc_pay_flush_buffers()
	  gst_rtp_sbc_pay_flush_buffers() is a misleading name. A better name would
	  be gst_rtp_sbc_pay_drain_buffers(), because that's what it does, it drains
	  any leftover queued data and pushes it downstream. "Flushing" in GStreamer
	  typically means to throw away any queued data and not process/push it
	  downstream.
	  Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>

2020-03-24 13:31:00 +0100  Michal Dzik <michal.dzik@streamunlimited.com>

	* gst/rtp/gstrtpsbcpay.c:
	  rtp: fix adapter flushing in sbc payloader
	  GstAdapter must be flushed in some cases (flush, new segment, state change)
	  Without it, it may, for example, push some leftover buffer from old
	  segment in new segment. This, in general, breaks timestamps.
	  See GstAdapter documentation for more.
	  Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>

2020-08-18 20:16:06 +0200  Jakub Adam <jakub.adam@collabora.com>

	* ext/vpx/gstvpxenc.c:
	  vpxenc: add colorspace information into VP9 bitstream
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/871>

2021-03-26 16:26:22 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Use default colorimetry if that in caps is unknown
	  Some streams have unknown colorimetry in caps, but v4l2object sets
	  default values for each primaries. It will cause check colorimetry
	  fail when do gst_v4l2_video_colorimetry_matches().
	  To fix this, need to keep the unknown colorimetry in caps same as
	  the default value set by v4l2object.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/923>

2021-03-31 16:37:56 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Take segment stop into account when need_segment
	  Otherwise, in the case of e.g. a deferred seek event, the segment stop
	  would be replaced with GST_CLOCK_TIME_NONE.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/929>

2021-03-29 16:45:26 +0200  Val Doroshchuk <val@sevendof.com>

	* ext/qt/gstqtoverlay.cc:
	* ext/qt/gstqtoverlay.h:
	  gstqtoverlay: Add initialization and finalization to qml-scene prop
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/924>

2021-03-31 10:21:59 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.h:
	  qt: fix build warning with clang and c-linkage of user defined type
	  In file included from ../subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:17:
	  ../subprojects/gst-plugins-good/ext/qt/gstqtglutility.h:35:16: error: 'qt_opengl_native_context_from_gst_gl_context' has C-linkage specified, but returns user-defined type 'QVariant' which is incompatible with C [-Werror,-Wreturn-type-c-linkage]
	  QVariant       qt_opengl_native_context_from_gst_gl_context     (GstGLContext * context);
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/928>

2021-03-30 09:45:45 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* ext/qt/gstqtelement.cc:
	* ext/qt/gstqtelements.h:
	* ext/qt/gstqtoverlay.cc:
	* ext/qt/gstqtsink.cc:
	* ext/qt/gstqtsrc.cc:
	  qt: hotfix: allow per feature registration
	  Fixes #869
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/925>

2021-02-17 08:52:40 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/twolame/gsttwolamemp2enc.c:
	* ext/twolame/gsttwolamemp2enc.h:
	  twolame: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:49:03 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:38:46 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/cairo/gstcairo.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairooverlay.h:
	  cairo: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:34:34 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  y4m: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:32:26 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:29:40 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	  wavenc: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:27:24 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:27:12 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/monoscope/gstmonoscope.c:
	* gst/monoscope/gstmonoscope.h:
	  monoscope: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:22:47 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:19:52 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/id3demux/gstid3demux.c:
	* gst/id3demux/gstid3demux.h:
	  id3demux: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:16:33 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/icydemux/gsticydemux.c:
	* gst/icydemux/gsticydemux.h:
	  icydemux: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:14:26 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	  goom2k1: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:11:26 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/cutter/gstcutter.c:
	* gst/cutter/gstcutter.h:
	  cutter: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:11:14 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/goom/gstgoom.c:
	* gst/goom/gstgoom.h:
	  goom: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 17:10:33 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 16:34:48 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* sys/oss4/gstoss4audioplugin.c:
	* sys/oss4/meson.build:
	* sys/oss4/oss4-audio.c:
	* sys/oss4/oss4-audio.h:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-sink.h:
	* sys/oss4/oss4-source.c:
	* sys/oss4/oss4-source.h:
	  oss4: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 16:11:36 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* sys/oss/gstossaudio.c:
	* sys/oss/gstossaudioelement.c:
	* sys/oss/gstossaudioelements.h:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss/meson.build:
	  oss: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 15:56:35 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/auparse/gstauparse.c:
	* gst/auparse/gstauparse.h:
	  auparse: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 15:29:06 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/gstv4l2element.c:
	* sys/v4l2/gstv4l2elements.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/meson.build:
	  v4l2: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 15:05:43 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstgamma.h:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideobalance.h:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	* gst/videofilter/gstvideomedian.c:
	* gst/videofilter/gstvideomedian.h:
	* gst/videofilter/plugin.c:
	  videofilter: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 14:58:57 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocropelement.c:
	* gst/videocrop/gstvideocropelements.h:
	* gst/videocrop/gstvideocropplugin.c:
	* gst/videocrop/meson.build:
	  videocrop: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 14:54:15 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 14:49:56 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudp.c:
	* gst/udp/gstudpelement.c:
	* gst/udp/gstudpelements.h:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/udp/meson.build:
	  udp: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 14:43:32 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmpte.h:
	* gst/smpte/gstsmptealpha.c:
	* gst/smpte/gstsmptealpha.h:
	* gst/smpte/plugin.c:
	  smpte: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 14:38:37 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	  shapewipe: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 14:35:51 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtsp.c:
	* gst/rtsp/gstrtspelement.c:
	* gst/rtsp/gstrtspelements.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/meson.build:
	  rtsp: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 14:24:33 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpdtmfmux.h:
	* gst/rtpmanager/gstrtpfunnel.c:
	* gst/rtpmanager/gstrtpfunnel.h:
	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	* gst/rtpmanager/gstrtphdrext-rfc6464.h:
	* gst/rtpmanager/gstrtphdrext-twcc.c:
	* gst/rtpmanager/gstrtphdrext-twcc.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpjitterbuffer.h:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpptdemux.h:
	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxreceive.h:
	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpsession.h:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
	* gst/rtpmanager/gstrtpst2022-1-fecdec.h:
	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
	* gst/rtpmanager/gstrtpst2022-1-fecenc.h:
	  rtpmanager: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 13:49:15 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrglimiter.h:
	* gst/replaygain/gstrgvolume.c:
	* gst/replaygain/gstrgvolume.h:
	* gst/replaygain/replaygain.c:
	* gst/replaygain/rganalysis.h:
	  replaygain: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 13:43:44 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/multipart/multipart.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	* gst/multipart/multipartmux.c:
	* gst/multipart/multipartmux.h:
	  multipart: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 12:04:26 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/multifile/gstimagesequencesrc.c:
	* gst/multifile/gstimagesequencesrc.h:
	* gst/multifile/gstmultifile.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstmultifilesrc.h:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/gstsplitfilesrc.h:
	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multifile/gstsplitmuxsrc.h:
	  multifile: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 11:14:17 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/matroska/gstmatroskaelement.c:
	* gst/matroska/gstmatroskaelements.h:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska.c:
	* gst/matroska/meson.build:
	* gst/matroska/webm-mux.c:
	  matroska: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 10:59:34 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	  level: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 10:57:58 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-decode.h:
	* gst/law/alaw-encode.c:
	* gst/law/alaw-encode.h:
	* gst/law/alaw.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-decode.h:
	* gst/law/mulaw-encode.c:
	* gst/law/mulaw-encode.h:
	* gst/law/mulaw.c:
	  law: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 10:26:40 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/isomp4/gstisomp4element.c:
	* gst/isomp4/gstisomp4elements.h:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/isomp4-plugin.c:
	* gst/isomp4/meson.build:
	* gst/isomp4/qtdemux.c:
	  isomp4: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 09:57:27 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/interleave/deinterleave.c:
	* gst/interleave/gstinterleaveelements.h:
	* gst/interleave/interleave.c:
	* gst/interleave/plugin.c:
	  interleave: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-16 09:51:16 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/flx/gstflxdec.c:
	* gst/flx/gstflxdec.h:
	  flx: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 17:37:09 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvelement.c:
	* gst/flv/gstflvelements.h:
	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvplugin.c:
	* gst/flv/meson.build:
	  flv: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 17:27:51 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer.h:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/equalizer/gstiirequalizernbands.c:
	* gst/equalizer/gstiirequalizerplugin.c:
	* gst/equalizer/meson.build:
	  equalizer: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 15:37:52 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gsteffectv.c:
	* gst/effectv/gsteffectv.h:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	  effectv: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 15:03:10 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/dtmf/gstdtmf.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  dtmf: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 14:55:15 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/debugutils/breakmydata.c:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/gstcapsdebug.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstdebug.c:
	* gst/debugutils/gstdebugutilselements.h:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavigationtest.h:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	  debugutils: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 13:38:21 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/avi/gstavi.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavielement.c:
	* gst/avi/gstavielements.h:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/avi/meson.build:
	  avi: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 13:02:59 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautodetect.c:
	* gst/autodetect/gstautodetect.h:
	* gst/autodetect/gstautodetectelement.c:
	* gst/autodetect/gstautodetectelements.h:
	* gst/autodetect/gstautodetectplugin.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/autodetect/meson.build:
	  autodetect: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 13:00:38 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstaudioparserselements.h:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/audioparsers/plugin.c:
	  audioparsers: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 12:44:31 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/apetag/gstapedemux.c:
	* gst/apetag/gstapedemux.h:
	  apetag: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-15 11:00:46 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvpxelement.c:
	* ext/vpx/gstvpxelements.h:
	* ext/vpx/meson.build:
	* ext/vpx/plugin.c:
	  vpx: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 17:26:36 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/taglib/gsttaglibelement.c:
	* ext/taglib/gsttaglibelements.h:
	* ext/taglib/gsttaglibplugin.c:
	* ext/taglib/meson.build:
	  taglib: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 17:09:19 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/qt/gstplugin.cc:
	* ext/qt/gstqtelement.cc:
	* ext/qt/gstqtelements.h:
	* ext/qt/gstqtoverlay.cc:
	* ext/qt/gstqtsink.cc:
	* ext/qt/gstqtsrc.cc:
	* ext/qt/meson.build:
	* ext/qt/qtplugin.pro:
	  qt: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 16:09:53 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/speex/gstspeex.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexelement.c:
	* ext/speex/gstspeexelements.h:
	* ext/speex/gstspeexenc.c:
	* ext/speex/meson.build:
	  speex: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 16:04:16 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/soup/gstsoup.c:
	* ext/soup/gstsoupelement.c:
	* ext/soup/gstsoupelements.h:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/meson.build:
	  soup: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 15:53:19 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/raw1394/gst1394.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gstdv1394src.h:
	* ext/raw1394/gsthdv1394src.c:
	* ext/raw1394/gsthdv1394src.h:
	  raw1394: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 15:47:46 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/wavpack/gstwavpack.c:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackelement.c:
	* ext/wavpack/gstwavpackelements.h:
	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/meson.build:
	  wavpack: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 15:35:11 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 15:27:31 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiochebband.h:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiocheblimit.h:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audiodynamic.h:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audioecho.h:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofirfilter.h:
	* gst/audiofx/audiofx.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioiirfilter.h:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audioinvert.h:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiokaraoke.h:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiopanorama.h:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsincband.h:
	* gst/audiofx/audiowsinclimit.c:
	* gst/audiofx/audiowsinclimit.h:
	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempo.h:
	* gst/audiofx/gststereo.c:
	* gst/audiofx/gststereo.h:
	  audiofx: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 13:16:28 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstasteriskh263.h:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpL24depay.h:
	* gst/rtp/gstrtpL24pay.c:
	* gst/rtp/gstrtpL24pay.h:
	* gst/rtp/gstrtpL8depay.c:
	* gst/rtp/gstrtpL8depay.h:
	* gst/rtp/gstrtpL8pay.c:
	* gst/rtp/gstrtpL8pay.h:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3depay.h:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpac3pay.h:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvdepay.h:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpbvpay.h:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltdepay.h:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpceltpay.h:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvdepay.h:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpdvpay.h:
	* gst/rtp/gstrtpelement.c:
	* gst/rtp/gstrtpelements.h:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722depay.h:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg722pay.h:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723depay.h:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg723pay.h:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726depay.h:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg726pay.h:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729depay.h:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpg729pay.h:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstdepay.h:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261depay.h:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph261pay.h:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263depay.h:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcdepay.h:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpilbcpay.h:
	* gst/rtp/gstrtpisacdepay.c:
	* gst/rtp/gstrtpisacdepay.h:
	* gst/rtp/gstrtpisacpay.c:
	* gst/rtp/gstrtpisacpay.h:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kdepay.h:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpj2kpay.h:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegdepay.h:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpjpegpay.h:
	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvdepay.h:
	* gst/rtp/gstrtpklvpay.c:
	* gst/rtp/gstrtpklvpay.h:
	* gst/rtp/gstrtpldacpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp1sdepay.h:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tdepay.h:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp2tpay.h:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4adepay.h:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4apay.h:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gdepay.h:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4gpay.h:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vdepay.h:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpadepay.h:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpapay.h:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmparobustdepay.h:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvdepay.h:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtpmpvpay.h:
	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopusdepay.h:
	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtpopuspay.h:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmadepay.h:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmudepay.h:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtppcmupay.h:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqcelpdepay.h:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpqdmdepay.h:
	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpredenc.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsbcdepay.h:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsbcpay.h:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirendepay.h:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpsirenpay.h:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexdepay.h:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpspeexpay.h:
	* gst/rtp/gstrtpstorage.c:
	* gst/rtp/gstrtpstreamdepay.c:
	* gst/rtp/gstrtpstreamdepay.h:
	* gst/rtp/gstrtpstreampay.c:
	* gst/rtp/gstrtpstreampay.h:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtpsv3vdepay.h:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheoradepay.h:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtptheorapay.h:
	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbisdepay.h:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvorbispay.h:
	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp8pay.h:
	* gst/rtp/gstrtpvp9depay.c:
	* gst/rtp/gstrtpvp9depay.h:
	* gst/rtp/gstrtpvp9pay.c:
	* gst/rtp/gstrtpvp9pay.h:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawdepay.h:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtp/gstrtpvrawpay.h:
	* gst/rtp/meson.build:
	* tests/check/meson.build:
	  rtp: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 11:12:34 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/pulse/gstpulseelement.c:
	* ext/pulse/gstpulseelements.h:
	* ext/pulse/meson.build:
	* ext/pulse/plugin.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulse: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 10:41:29 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/mpg123/gstmpg123audiodec.c:
	* ext/mpg123/gstmpg123audiodec.h:
	  mpeg123: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 10:33:50 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/libpng/gstpng.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngdec.h:
	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	  libpng: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 10:27:18 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/lame/gstlamemp3enc.c:
	* ext/lame/gstlamemp3enc.h:
	* ext/lame/plugin.c:
	  lame: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 10:26:26 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/libcaca/gstcacaplugin.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libcaca/gstcacasink.h:
	* ext/libcaca/gstcacatv.c:
	* ext/libcaca/gstcacatv.h:
	* ext/libcaca/meson.build:
	  libcaca: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 10:09:46 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/jpeg/gstjpeg.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegelements.h:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegplugin.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/jpeg/meson.build:
	  jpeg: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 09:56:36 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 08:57:55 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* ext/gdk_pixbuf/gstgdkpixbufelement.c:
	* ext/gdk_pixbuf/gstgdkpixbufelements.h:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/meson.build:
	  gdk_pixbuf: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-12 08:48:21 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gstgtksink.c:
	* ext/gtk/gstgtksink.h:
	* ext/gtk/gstplugin.c:
	  gtk: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-11 19:53:30 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/flac/gstflac.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacelement.c:
	* ext/flac/gstflacelements.h:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/flac/meson.build:
	  flac: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-02-11 18:57:03 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/dv/gstdv.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvelement.c:
	* ext/dv/gstdvelements.h:
	* ext/dv/meson.build:
	  dv: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2020-08-14 15:27:31 -0400  Julian Bouzas <julian.bouzas@collabora.com>

	* ext/aalib/gstaaplugin.c:
	* ext/aalib/gstaasink.c:
	* ext/aalib/gstaasink.h:
	* ext/aalib/gstaatv.c:
	* ext/aalib/gstaatv.h:
	* ext/aalib/meson.build:
	  aalib: allow per feature registration
	  Split plugin into features including
	  dynamic types which can be indiviually
	  registered during a static build.
	  More details here:
	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>

2021-03-19 17:19:43 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* docs/gst_plugins_cache.json:
	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	* tests/check/elements/videocrop.c:
	  videocrop: handle non raw caps features
	  Currently, videocrop, only negotiates raw caps (system memory) because
	  it's the type of memory it can modify. Nonetheless, it's also possible
	  for the element to handle non-raw caps when only adding the crop meta
	  is possible, in other words, when downstream buffer pools expose the
	  crop API.
	  This patch enable non-raw caps negotiation. If downstream doesn't
	  expose crop API and negotiated caps are featured, the negotiation
	  fails.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/915>

2021-03-19 10:35:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Don't special-case G_SIGNAL_RUN_CLEANUP stage in signal accumulators
	  All these signals don't run the class handler in the CLEANUP stage.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/913>

2021-03-19 10:34:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/shout2/gstshout2.c:
	  shout2: Don't register signal without class handler with G_SIGNAL_RUN_CLEANUP
	  There is no class handler to run during the CLEANUP stage.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/913>

2021-03-23 16:59:28 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Avoid colorimetry mismatch for streams with invalid colorimetry
	  video-info sets gst colorimetry to default value when colorimetry in caps
	  is unparsable or invalid. Then v4l2object uses this gst colorimetry to do
	  mapping with v4l2 colorimetry. This may cause colorimetry mismatch when
	  check mapped gst colorimetry with that read from caps directly.
	  To fix this, need to correct gst colorimetry as that parsed from video-info
	  when check gst_v4l2_video_colorimetry_matches().
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/917>

2021-03-19 10:52:26 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Add support for hdr10 stream playback
	  Colorimetry of hdr10 video is bt2100-pq with transfer as
	  GST_VIDEO_TRANSFER_SMPTE2084. So map GST_VIDEO_TRANSFER_SMPTE2084
	  to V4L2_XFER_FUNC_SMPTE2084 to support hdr10 stream playback.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/888>

2021-03-20 10:41:29 -0500  Sid Sethupathi <sid.sethupathi@gmail.com>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: fix broken link in docs
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/916>

2021-03-18 17:42:02 +0000  Alba Mendez <me@alba.sh>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix more signals
	  Behaviour change in GLib causes select-stream signal to discard
	  the value returned by handlers. See !909 for more info.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/912>

2021-03-18 19:52:53 +1100  Matthew Waters <matthew@centricular.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/pulse/pulsesink.h:
	* ext/qt/gstqsgtexture.cc:
	* ext/qt/gstqtglutility.cc:
	* ext/qt/qtglrenderer.cc:
	* ext/qt/qtitem.cc:
	* ext/qt/qtwindow.cc:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxenc.c:
	* gst/audioparsers/gstac3parse.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/ximage/ximageutil.c:
	  gst: don't use volatile to mean atomic
	  volatile is not sufficient to provide atomic guarantees and real atomics
	  should be used instead.  GCC 11 has started warning about using volatile
	  with atomic operations.
	  https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719
	  Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/911>

2021-03-17 15:54:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	  Update docs cache and fix before-send signal doc syntax
	  The docs for before-send were missing because of this
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>

2021-03-17 13:18:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix accumulation of before-send signal return values
	  Since glib 2.62, the accumulated return values in RUN_CLEANUP override the
	  accumulated return values in RUN_FIRST. Since:
	  1. We have a default handler that always returns TRUE, and
	  2. User handlers are only run in RUN_FIRST, and
	  3. Our accumulator just takes the latest return value
	  We were discarding the return value from the user handler and always
	  sending messages even if the user handler said not to. See
	  https://gitlab.gnome.org/GNOME/glib/-/issues/2352 for more details.
	  This signal does not need RUN_CLEANUP or RUN_FIRST, so just change it
	  to RUN_LAST so that it's emitted exactly once and accumulated once.
	  With this fix, this signal can now be used to intercept PAUSE when
	  going to GST_STATE_NULL so that the server does a TEARDOWN (if
	  necessary) and not a PAUSE, which will confuse other RTSP clients when
	  playing shared media.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>

2021-03-17 11:32:08 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  Revert unusable workaround for PAUSE being sent when going NULL
	  Directly setting rtspsrc to the NULL state before putting the pipeline
	  in the NULL state usually works, but it can cause a deadlock in some
	  cases, so it's not a reliable mechanism to fix this.
	  This reverts commit f37afdafff1fd0a339966116261f5cd0de53f5d1:
	  "rtspsrc: Fix state changes from PAUSED to PLAYING"
	  and commit 76d624b2df5594a82269b94dffe8766a372d059d:
	  "rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL"
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/908>

2021-03-16 19:25:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix parsing of the mediaclk:direct= field
	  Due to an off-by-one when parsing the string, the most significant digit
	  or the clock offset was skipped when parsing the offset.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/907>

2021-03-16 00:08:43 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix state changes from PAUSED to PLAYING
	  This was accidentally broken in the last commit that touched this
	  because I missed the fall-through in the case immediately above this.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/906>

2021-03-04 13:05:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Fix extraction of multichannel WavPack
	  The old code had a couple of issues that all lead to potential memory
	  safety bugs.
	  - Use a constant for the Wavpack4Header size instead of using sizeof.
	  It's written out into the data and not from the struct and who knows
	  what special alignment/padding requirements some C compilers have.
	  - gst_buffer_set_size() does not realloc the buffer when setting a
	  bigger size than allocated, it only allows growing up to the maximum
	  allocated size. Instead use a GstAdapter to collect all the blocks
	  and take out everything at once in the end.
	  - Check that enough data is actually available in the input and
	  otherwise handle it an error in all cases instead of silently
	  ignoring it.
	  Among other things this fixes out of bounds writes because the code
	  assumed gst_buffer_set_size() can grow the buffer and simply wrote after
	  the end of the buffer.
	  Thanks to Natalie Silvanovich for reporting.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/859
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/902>

2021-03-03 11:31:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Initialize track context out parameter to NULL before parsing
	  Various error return paths don't set it to NULL and callers are only
	  checking if the pointer is NULL. As it's allocated on the stack this
	  usually contains random stack memory, and more often than not the memory
	  of a previously parsed track.
	  This then causes all kinds of memory corruptions further down the line.
	  Thanks to Natalie Silvanovich for reporting.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/858
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/902>

2021-03-15 12:57:19 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL
	  This usually doesn't matter, but it is disruptive when streaming from
	  a shared media since it will pause all other clients when any client
	  exits.
	  This new behaviour is opt-in and should be safe because you need to
	  set the NULL state on rtspsrc directly, instead of just on the
	  pipeline. See the updated documentation for an explanation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/901>

2021-01-18 15:54:43 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: handle GST_VIDEO_TRANSFER_BT601
	  V4L2 makes no difference between the BT.601 and BT.709 transfer
	  functions [1], but GStreamer does since 1.18 [2].
	  Adapt gst_v4l2_object_get_colorspace() and
	  gst_v4l2_object_set_format_full().
	  [1] https://linuxtv.org/downloads/v4l-dvb-apis-new/userspace-api/v4l/colorspaces-details.html#colorspace-smpte-170m-v4l2-colorspace-smpte170m
	  [2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/724
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/856>

2021-03-11 22:22:15 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix title of a few properties docstrings
	  GstRtspSrc -> GstRTSPSrc
	  This would have been noticed by the since checker, but those
	  properties were introduced prior to that.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/899>

2021-03-07 21:25:01 +0000  Vladimir Menshakov <vladimir.menshakov@gmail.com>

	* docs/gst_plugins_cache.json:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackdec.h:
	  wavpackdec: Add floating point format support
	  This commit negotiate F32 audio format if MODE_FLOAT used in wavpack file.
	  Wavpack float mode is always in 32-bit IEEE format.
	  The following pipeline plays distorted audio if source file is encoded in float mode:
	  gst-launch-1.0 filesrc ... ! wavpackparse ! wavpackdec ! pulsesink
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/894>

2021-03-04 16:40:06 +1100  Matthew Waters <matthew@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska: also support push-mode from seek events sent to the element
	  Otherwise sending seek events would fail to actually seek.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/891>

2021-02-26 10:49:10 +0100  Marc Leeman <m.leeman@televic.com>

	* gst/rtsp/gstrtspsrc.c:
	  gstrtspsrc: 551 should not result in an unhandled error
	  Some cameras (e.g. HikVision DS-2CD2732F-IS) return "551 Option
	  not supported" when a command is sent that is not implemented
	  (e.g. PAUSE). Instead; it should return "501 Not Implemented".
	  This is wrong, as previously, the camera did announce support for PAUSE
	  in the OPTIONS.
	  In this case, handle the 551 as if it was 501 to avoid throwing errors
	  to application level. */
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/885>

2021-03-01 14:32:40 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Do not expose profiles/levels in vp8/vp9 template caps
	  Vp8/vp9 supported profiles/levels are listed in decoder sink caps, but
	  there is no parser for these two formats and the demuxers also don't have
	  these information. It causes negotiation fail between demuxers and decoder
	  when check caps "accept = gst_caps_is_subset (caps, template_caps);".
	  To fix this, need to remove profiles/levels for vp8/vp9 formats in decoder
	  sink caps.
	  Fix #854
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/887>

2021-03-03 18:30:39 +0900  Seungha Yang <seungha@centricular.com>

	* gst/rtpmanager/gstrtphdrext-twcc.h:
	  rtpmanager: Fix an MSVC compile warning
	  We don't expect this object is a part of public library.
	  gstrtphdrext-twcc.c(45): warning C4273: 'gst_rtp_header_extension_twcc_get_type': inconsistent dll linkage
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/889>

2021-02-24 13:25:43 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: fix src side frame rate negotiation
	  Negotiating v4l2h264dec ! v4l2h264enc transcoding pipelines fails in
	  case the encoder does not accept framerate=(fraction)0/1.
	  The acquired caps used for downstream negotiation are determined from
	  gst_v4l2_object_acquire_format(), which sets the GstVideoInfo::fps_n
	  and ::fps_d fields to 0.
	  To fix this, copy the frame rate from the sink side.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/882>

2021-02-16 16:20:05 +0200  Jordan Petridis <jpetridis@gnome.org>

	* sys/rpicamsrc/meson.build:
	  rpicamsrc: depend on posix threads and vchiq_arm
	  Could only test on rpi 3b+
	  Close #839
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/875>

2021-02-11 14:48:07 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Silence traces around unsupported source change
	  Don't be too spamy about unsupported source change flags as these will be
	  commonly extended in the future.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-11 14:24:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Move preferred resolution query before the probe
	  As we lock the DV_TIMINGS (and standards in the future), we need to probe the
	  caps after, otherwise, we may endup fixating to an unsupported resolution,
	  which would lead to a not-negotiated error.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-10 16:37:01 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2src: Calculate framerate from DV timings
	  And use this framerate in our preference. Note that we also flush
	  the probed caps as it seems that the format enumeration may change
	  when a new source change event get triggered.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-10 15:52:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2rc: Add DV_TIMINGS query and locking
	  This adds support to DV_TIMINGS query and locking. The timing width and
	  height is then used as a preference.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-10 15:49:03 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Force renegotiation on resolution change
	  As mandated by the specification, make sure to cycle through streamoff
	  / streamon regardless if the caps have changed or not.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-10 14:52:14 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Remove unused streaming member
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-10 10:48:48 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Refactor to use PreferredCapsInfo structure
	  Avoid passing around a bare structure for the preference, this removes
	  the need to copy and free that structure and simplify the code. Also
	  fix a type in the structure name, Prefered -> Preferred.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-08 17:27:20 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Stub preferred resolution support
	  This stubs the ability to use preferred resolution from digital
	  video timings, analog TV standards or driver reported native
	  resolution.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-09 14:44:02 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: Subscribe source_change for the current input
	  When we subscribe for source-change event, we need to specify for which
	  input. Make sure we subscribe for the current input.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-08 17:26:20 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2src: Add input signal status detection
	  As part of the support to select a preferred size, we can also
	  detect the signal status. This is a split patch so that feature
	  is separated to ease review.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-08 17:24:00 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: Add helper to query input status
	  This is a wrapper around ENUM_INPUT renamed for readability.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-08 17:22:37 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: Fix input/output index sign
	  This is an unsigned integer in the kernel API.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-04 16:59:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Add source resolution change support
	  This patch adds support for source resolution change detection.
	  Resolution change is signaled by drivers when a change in the detected
	  signal have been detected. This is notably seen on HDMI receivers.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-04 14:13:32 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Handle resolution change event
	  This patch adds the detection, dequeuing and reporting of the SOURCE_CHANGE
	  event when the CH_RESOLUTION flag is set. The acquire function will now return
	  a new custom success called GST_V4L2_FLOW_RESOLUTION_CHANGE. In order to use
	  this new feature, elements must enable it by calling:
	  gst_v4l2_buffer_pool_enable_resolution_change (pool);
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-04 11:01:38 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2object: Add event helpers
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2021-02-04 10:10:34 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: use FLOW_LAST_BUFFER
	  This uses the GST_V4L2_FLOW_LAST_BUFFER alias instead of
	  GST_FLOW_CUSTOM_SUCCESS to make the code more readable.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>

2018-12-10 14:10:05 +0100  Lucas Stach <l.stach@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: prefer NV12 over I420
	  Considering NV12 an 'odd' format is a historical artifact. This format
	  is now quite common, and usually preferable to I420 due to more memory
	  friendly access patterns.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/857>

2021-02-18 10:34:25 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/wavparse/gstwavparse.c:
	* tests/check/elements/wavparse.c:
	  wavparse: fix seeking in READY state
	  wavparse claims to be able to support seeking in the READY state by
	  saving the pending seek event and actually seeking later after having parsed the
	  header.
	  Problem was that this seek event was reset on the READY to PAUSED
	  transition, making all this code useless. Fixing it by stop resetting
	  on READY to PAUSED transition as we already reset on PAUSED to READY
	  and when initiating the element.
	  Note that DTS marker detection isn't support in such scenario as
	  gst_type_find_helper_for_buffer() needs a buffer containing the
	  beginning of the stream.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>

2021-02-18 10:05:03 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* tests/check/elements/wavparse.c:
	  tests: wavparse: factor out create_pipeline()
	  No semantic change.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>

2021-02-18 00:34:02 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	  docs: update plugins cache with new h264 / vp8 depay properties
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>

2020-12-09 01:40:45 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: expose request-keyframe property
	  When set, the depayloader will request new keyframes on packet
	  loss
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>

2020-12-09 01:34:20 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	  rtpvp8depay: expose request-keyframe property
	  When set, the depayloader will request new keyframes on packet
	  loss
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>

2020-12-09 01:24:57 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: expose wait-for-keyframe property
	  Similar to rtpvp8depay, when packet loss occurs, the depayloader
	  starts waiting for a keyframe.
	  We try to only stop waiting when all the packets for the new keyframe
	  have been received, by only resetting waiting_for_keyframe when
	  encountering the first packet of a keyframe, this is slightly
	  fragile because there is no bit that explicitly marks the start
	  of an access unit, so we rely on the existing picture_start
	  detection code.
	  As a consequence, the property is only meaningful when outputting
	  access units, and is ignored when outputting NALs directly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>

2021-02-18 00:36:43 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/videomixer/videomixer2.c:
	  videomixer: document as deprecated
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/878>

2021-02-16 22:20:17 +1100  Ashley Brighthope <ashley.b@reddegrees.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Fixed INFO chunk corruption, caused by odd sized data not being padded. Code style was updated.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/873>

2020-12-07 19:51:35 +0100  Jakub Adam <jakub.adam@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: add info regarding (non-standard) multichannel support
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>

2020-12-07 16:50:01 +0100  Jakub Adam <jakub.adam@collabora.com>

	* docs/gst_plugins_cache.json:
	  docs: update plugins cache for rtpopus
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>

2020-12-01 20:09:58 +0100  Jakub Adam <jakub.adam@collabora.com>

	* tests/check/elements/rtpopus.c:
	  tests: add rtpopus multichannel test cases
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>

2020-12-01 16:43:32 +0100  Jakub Adam <jakub.adam@collabora.com>

	* gst/rtp/gstrtpopusdepay.c:
	  rtpopusdepay: support libwebrtc-compatible multichannel payload
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>

2020-11-30 21:49:48 +0100  Jakub Adam <jakub.adam@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: support libwebrtc-compatible multichannel payload
	  When the audio has more than 2 channels, add optional fields to output
	  caps from which webrtcbin can generate SDP in the syntax recognized by
	  "multiopus" codec present in libwebrtc [1].
	  e.g. for 5.1 audio:
	  a=rtpmap:96 multiopus/48000/6
	  a=fmtp:96 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5
	  [1] https://webrtc-review.googlesource.com/c/src/+/129768
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>

2020-11-30 22:10:14 +0100  Jakub Adam <jakub.adam@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: make use of gst_rtp_base_payload_set_outcaps_structure()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>

2021-02-09 19:31:28 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/effectv/LICENSE:
	  effectv: Remove redundant license file
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/869>

2021-02-05 00:55:12 +0000  Kevin Song <kevinbing.song@gmail.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  Apply 1 suggestion(s) to 1 file(s)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>

2021-02-05 00:55:04 +0000  Kevin Song <kevinbing.song@gmail.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  Apply 1 suggestion(s) to 1 file(s)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>

2021-02-04 13:43:17 +0800  Bing Song <bing.song@nxp.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: support resolution change stream encode.
	  Resolution change stream transcoding will drain before send new video
	  frame buffer. Need encode video frame after process EOS.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>

2021-02-04 11:44:53 +0100  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: added support for cbcs encryption scheme
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/865>

2021-01-21 18:04:58 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
	* gst/rtpmanager/gstrtphdrext-rfc6464.h:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/meson.build:
	* tests/check/elements/rtphdrextrfc6464.c:
	* tests/check/meson.build:
	  rtp: add rtphdrextrfc6464
	  Header Extension for Client-to-Mixer Audio Level Indication as
	  defined in RFC 6464.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>

2020-06-16 12:01:30 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	* tests/check/elements/level.c:
	  level: add GstRTPAudioLevelMeta on buffers
	  This meta can be used by a RTP payloader to send the level information
	  to the peer.
	  Part of https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/446
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>

2021-02-03 17:10:20 +0200  Robert Swain <robert.swain@gmail.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Provide documentation for GST_DEINTERLACE_BUFFER_STATE
	  More information available in
	  https://gstconf.ubicast.tv/videos/interlacing-and-telecine-in-gstreamer/
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>

2021-01-30 16:16:13 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: Fix telecine/onefield mixup
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/838
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>

2021-01-30 15:49:23 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: Better alternate support
	  Improve line offset halving based on whether this field is top or
	  bottom.
	  Also handle the buffer state the same as mixed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>

2021-01-14 01:12:06 +0800  Bing Song <bing.song@nxp.com>

	* sys/v4l2/gstv4l2h265codec.c:
	  v4l2h265codec: fix HEVC profile string issue.
	  Keep HEVC profile compatible with other module.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/850>

2020-12-15 10:41:40 +0800  Bing Song <bing.song@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Need keep same transfer as input caps.
	  GST_VIDEO_TRANSFER_BT2020_12 and GST_VIDEO_TRANSFER_BT2020_10 will
	  be mapped to V4L2_XFER_FUNC_709. Need check input caps when map
	  V4L2_XFER_FUNC_709 back to GST_VIDEO_TRANSFER_BT2020_12 and
	  GST_VIDEO_TRANSFER_BT2020_10
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/816
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/841>

2020-12-07 10:01:53 +0100  Tobias Ronge <tobiasr@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Do not wait for response while flushing
	  Due to the may_cancel flag in GstRTSPConnection, receiving might not get
	  cancelled when supposed to. In this case, gst_rtsp_src_receive_response
	  will have to wait until timeout instead but if busy receiving RTP
	  data, this timeout will never occur.
	  With this patch, gst_rtsp_src_receive_response returns GST_RTSP_EINTR
	  if flushing is set to TRUE instead of continuing to receive.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/831>

2021-01-14 19:13:03 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/dv/meson.build:
	  meson: allow libdv subproject fallback
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/854>

2020-12-21 13:55:58 +0100  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Allow streams with no specified protection system ID
	  This is necessary in cases like CMAF where there won't be any events
	  passing thru.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/852>

2021-01-07 16:57:27 +0800  Hou Qi <qi.hou@nxp.com>

	* docs/gst_plugins_cache.json:
	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Map correct video format for RGBA
	  Map V4L2_PIX_FMT_RGBA32 pixel format to GST_VIDEO_FORMAT_RGBA instead of
	  GST_VIDEO_FORMAT_RGB video format to support RGBA.
	  Fixes #823
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/848>

2021-01-02 13:06:16 +0530  Sanchayan Maity <sanchayan@asymptotic.io>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix marker links
	  These should be with a single ':'. The double '::' results in a CI with
	  build failure message like below.
	  ERROR: [links]: (mandatory-link-not-found): Mandatory link Link GstSocketTimestamp -> None (GstSocketTimestamp) could not be resolved
	  ERROR: [check-missing-since-markers]: (missing-since-marker): Missing since marker for udpsrc:socket-timestamp
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/828>

2020-12-17 11:24:07 +0530  Sanchayan Maity <sanchayan@asymptotic.io>

	* docs/gst_plugins_cache.json:
	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: Allow use of socket control message timestamps for DTS
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/828>

2020-12-09 20:20:18 +1100  Matthew Waters <matthew@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	* tests/check/elements/videoflip.c:
	  videoflip: fix possible crash when setting the video-direction while running
	  A classic case of not enough locking.
	  One interesting thing with this is the interaction between the
	  rotation value and caps negotiation.  i.e. the width/height of the caps
	  can be swapped depending on the video-direction property.  We can't lock
	  the entirety of the caps negotiation for obvious reasons so we need to
	  do something else.  This takes the approach of trying to use a single
	  rotation value throughout the entirety of the negotiation and then
	  subsequent output frame in a kind of latching sequence.
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/792
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>

2020-12-09 19:49:47 +1100  Matthew Waters <matthew@centricular.com>

	* tests/check/elements/videoflip.c:
	* tests/check/meson.build:
	  tests: add tests for videoflip
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>

2020-12-30 13:38:46 +0100  Ignacio Casal Quinteiro <qignacio@amazon.com>

	* gst/deinterlace/meson.build:
	  deinterlace: force -DPREFIX on macos
	  This is due to a bug in meson where it will not detect properly
	  the compiler if the symbols need an undercore.
	  https://github.com/mesonbuild/meson/issues/5482
	  Fixes #821
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/845>

2020-12-15 11:36:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/842>

2020-12-10 14:27:49 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Avoid deadlock when releasing a pad from a running muxer
	  Might not drain correctly
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/838>

2020-12-11 11:24:14 +0800  Hou Qi <qi.hou@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Use active resolution during fallback colorspace probe
	  For legacy drivers that don't implement ENUM_FRAMESIZE, use active
	  resolution to probe colorspace. This can improve the accuracy of the
	  result when the colorspace depends on the resolution. This fixes a
	  wrong colorspace issue on board with vendor bsp at resolution 2560x1440.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/830>

2020-12-12 04:02:37 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
	  rtpst2022-1-fecdec: don't xor out of bounds
	  When reconstituting packets from a stream with variable packet
	  sizes, don't xor larger packets past the length of the protected
	  packet
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>

2020-12-12 04:00:41 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
	  rtpst2022-1-fecenc: memset when reallocating xored payload
	  When protecting packets with a variable payload length, we
	  reallocate the xored payload when needed. It is a good idea
	  to memset the extended memory to 0 so that we don't xor
	  data with garbage!
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>

2020-12-12 03:56:11 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
	  rtpst2022-1-fec-*: protect additional RTP header fields
	  While the standard is a bit vague about whether the padding,
	  extension and marker bits should be protected:
	  > The usage, by senders and receivers, of the following bits shall
	  > be defined by the associated video/audio transport standards:
	  It is obviously necessary and useful for some formats (eg VP8)
	  that those indeed be protected.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>

2020-12-12 03:28:56 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/splitmuxsink.c:
	  splitmuxsink: Unit test - check format/opened/closed sequence
	  Check the sequence of format-location/fragment-opened/fragment-closed
	  events is respected. There should be 1 format-location call for each
	  fragment-opened message, and 1 fragment-closed for each.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>

2020-12-09 00:40:52 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Fix for 'reference bytes muxed' check.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798
	  introduced a check in the need-new-fragment logic to avoid starting a
	  new fragment unless there has been some data on the reference stream,
	  but the check is done against the number of bytes that have been
	  received on the input, not the number that were released for output
	  into the current fragment.
	  Fix the check to remember and test against bytes that have been sent
	  for output.
	  This also fixes a problem where starting a new fragment fails to
	  request a new filename from the format-location signal.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>

2020-09-15 00:27:24 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Add debug for fragment opened/closed msgs
	  When posting fragment-opened and fragment-closed messages,
	  put a debug statement in the logs
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>

2020-08-18 16:06:14 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Convert asserts into element errors.
	  Change some g_assert into element errors so that they can be
	  caught and the pipeline shut down.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>

2020-07-10 15:36:54 +1000  Matthew Waters <matthew@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtpfunnel.c:
	* gst/rtpmanager/gstrtphdrext-twcc.c:
	* gst/rtpmanager/gstrtphdrext-twcc.h:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/meson.build:
	  rtpmanager: update for rtp header extensions
	  Provide an implementation of the transport-wide-cc header extension and
	  use it in rtpfunnel.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/808>

2020-11-15 11:30:07 +0000  Jose Quaresma <quaresma.jose@gmail.com>

	* sys/rpicamsrc/meson.build:
	  rpicamsrc: add vchostif library as it is required to build successful
	  fix: undefined reference to `vc_gencmd'
	  /usr/src/debug/gstreamer1.0-plugins-good/1.18.1-r0/build/../gst-plugins-good-1.18.1/sys/rpicamsrc/RaspiCamControl.c:1440: undefined reference to `vc_gencmd'
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/818>

2020-11-25 17:51:24 +0100  Marijn Suijten <marijns95@gmail.com>

	* tests/check/elements/rtp-payloading.c:
	  tests/rtp-payloading: Use new AudioFormatInfo::fill_silence function
	  The function is renamed to be properly associated with AudioFormatInfo
	  (its instance) instead of AudioFormat (an unrelated enum), see [1] for
	  the rename itself.
	  [1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940

2020-11-24 22:11:50 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/deinterlace/meson.build:
	* meson.build:
	  deinterlace: Enable x86 assembly with nasm on MSVC
	  We need to remove x86inc.asm from the list of compiled assembly files
	  because it is not supposed to be compiled separately. It is directly
	  included by yadif.asm, and it exports no symbols.
	  The object file was getting ignored on all platforms except on msvc
	  where it was causing a linker hang when building with debugging
	  enabled because the object file had no debug symbols (or similar).
	  We've seen this before in FFmpeg too, which uses nasm:
	  https://gitlab.freedesktop.org/gstreamer/meson-ports/ffmpeg/-/merge_requests/46
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/825>

2020-11-19 17:47:21 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtoverlay.cc:
	* ext/qt/gstqtsink.cc:
	  qml: add some docs on display and contexts
	  Especially considering some dynamic pipeline scenarios.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/822>

2020-11-18 20:09:24 +0100  Tim Schneider <tim.schneider94@t-online.de>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Added "src->started = FALSE;" to gst_rpi_cam_src_stop
	  Makes the element reusable multiple times after a state change back to READY.
	  Fixes #105
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/823>

2020-11-12 09:32:30 +0800  Bing Song <bing.song@nxp.com>

	* docs/gst_plugins_cache.json:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: caps negotiate wrong as interlace feature
	  gst_caps_simplify() will move interlace format before normal video
	  format. It will cause caps negotiate prefer interlaced caps which
	  isn't expected. Seperate normal caps and interlaced caps and then
	  merge it will keep prefer progress video format.
	  Add ARGB/BGRA for interlaced caps.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/802
	  Part-of <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/813>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/813>

2020-11-13 21:25:42 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: never send on a non-internal source
	  This will end up as a "received" packet, due to the code in
	  source_push_rtp, which will think this is a packet being received.
	  Instead drop the packet and hope that either:
	  1. Something upstream responds to the GstRTPCollision event and changes
	  SSRC used for sending.
	  2. That the application responds to the "on-ssrc-collision" signal, and
	  forces the sender (payloader) to change its SSRC.
	  3. That the BYE sent to the existing user of this SSRC will respond to
	  the BYE, and that we timeout this source, so we can continue sending
	  using the chosen SSRC.
	  The test reproduces a scenario where we previously would have sent
	  on a non-internal source.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>

2020-11-13 12:39:53 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: rewrite timeout-check to avoid underflow
	  If current_time is < collision_timeout, we get an uint64 underflow, and
	  the check will trigger prematurely.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>

2020-11-13 14:58:44 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Fix caps change handling
	  In baseparse we set the fixed caps flag on all src pads, therefore the
	  source pad caps query in get_allowed_caps will return the current caps.
	  Current caps won't necessarily intersect with the new caps (e.g. sample
	  rate change). Replace get_allowed_caps with peer_query_caps.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/816>

2020-11-12 23:39:21 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/qtdemux.c:
	  tests: qtdemux: fix typo in caps field
	  timesacle -> timescale
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>

2020-11-12 23:38:21 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/qtdemux.c:
	  tests: qtdemux: fix crash on 32-bit architectures
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/803
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>

2020-09-14 13:12:50 +0530  Sanchayan Maity <sanchayan@asymptotic.io>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpldacpay.c:
	* gst/rtp/gstrtpldacpay.h:
	* gst/rtp/meson.build:
	  rtp: ldacpay: Add LDAC RTP payloader
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/757>

2020-11-03 15:58:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/qt/gstqsgtexture.cc:
	* ext/qt/gstqsgtexture.h:
	* ext/qt/qtitem.cc:
	  qmlglsink: Keep old buffers around a bit longer if they were bound by QML
	  We don't know exactly when QML will stop using them but it should be
	  safe to unref them after at least 2 more buffers were bound.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/810>

2020-11-10 18:18:12 +0000  ChrisDuncanAnyvision <chrisd@anyvision.co>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Ensure same group-id used for both TCP/UDP stream-start events
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>

2020-11-10 16:17:23 +0000  ChrisDuncanAnyvision <chrisd@anyvision.co>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use consistent URI hashed stream-id for UDP and TCP/Interleaved streams
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>

2020-11-04 18:43:04 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	  meson: Enable some MSVC warnings for parity with GCC/Clang
	  This makes it easier to do development with MSVC by making it warn
	  on common issues that GCC/Clang error out for in our CI configuration.
	  Continuation from https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/223
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/809>

2020-10-15 21:42:40 -0400  Olivier Crête <olivier.crete@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpsource: Report for which local SSRC is a remote RB reporting on
	  This is useful in the Bundle case because there may be multiple local
	  and remote SSRCs in the same session.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/776>

2020-10-29 15:58:38 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpisacdepay.c:
	* gst/rtp/gstrtpisacpay.c:
	  docs: update plugins cache
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>

2020-03-20 13:15:33 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpisacdepay.c:
	* gst/rtp/gstrtpisacdepay.h:
	* gst/rtp/meson.build:
	  rtp: add rtpisacdepay
	  Depayload for the iSAC audio codec.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>

2020-03-20 13:15:33 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpisacpay.c:
	* gst/rtp/gstrtpisacpay.h:
	* gst/rtp/meson.build:
	  rtp: add rtpisacpay
	  Payload for the iSAC audio codec.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>

2020-11-01 18:36:49 +0000  Dinesh Manajipet <saidinesh5@gmail.com>

	* ext/qt/qtitem.cc:
	  qmlglsink: Set qtitem's implicit width/height
	  This can be useful to let the layouts automatically resize qtitem
	  and also easily query a video's width/height from QML
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/802>

2020-11-01 10:30:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Release pads via GstAggregator
	  See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/801>

2020-10-26 12:40:49 +1100  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: support muxing multiple codec_data for h264/h265
	  Each codec_data is put into its own SampleTableEntry inside the stsd.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/787>

2020-10-29 14:54:16 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstnavseek.h:
	  navseek: add hold_eos property
	  This property will tell the element to hold
	  the EOS event and keep it until the next
	  keystroke.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/792>

2020-10-31 12:52:04 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/splitmuxsrc.c:
	  splitmuxsrc: Fix comment in a test
	  Fix a comment in the splitmuxsrc robust muxing test so it
	  describes the test properly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>

2020-10-31 12:49:08 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Change EOS catching logic.
	  Add a new state for ending the overall stream, and use it to decide
	  whether to pass the final EOS message up the bus instead of dropping
	  it. Fixes a small race that makes the testsuite sometimes not generate
	  the last fragment(s) sometimes because the wrong EOS gets
	  allowed through too early.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>

2020-10-31 02:19:07 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Don't use the element state lock
	  Using the element state lock to avoid splitmuxsink shutting
	  down while doing element manipulations can lead to a deadlock on
	  shutdown if a fragment switch happens at exactly the wrong moment.
	  Use a private mutex and a shutdown boolean instead.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>

2020-10-30 03:38:15 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't busy loop on a non-ready pad.
	  If a pad gets into the check_completed_gop method and then
	  the underlying conditions change on the reference context,
	  things could get stuck in a busy loop when the context should
	  instead jump back out and wait for more data.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>

2020-10-30 03:36:51 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Mark running=false on shutdown.
	  Make sure that any late gst_element_call_async() callbacks
	  know that the elements is shutting down and bail out instead
	  of operating on the element we're trying to stop.
	  Fixes a spurious test failure in elements_splitmuxsrc
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>

2020-10-29 02:36:35 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Forward EOS messages from async fragments.
	  Re-enable forwarding EOS messages from fragments that are completing
	  asynchronously, so that splitmuxsink itself won't go EOS until they
	  are complete. This was disabled to work around a bug in core that
	  is fixed in
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/683
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>

2020-09-17 22:56:01 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Never start a new fragment with no reference buffers
	  If there has been no bytes from the reference stream muxed into
	  the current fragment, then time can't have advanced, there's no
	  GOP... this fragment would be broken or empty, so wait for some
	  data on the reference buffer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>

2020-10-29 02:38:16 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Chain up when releasing pad, and fix some locking.
	  Release pads by calling up into aggregator so it can do the right
	  things. Don't clean up the pad until after that.
	  Add some missing locks around some accesses to shared pad state.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797>

2018-08-13 15:35:11 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp9depay.c:
	* gst/rtp/gstrtpvp9depay.h:
	* tests/check/elements/rtpvp9.c:
	  rtpvp9depay: Improve SVC parsing, aggregate all layers
	  - Fix start and end of picture to support multiple layers. Start of
	  picture is the first packet of the base layer, while end of picture
	  is when the marker bit is set (last packet of the enhancement
	  layers).
	  - All "layers" (aka "frames") of a picture are pushed downstream in a
	  single buffer when picture is complete.
	  - Forgive SID=0 for enhancement layers (invalid, but Chrome and
	  Firefox sends it)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/773>

2020-10-30 03:09:48 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	* tests/check/elements/rtpvp8.c:
	  rtpvp8depay: Send lost events when marker bit is missing
	  This means the previous frame was incomplete.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/796>

2020-10-14 23:17:53 +0200  Knut Saastad <Knut@bitflow.io>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp9depay: detect incomplete frames and bail out
	  If a packet with the B bit set arrives but we haven't received
	  a packet with the marker or E bits set to end the previous frame,
	  we know the current frame was incomplete.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/795>

2020-10-14 23:17:53 +0200  Knut Saastad <Knut@bitflow.io>

	* gst/rtp/gstrtpvp9depay.c:
	  rtpvp9depay: detect incomplete frames and bail out
	  If a packet with the B bit set arrives but we haven't received
	  a packet with the marker or E bits set to end the previous frame,
	  we know the current frame was incomplete.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>

2020-10-14 01:28:50 +0200  Mikhail Fludkov <misha@pexip.com>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	* gst/rtp/gstrtpvp9depay.c:
	* gst/rtp/gstrtpvp9depay.h:
	* tests/check/elements/rtpvp8.c:
	* tests/check/elements/rtpvp9.c:
	  rtpvp*depay: possibly forward might-have-been-fec PacketLost events
	  This is ad adaptation of a Pexip patch for dealing with spurious
	  GstRTPPacketLost events caused by lost ulpfec packets: as FEC packets
	  under that scheme are spliced in the same sequence domain as the media
	  packets, it is not generally possible to determine whether a lost packet
	  was a FEC packet or a media packet.
	  When upstreaming pexip's ulpfec patches, we decided to drop all lost
	  events at the base depayloader level, and where the original patch
	  from pexip was making use of picture ids and marker bits to determine
	  whether a packet should be forwarded, this patch makes use of those
	  to determine whether they should be dropped instead (by removing their
	  might-have-been-fec field).
	  Spurious lost events coming out of the depayloader can cause the
	  decoder to stop decoding until the next keyframe and / or request a new
	  keyframe, and while this is not desirable it makes sense to forward
	  that information when we have other means to determine whether a lost
	  packet was indeed a FEC packet, as is the case with VP8 / VP9 payloads
	  when they carry a picture id.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>

2020-10-20 23:22:36 +1100  Jan Schmidt <jan@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Preserve SPS/PPS arrival order.
	  Even if SPS/PPS haven't changed, make sure to move them to the
	  end of the tracking array if needed, so we always know what the
	  most recent entries are, in case we need to discard the oldest
	  when generating codec_data.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>

2020-10-17 00:05:15 +1100  Jan Schmidt <jan@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Warn when max SPS/PPS are collected in AVC mode.
	  The AVC codec_data has a flaw that it can only accomodate
	  31 SPS headers, even though H.264 can have 32, and 255 PPS,
	  when there can be 256 in H.264. When streaming RTP some
	  clients like to cycle through SPS/PPS ids when changing
	  configuration and can eventually accumulate a full set.
	  In that case, we have no choice but to discard one (oldest)
	  entry, or else the count written into the codec_data is wrong
	  and downstream decoding failures ensue.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>

2020-10-28 00:29:05 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtptimerqueue.c:
	* gst/rtpmanager/rtptimerqueue.h:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/rtptimerqueue.c:
	  rtpjitterbuffer: don't send multiple instant RTX for the same packet
	  Due to us not properly acknowleding the time when the last RTX was sent
	  when scheduling a new one, it can easily happen that due to the packet
	  you are requesting have a PTS that is slightly old (but not too old when
	  adding the latency of the jitterbuffer), both its calculated second and
	  third (etc.) timeout could already have passed. This would lead to a burst
	  of RTX requests, which acts completely against its purpose, potentially
	  spending a lot more bandwidth than needed.
	  This has been properly reproduced in the test:
	  test_rtx_not_bursting_requests
	  The good news is that slightly re-thinking the logic concerning
	  re-requesting RTX, made it a lot simpler to understand, and allows us
	  to remove two members of the RtpTimer which no longer serves any purpose
	  due to the refactoring. If desirable the whole "delay" concept can actually
	  be removed completely from the timers, and simply just added to the timeout
	  by the caller of the API. But that can be a change for a another time.
	  The only external change (other than the improved behavior around bursting
	  RTX) is that the "delay" field now stricly represents the delay between
	  the PTS of the RTX-requested packet and the time it is requested on,
	  whereas before this calculation was more about the theoretical calculated
	  delay. This is visible in three other RTX-tests where the delay had
	  to be adjusted slightly. I am confident however that this change is
	  correct.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/789>

2020-10-27 23:43:49 +1100  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Fix sparse stream crash
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656
	  introduced an invalid memory access when debug is enabled, by casting
	  the wrong pointer to a GstCollectPad. Fixing that showed the original
	  change was incorrect and leads to an infinite loop in the
	  testsuite. This patch fixes both problems.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/788>

2020-10-22 15:29:01 -0300  Thibault Saunier <tsaunier@igalia.com>

	* ext/vpx/gstvpxenc.c:
	  vpx: Fix the check to unfixed/unknown framerate to set bitrate
	  0/1 means unknown framerate not X/0 (which is illegal).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/782>

2020-10-22 09:17:26 -0400  Arun Raghavan <arun@asymptotic.io>

	* gst/rtp/gstrtputils.c:
	  rtputils: Count metas with an empty tag list for copying/keeping
	  The GstMetaInfos registered in core do not set their tags to NULL, but
	  instead use an empty list (non-NULL list with a single NULL value).
	  Let's check explicitly for that so as to not miss some metas.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/779>

2020-10-16 16:05:45 -0700  Bastien Reboulet <bastien.reboulet@gmail.com>

	* ext/qt/qtitem.cc:
	  qmlglsink: fix crash when created/destroyed in quick succession
	  The crash is caused by a race condition where the render thread
	  calls a method on the QtGLVideoItem instance that was
	  previously destroyed by the main thread.
	  Also, less frequently, QtGLVideoItem::onSceneGraphInitialized
	  is called when QQuickItem::window is null, also causing a crash.
	  Fixes #798
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/777>

2020-10-19 18:23:25 +0300  Sebastian Dröge <sebastian@centricular.com>

	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2codec: Garbage collect old frames if they accumulate because of codec bugs
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>

2020-10-19 17:56:04 +0300  Sebastian Dröge <sebastian@centricular.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2codec: Pass system frame number as timestamp and use it to retrieve back frames reliably
	  System frame numbers are supposed to be unique and correct drivers are
	  passing through timestamps without modification from the output/sink to the
	  capture/src side.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>

2020-09-24 13:13:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: Add clear-ssrc action
	  This action signal will delegate to clear-ssrc onto the rtpssrcdemux element
	  associated with the session. This allow rtpbin users to clear pads and
	  elements for a specific ssrc that is known to no longer be in use. This
	  happens when a pad is reused in rtpsrc or ristsrc.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/736>

2017-09-08 20:02:13 +0100  John-Mark Bell <jmb@pexip.com>

	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp8pay.h:
	* tests/check/elements/rtpvp8.c:
	* tests/check/meson.build:
	  rtpvp8pay: payload temporally scaled bitstreams.
	  Co-Authored-By: Vincent Sanders <vince@pexip.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>

2017-11-17 15:11:41 +0100  Stian Selnes <stian@pexip.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp8pay.h:
	  rtpvp8pay: Add picture-id-offset property
	  Add property to set the initial value for picture-id. RFC7741 says
	  that picture-id MAY be initialized to a random value, thus it's also
	  valid to simply set it to a fixed initial value. A fixed value is very
	  useful for testing.
	  Default behavior is not changed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>

2017-03-16 15:23:28 +0100  Mikhail Fludkov <misha@pexip.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: move duplicate code to separate functions
	  Two new functions to modify picture id:
	  gst_rtp_vp8_pay_picture_id_reset - picks random picture id of
	  appropriate bitsize
	  gst_rtp_vp8_pay_picture_id_increment - increments picture id taking
	  care of wrapping
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>

2017-09-08 08:13:05 +0100  John-Mark Bell <jmb@pexip.com>

	* docs/gst_plugins_cache.json:
	* ext/vpx/gstvpxenc.c:
	  vp8enc: expect bps for temporal-scalability-target-bitrate.
	  Consistency with target-bitrate is less surprising and with
	  modern libvpx additional configuration is required to make
	  temporal scaling work.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>

2017-09-08 08:19:20 +0100  John-Mark Bell <jmb@pexip.com>

	  vp8enc: finish support for temporally scaled encoding
	  - introduce two new properties:
	  * temporal-scalability-layer-flags:
	  Provide fine-grained control of layer encoding to the
	  outside world. The flags sequence should be a multiple of
	  the periodicity and is indexed by a running count of encoded
	  frames modulo the sequence length.
	  * temporal-scalability-layer-sync-flags:
	  Specify the pattern of inter-layer synchronisation (i.e.
	  which of the frames generated by the layer encoding
	  specification represent an inter-layer synchronisation).
	  There must be one entry per entry in
	  temporal-scalability-layer-flags.
	  - apply temporal scalability settings and expose as buffer
	  metadata.
	  This allows the codec to allocate a given frame to the correct
	  internal bitrate allocator. Additionally, all the
	  non-bitstream metadata needed to payload a temporally scaled
	  stream is now attached to each output buffer as a
	  GstVideoVP8Meta.
	  - add unit test for temporally scaled encoding.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>

2020-10-15 18:21:54 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* gst/isomp4/qtdemux.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/udp/gstudp.c:
	* meson.build:
	  meson: update glib minimum version to 2.56
	  In order to support the symbol g_enum_to_string in various
	  project using GStreamer ( gst-validate etc.), the glib minimum
	  version should be 2.56.0.
	  Remove compat code as glib requirement
	  is now > 2.56
	  Version used by Ubuntu 18.04 LTS
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/774>

2020-10-14 14:30:34 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
	  rtpst2022-1-fecenc: fix input seqnum check
	  We need to cast the incremented last seqnum to guint16 for
	  consistent checks on wraparound
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/770>

2020-09-12 09:02:30 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Correct time types
	  - last_dts is in milliseconds, not nanoseconds as expected for
	  GstClockTime. Make it a generic guint64.
	  - Use GstClockTime for the fields that actually contain nanoseconds.
	  None of them should become negative.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/766>

2020-10-09 09:31:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
	  rtpst2022-1-fecenc: Don't unconditionally use GLib 2.60 APIs
	  g_queue_clear_full() in this case.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/768>

2020-10-08 18:54:55 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/rtpulpfeccommon.c:
	  rtpulpfec: fix potential alignment issue in xor function
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753#note_646453
	  for context
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>

2020-10-06 03:03:13 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
	* gst/rtpmanager/gstrtpst2022-1-fecenc.h:
	* gst/rtpmanager/meson.build:
	* tests/check/elements/rtpst2022-1-fecenc.c:
	* tests/check/meson.build:
	  rtpmanager: implement SMPTE 2022-1 FEC encoder
	  + improve integration of FEC encoders in rtpbin
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>

2020-10-06 03:13:30 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
	* gst/rtpmanager/gstrtpst2022-1-fecdec.h:
	* gst/rtpmanager/meson.build:
	* tests/check/elements/rtpst2022-1-fecdec.c:
	* tests/check/meson.build:
	  rtpmanager: implement SMPTE 2022-1 FEC decoder
	  + improve integration of FEC decoders in rtpbin
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>

2020-07-08 17:28:31 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	* tests/check/elements/rtpfunnel.c:
	  rtpfunnel: Also forward custom sticky event
	  This is useful to track metadata about each group of packets
	  Also include a unit test
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/666>

2020-09-29 09:44:54 -0300  Thibault Saunier <tsaunier@igalia.com>

	* docs/gst_plugins_cache.json:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  isomp4: Rename GstQTMux to GstBaseQTMux to avoid breaking API
	  Since 52b63de19ada283c1180c8fc00cacb1465fdf10f the qtmux GType was
	  renamed GstQTMuxElement which breaks presets, revert that change.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/755>

2020-09-28 18:25:21 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpklvpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp9pay.c:
	  rtp: Fix allocations to support source-info property
	  Use gst_rtp_base_payload_allocate_output_buffer() instead of
	  gst_rtp_buffer_new_allocate() in order to allocate RTP buffer with
	  correct number of CSRCs according to the meta.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>

2015-10-23 11:08:56 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: Fix allocation to support source-info property
	  Use gst_rtp_base_payload_allocate_output_buffer() in order to allocate
	  RTP buffer with correct number of CSRCs according to the meta.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/314
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>

2020-09-28 15:36:00 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: output the correct limits in error messages
	  Having the current bytes being less than the limit was confusing!
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>

2020-07-31 16:47:37 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* tests/check/elements/qtmux.c:
	  qtmux: properly support initial caps nego failure
	  Scenario:
	  - gap event causes h264parse to push made up caps that may fail checks
	  inside qtmux (e.g missing codec_data).
	  - the caps event has already been marked as received and is sticky on
	  the sink pad
	  - gst_qt_mux_pad_can_renegotiate() will retrieve the failed caps event
	  using gst_pad_get_current_caps() and reject the correct updated caps
	  with codec_data.
	  - Failure!
	  Keep track of the configured caps ourselves instead of relying on the
	  sticky event on the pad.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>

2020-07-22 15:34:44 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: support non-seekable downstream mode
	  Write an mdat per buffer in that case.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>

2020-09-23 15:25:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Remove the rtpjitterbuffer with the stream
	  Since !348, the jitterbuffer was only removed with the session. This restores
	  the original behaviour and removes the jitterbuffer when the stream is
	  removed. This avoid accumulating jitterbuffer objects into the bin when a
	  session is reused.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>

2020-09-23 13:26:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Cleanup dead code
	  The rtpjitterbuffer is now part of the session elements, we no longer need
	  to do the ref_sink dance when signalling it. It is already owned by the bin
	  when signalled. Also, the code that handles generic session elements already
	  handle the ref_sink() calls since:
	  03dc22951bacb6fdc3868c8f801e6a52c33a745f
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>

2020-09-18 16:09:20 +1000  Matthew Waters <matthew@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	* tests/check/elements/rtph264.c:
	  rtph26*depay: drop FU's without a corresponding start bit
	  If we have not received a FU with a start bit set, any subsequent FU
	  data is not useful at all and would result in an invalid stream.
	  This case is constructed from multiple requirements in
	  RFC 3984 Section 5.8 and RFC 7798 Section 4.4.3.  Following are excerpts
	  from RFC 3984 but RFC 7798 contains similar language.
	  The FU in a single FU case is forbidden:
	  A fragmented NAL unit MUST NOT be transmitted in one FU; i.e., the
	  Start bit and End bit MUST NOT both be set to one in the same FU
	  header.
	  and dropping is possible:
	  If a fragmentation unit is lost, the receiver SHOULD discard all
	  following fragmentation units in transmission order corresponding to
	  the same fragmented NAL unit.
	  The jump in seqnum case is supported by this from the specification
	  instead of implementing the forbidden_zero_bit mangling:
	  If a fragmentation unit is lost, the receiver SHOULD discard all
	  following fragmentation units in transmission order corresponding to
	  the same fragmented NAL unit.
	  A receiver in an endpoint or in a MANE MAY aggregate the first n-1
	  fragments of a NAL unit to an (incomplete) NAL unit, even if fragment
	  n of that NAL unit is not received.  In this case, the
	  forbidden_zero_bit of the NAL unit MUST be set to one to indicate a
	  syntax violation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/730>

2020-09-20 21:06:19 +0900  Seungha Yang <seungha@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Response caps query from srcpad
	  ... and chain up to default query handler for unhandled query types.
	  Unhandled query shouldn't be returned with FALSE if there's no special needs.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/731>

2020-09-16 12:15:09 +1000  Matthew Waters <matthew@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux-doc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: make documentation happy
	  introduce a base qtmux class that we can install documentation snippets
	  on instead of duplicating across alll the isomp4 elements
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>

2020-05-28 19:40:24 +1000  Matthew Waters <matthew@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/isomp4/gstqtmuxmap.c:
	* tests/check/elements/qtmux.c:
	  isomp4/mux: add a fragment mode for initial moov with data
	  Used by some proprietary software for their fragmented files.
	  Adds some support for multi-stream fragmented files
	  Flow is as follows.
	  1. The first 'fragment' is written as a self-contained fragmented
	  mdat+moov complete with an edit list and durations, tags, etc.
	  2. Subsequent fragments are written with a mdat+moof and each stream is
	  interleaved as data arrives (currently ignoring the interleave-*
	  properties).  data-offsets in both the traf and the trun ensure
	  data is read from the correct place on demuxing.  Data/chunk offsets
	  are also kept for writing out the final moov.
	  3. On finalisation, the initial moov is invalidated to a hoov and the
	  size of the first mdat is extended to cover the entire file contents.
	  Then a moov is written as regularly would in moov-at-end mode (the
	  default).
	  This results in a file that is playable throughout while leaving a
	  finalised file on completion for players that do not understand
	  fragmented mp4.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>

2020-06-25 16:37:56 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: increase some logging on streams and sample parsing
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>

2020-06-25 16:35:45 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: bail out when encountering an atom with a size of 0
	  A size 0 atom means the atom extends to the end of the file.  No further
	  valid atoms will ever follow.  Avoids a subsequent scan for an atom from
	  one byte earlier after encountering a size 0 atom.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>

2020-06-25 16:33:04 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix subsequent moof parsing after moov with valid samples
	  reset the moof_offset back to its original value like is done in the
	  error case just before.
	  Fixes subsequent parsing of a moof following a moov that contains valid
	  samples in a non-streaming fragmented mp4.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>

2020-06-25 16:30:28 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: extend edit list when fragmented
	  When we are fragmented, the edit list may only refer to the portion of
	  the media that is in the moov.  Extend the edit list stop time when we
	  if there is only one qt segment and we are reading a fragmented file.
	  Fixes playback of some fragmented mp4 files generated by proprietary
	  programs.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>

2020-09-15 14:22:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* meson_options.txt:
	  meson: Allow overriding qt5 feature
	  This will allow controlling that feature from gst-build
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/729>

2015-11-17 19:14:01 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Implement segment query
	  Fixes #239
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/713>

2020-09-14 10:15:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: Allow lower-case "aac-hbr" instead of correct "AAC-hbr"
	  Various live555 based products are using the wrong "mode" string or
	  seem to assume case-insensitive matching, which is wrong.
	  Examples for this are the Yuan SC6C0N1 mini and the Kiloview E2.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/727>

2020-05-02 02:21:00 +0200  Stefan Brüns <stefan.bruens@rwth-aachen.de>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add support for AAX encrypted audio streams
	  This is modelled after the DASH Common Encryption scheme, but is somewhat
	  simpler as more parts are fixed, i.e. just one encryption scheme.
	  The output caps are fixed to 'application/x-aavd'. All information
	  required for decryption are part of the 'adrm' atom, which is passed
	  on as a property. The property is attached to the buffer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>

2020-05-02 02:20:44 +0200  Stefan Brüns <stefan.bruens@rwth-aachen.de>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add 'aavd' and related fourcc codes for AAX encrypted audio
	  The 'aavd' box is contained in the 'stsd' sample description. The 'aavd'
	  box follows the layout of an 'mp4a' entry, i.e. it contains a single
	  standard 'esds' extension box, and the two proprietary 'adrm' and 'aabd'
	  extension boxes.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>

2014-06-23 08:46:37 +0200  Haakon Sporsheim <haakon@pexip.com>

	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxdec.h:
	  vpxdec: request a sync point on decoder errors
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/725>

2020-09-13 18:31:57 +0200  Camilo Celis Guzman <camilo@pexip.com>

	* gst/rtp/gstrtpvrawpay.c:
	  rtp/vrawpay: use alloc_output_buffer from base class
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/726>

2020-09-07 23:20:58 +0800  Ricky Tang <ricky@deepsentinel.com>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix push-backchannel-buffer parameter mismatch
	  When using python, signal parameter must match with function.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/724>

2020-09-10 11:24:32 +0200  Jérôme Laheurte <jlaheurte@quividi.net>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: check buffer size before dereferencing. Fixes #541
	  Some cameras (Panacast) have buggy drivers/firmware which send
	  invalid JPEG frames, containing no data, which makes jpegdec
	  crash because it assumes the frame is at least 2 bytes long.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/723>

2020-09-10 11:11:00 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Improve logging of gst_flv_mux_buffer_to_tag_internal
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>

2020-09-09 15:12:53 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer
	  Besides looking like the correct place to put this, it allows us to drop
	  the entire aggregator queue. The old implementation only dropped at most
	  one buffer for each call of aggregate.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>

2020-09-08 17:35:50 +0200  Havard Graff <havard@pexip.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: plug memory-leak
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/718>

2020-08-28 18:09:15 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	* ext/vpx/meson.build:
	  vp9enc: expose row-mt property
	  With recent libvpx versions, multithreading can be enabled on
	  a per-tile basis, instead of on a per tile-column basis.
	  In combination with the new tile-rows property, this allows the
	  encoder to make much better use of the available CPU power.
	  Bump minimum libvpx version to 1.7.0
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>

2020-08-28 17:45:48 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* ext/vpx/gstvpxenc.c:
	  vpxenc: change default for deadline to good quality
	  Having the deadline set to best quality causes the encoder
	  to be absurdly slow, most real-life users will want the good
	  quality tradeoff instead.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>

2020-08-28 17:39:47 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	  vp9enc: expose tile-columns and tile-rows properties
	  Based on patch by Stian Selnes <stian@pexip.com>.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>

2020-08-28 17:35:26 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/vpx/gstvpxenc.c:
	* ext/vpx/gstvpxenc.h:
	  vpxenc: add configure_encoder virtual method
	  For subclasses to expose format-specific properties
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>

2020-09-08 20:57:33 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: fix sink pad release while PLAYING
	  - Release the split mux lock while removing the probes
	  - Flush the sinkpad to unblock other pads
	  - Turn check_completed_gop into a do while statement, when
	  waking up we want to recheck whether the current GOP is
	  ready for sending
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/719>

2017-10-31 09:40:33 +0000  John-Mark Bell <jmb@pexip.com>

	* tests/check/elements/vp8enc.c:
	  vp8enc: improve unit tests
	  - make test_encode_simple cope with libvpx built with
	  CONFIG_REALTIME_ONLY. Sadly, there's no way to detect this at
	  runtime beyond trying to set lag-in-frames to >0, pushing a
	  buffer and catching the GST_FLOW_NOT_NEGOTIATED return.
	  - fix bitrot in test_encode_simple_when_bitrate_set_to_zero.
	  - port test_encode_simple to GstHarness and introduce a separate
	  test for the lag-in-frames property.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/708>

2020-08-21 16:03:09 +0200  Jakub Adam <jakub.adam@collabora.com>

	* docs/gst_plugins_cache.json:
	  docs: Update plugin cache
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>

2020-03-24 19:35:07 +0100  Jakub Adam <jakub.adam@collabora.com>

	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvpxenc.c:
	  vpx: Support GST_VIDEO_FORMAT_I422_10LE
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>

2020-03-24 17:16:59 +0100  Jakub Adam <jakub.adam@collabora.com>

	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvpxenc.c:
	  vpx: Support GST_VIDEO_FORMAT_I420_10LE
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>

2020-03-23 21:44:30 +0100  Jakub Adam <jakub.adam@collabora.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvpxenc.c:
	  vp9enc: support GST_VIDEO_FORMAT_Y444
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>

2020-09-08 17:30:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* .gitlab-ci.yml:
	  ci: include template from gst-ci master branch again

2020-09-08 16:58:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* meson.build:
	  Back to development

=== release 1.18.0 ===

2020-09-08 00:05:14 +0100  Tim-Philipp Müller <tim@centricular.com>

	* .gitlab-ci.yml:
	* ChangeLog:
	* NEWS:
	* README:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.18.0

2020-09-07 22:39:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* scripts/dist-translations.py:
	* scripts/meson.build:
	  meson: dist pot file in tarballs
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/716>

2020-09-07 12:13:18 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  gst: Update for gst_video_transfer_function_*() function renaming
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/715>

2020-08-31 15:01:32 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Avoid crash when best pad gets flushed
	  The 'best' pad might receive a flush event between us picking it and us
	  popping the buffer. In this case, the buffer will be missing.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/711>

2020-08-31 13:43:42 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Correct breaks in gst_flv_mux_find_best_pad
	  The code seems to use `continue` and `break` as if both refer to the
	  surrounding `while` loop. But because `break` breaks out of the
	  `switch`, they actually have the same effect.
	  This may have caused the loop not to terminate when it should. E.g. when
	  `skip_backwards_streams` drops a buffer we should abort the aggregation
	  and wait for all pads to be filled again. Instead, we might have just
	  selected a subsequent pad as our new "best".
	  Replace `break` with `done = TRUE; break`, and `continue` with `break`.
	  Then simplify the code a bit.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/710>

2020-05-13 11:31:38 +0200  Dmitriy Purgin <dpurgin@gmail.com>

	* ext/qt/README.md:
	* ext/qt/qtplugin.pro:
	  gstqmlgl: build on Windows with qmake without pkgconfig; update instructions on building for Windows
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/591>

2020-08-21 12:12:48 +0200  Philipp Zabel <philipp.zabel@gmail.com>

	* meson.build:
	  meson: fix build failure if orc is enabled but none of its users are
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/778
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/704>

2020-08-20 14:26:04 +0200  Zeid Bekli <zeidb@axis.com>

	* gst/rtp/gstrtpL16depay.c:
	  rtpL16depay: unref buffer on error
	  gst_rtp_L16_depay_process to unref buffer on wrong payload size or
	  reorder failure.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/702>

=== release 1.17.90 ===

2020-08-20 16:11:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.17.90

2020-08-18 10:27:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtputils.c:
	  rtputils: Don't call NULL GstMeta transform function
	  It's optional and if it does not exist then no transformation is
	  possible.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/701>

2020-08-13 15:27:25 -0400  Julian Bouzas <julian.bouzas@collabora.com>

	* gst/rtp/gstrtp.c:
	  rtp: Do not register rtpreddec and rtpredenc twice
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/699>

2020-08-12 12:21:43 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: Improve readability of "stats" docs by making the fields an actual list
	  Otherwise they end up all in the same line one after another.
	  Also add docs for the "avg-jitter" stats field of the jitterbuffer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/698>

2020-08-11 17:24:11 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2h264codec.c:
	  v4l2h264codec: Map newly defined profile/levels
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/697>

2020-08-11 17:18:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/types-compat.h:
	* sys/v4l2/ext/v4l2-common.h:
	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	  v4l2: Sync headers with kernel 5.9
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/697>

2020-08-06 13:15:10 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: use GstV4l2Error in gst_v4l2_open()
	  gst_v4l2_open() is called by gst_v4l2_device_provider_probe_device(),
	  where the GstV4l2Object is created without an associated GstElement.
	  If gst_v4l2_open() fails, it raises a bus message, but without an
	  element, a precondition check fails on
	  gst_element_message_full_with_details() generating a crash if running
	  with fatal-warnings debug mode.
	  GstV4l2Error is a helper to raise error bus messages when it is
	  appropiated. This patch changes the direct bus messages to this
	  helper, and the elements will actually send the error message.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/694>

2020-08-10 20:20:53 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/flv/gstflvmux.c:
	  flvmux: Return NEED_DATA when we drop a buffer
	  When we are dropping a buffer in find_best_pad (e.g. waiting for a
	  keyframe, or skipping backwards timestamp), return
	  GST_AGGREGATOR_FLOW_NEED_DATA to make sure we have enough data at the
	  next run. Otherwise, a stream that accidentally fell behind (e.g.
	  relinking race, or just waiting for a keyframe) will never get the
	  opportunity to catch up to the other one, because the other one will
	  always keep advancing.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>

2020-08-10 20:20:04 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/flv/gstflvmux.c:
	  flvmux: Return NEED_DATA when no best pad is found
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>

2020-08-10 20:17:38 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix possible crash on GST_ITERATOR_RESYNC
	  Wrong pointer type
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>

2020-08-10 15:49:55 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtglrenderer.cc:
	  qmlgloverlay: fix multiple elements with Qt 5.15
	  With Qt 5.15 multiple qmlgloverlay elements would produce:
	  ASSERT: "!m_gl->property(QSG_RENDERCONTEXT_PROPERTY).isValid()" in file /path/to/qt5/qtdeclarative/src/quick/scenegraph/qsgdefaultrendercontext.cpp, line 121
	  Workaround by setting the (seeminigly unused) property before
	  initialization.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/695>

2020-08-05 10:41:33 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtph264.c:
	* tests/check/elements/rtph265.c:
	  rtph26[45]pay: Change default aggregate-mode to "none" for backwards compatibility
	  We didn't aggregate at all in previous versions and there are apparently
	  various RTP implementations that don't handle aggregation well at all.
	  As part of this also document that for RTSP it is recommended to keep it
	  set to "none" while for WebRTC it should be set to "zero-latency".
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/692>

2020-07-24 16:58:34 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/meson.build:
	* ext/qt/meson.build:
	* meson.build:
	* tests/examples/gtk/meson.build:
	  build: update for gl pkg-config file split
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/680>

2020-07-31 13:50:13 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Make sure flushing doesn't block
	  * Trying to disconnect a stream from a running splitmuxsink by flushing
	  it results in the FLUSH_START blocking in the stream queue's
	  gst_pad_pause_task because the flush did not unblock
	  complete_or_wait_on_out, so add a check for ctx->flushing there.
	  * Add a GST_SPLITMUX_BROADCAST_INPUT so check_completed_gop notices
	  flushing changed and the incoming push is unblocked.
	  * Pass the FLUSH_STOP along to the muxer without waiting.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/687>

2020-08-04 15:49:43 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Wait until we have a clock
	  Otherwise it can happen that it tries to get the clock in PAUSED state
	  in live mode, which does not exist.
	  Thanks to Sebastian Dröge for helping debugging.
	  Fixes #775
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/691>

2020-07-31 11:05:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: extract bit depth from codec data for ALAC
	  The info in the sound sample description might not be
	  accurate if it's an older version atom.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/771
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/686>

2020-07-28 18:46:30 +0300  Jordan Petridis <jordan@centricular.com>

	* gst/auparse/gstauparse.c:
	  auparse: fix compiler warnings
	  GCC 10 was complaining like following. It really is complaining about default cases returning
	  with potentially unitialized *desval, but those cases in the switch should never be hit.
	  ```
	  ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c: In function 'gst_au_parse_chain':
	  ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:481:37: error: 'timestamp' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  481 |       GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
	  ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:482:36: error: 'duration' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  482 |       GST_BUFFER_DURATION (outbuf) = duration;
	  ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:480:34: error: 'offset' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  480 |       GST_BUFFER_OFFSET (outbuf) = offset;
	  cc1: all warnings being treated as errors
	  ```
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/671>

2020-07-29 14:06:55 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: drop stream-start message posted by the internal udp sink(s)
	  See #1368
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/685>

2020-07-22 16:24:15 +0900  Hosang Lee <hosang10.lee@lge.com>

	* tests/check/elements/qtdemux.c:
	  tests: qtdemux: test correct pad names are created
	  Test correct pad names are created in accordance to their media type
	  in mss mode.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/628>

2020-06-16 17:23:44 +0900  Hosang Lee <hosang10.lee@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: create correct pad names in encrypted streams
	  Refer to "original-media-type" when setting stream's subtype
	  for encrypted streams in mss mode.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/628>

2020-07-22 14:31:13 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Do caps renegotiation when it only adds fields
	  Matroskamux can accept caps renegotiation if the new caps is a
	  superset of the old one, meaning upstream added new info to
	  the caps.
	  Same logic as a5f22f03aa25b04726f78ae619f40b3b648f7d48 in qtmux.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/678>

2020-07-24 14:02:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	  rtpfunnel: protect internal srccaps with lock
	  These are modified from sink pad event handlers, so
	  could be accessed from multiple threads at the same
	  time.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/681>

2020-02-23 23:44:16 +0100  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	  rtpfunnel: copy caps before sending them in a caps-event
	  Reason being we don't want downstream to own a ref to our
	  internal caps.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/681>

2020-07-27 15:41:26 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: fix various documentation issues
	  Improper naming of properties, improper links, misc
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/684>

2020-07-24 17:13:04 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: hypothetical fix for data pointer calculation
	  mmal buffer header docs say data is valid for length bytes
	  from offset. In practice offset always seems to be 0 so
	  far though.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>

2020-07-24 16:35:43 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: mark buffers as header and keyframe/delta-unit
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>

2020-07-24 16:14:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: fix nal alignment of output buffers
	  We claim output buffers are nal-aligned, but that wasn't
	  actually true: We would push out a partial nal in case
	  the nal doesn't fit into the max encoder-selected output
	  buffer size, and then the next buffer would not start
	  with a sync marker. That's not right and makes h264parse
	  unhappy.
	  Instead accumulate buffers until we have a full frame
	  (we can't rely on the NAL_END flag, it's always set).
	  Fixes #768
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>

2020-07-13 23:43:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/meson.build:
	  rpicamsrc: fix "Could not find component vc.ril.camera" on recent raspios
	  Make extra sure all the required mmal libs such as libmmal_vc_client.so
	  actually get linked and stay linked. Otherwise the above error happens
	  it seems.
	  buster (10.4) with meson 0.55 and pi ref 2020-05-27
	  pi-gen, 825107f04027269db77426046f5085475b1ea22f, stage5
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>

2020-07-13 17:01:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* po/POTFILES:
	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.c:
	  rpicamsrc: deviceprovider: hook up i18n properly
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>

2020-07-13 16:55:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.c:
	  rpicamsrc: deviceprovider: advertise (M)JPEG as well
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>

2020-07-13 16:50:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.c:
	  rpicamsrc: deviceprovider: also advertise constrained-baseline profile
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>

2020-07-23 16:58:00 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* meson.build:
	  meson: add a plugin summary
	  This summary displays a list of plugins which
	  have been enabled.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/679>

2020-07-22 09:46:47 +0800  Haihua Hu <jared.hu@nxp.com>

	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: enhance v4l2 control interface to support string type CID
	  add string type cid support for v4l2 implementation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/676>

2020-07-01 15:17:47 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: add  Dolby Vision fourcc
	  This identifiers are registered in the MPEG-RA and defined
	  to be used by the Dolby Vision AVC/HEVC streams.
	  This is a first step to present the stream to the decoder.
	  Additional box parsing of DOVIConfigurationBox is necessary
	  to complete the media presentation with proper Dolby Vision
	  enhancements.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/658>

2020-05-17 15:51:09 +1000  Luke Yelavich <themuso@themuso.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Copy GstCapsFeatures to caps for source pad
	  Allows using imagefreeze with buffers in GLMemory. The following pipeline
	  works.
	  gst-launch-1.0 filesrc location=image.jpg ! jpegdec ! glupload ! \
	  imagefreeze ! glcolorconvert ! glimagesinkelement
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/594>

2020-07-20 18:20:59 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/rtptwcc.h:
	  rtpmanager: fix "redefinition of typedef RTPTWCCManager" compiler warning
	  G_DECLARE_FINAL_TYPE includes this typedef as well.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/675>

2020-07-17 16:39:25 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp*pay: Allocate using the base class for audio codecs
	  This is required to add RTP header extensions from the
	  meta automatically.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>

2020-07-14 13:14:09 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix segfault with illegal free
	  set_get_param_q is not a pointer so it is illegal to call g_queue_free_full().
	  Freeing the requests by popping them from the queue instead.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/669>

2020-07-15 14:40:42 +0300  Raul Tambre <raul@tambre.ee>

	* ext/qt/qtitem.cc:
	  QtGLVideoItem: Use QSharedPointer::data() for better compatibility
	  Older Qt versions didn't have QSharedPointer::get(), which is just a modern alias for QSharedPointer::data().
	  FAILED: ext/qt/libgstqmlgl.so.p/qtitem.cc.o
	  c++ -Iext/qt/libgstqmlgl.so.p -Iext/qt -I../ext/qt -I. -I.. -I../gst-libs -I/usr/include/glib-2.0 -I/usr/lib/aarch64-linux-gnu/glib-2.0/include -I/usr/include/gstreamer-1.0 -I/usr/include/orc-0.4 -I/usr/lib/aarch64-linux-gnu/gstreamer-1.0/include -I/usr/include/aarch64-linux-gnu/qt5/QtCore -I/usr/include/aarch64-linux-gnu/qt5 -I/usr/include/aarch64-linux-gnu/qt5/QtGui -I/usr/include/aarch64-linux-gnu/qt5/QtQml -I/usr/include/aarch64-linux-gnu/qt5/QtNetwork -I/usr/include/aarch64-linux-gnu/qt5/QtQuick -I/usr/include/aarch64-linux-gnu/qt5/QtX11Extras -I/usr/include/libdrm -flto -fdiagnostics-color=always -pipe -D_FILE_OFFSET_BITS=64 -std=c++11 -Wmissing-declarations -Wredundant-decls -Wwrite-strings -Winit-self -Wmissing-include-dirs -Wno-multichar -Wvla -Wpointer-arith -g -fdebug-prefix-map=/opt/good/src=. -Wformat -Werror=format-security -O3 -march=native -Wno-error -Wdate-time -fPIC -pthread -DHAVE_CONFIG_H -DHAVE_QT_X11 -DHAVE_QT_EGLFS -MD -MQ ext/qt/libgstqmlgl.so.p/qtitem.cc.o -MF ext/qt/libgstqmlgl.so.p/qtitem.cc.o.d -o ext/qt/libgstqmlgl.so.p/qtitem.cc.o -c ../ext/qt/qtitem.cc
	  In file included from /usr/include/gstreamer-1.0/gst/gst.h:55:0,
	  from /usr/include/gstreamer-1.0/gst/video/video.h:23,
	  from ../ext/qt/qtitem.cc:27:
	  ../ext/qt/qtitem.cc: In destructor ‘virtual QtGLVideoItem::~QtGLVideoItem()’:
	  ../ext/qt/qtitem.cc:138:86: error: ‘class QSharedPointer<QtGLVideoItemInterface>’ has no member named ‘get’
	  GST_INFO ("%p Destroying QtGLVideoItem and invalidating the proxy %p", this, proxy.get());
	  ^
	  /usr/include/gstreamer-1.0/gst/gstinfo.h:682:31: note: in definition of macro ‘GST_CAT_LEVEL_LOG’
	  (GObject *) (object), __VA_ARGS__);    \
	  ^~~~~~~~~~~
	  ../ext/qt/qtitem.cc:138:3: note: in expansion of macro ‘GST_INFO’
	  GST_INFO ("%p Destroying QtGLVideoItem and invalidating the proxy %p", this, proxy.get());
	  ^
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/673>

2020-07-14 14:24:20 +0100  Justin Chadwell <justin.chadwell@pexip.com>

	* gst/isomp4/qtdemux.c:
	* tests/check/elements/qtdemux.c:
	  qtdemux: fix allocation explosion with stsd entries
	  Previously, the user input for stsd entries is trusted completely, and
	  so a maliciously crafted file could choose the length of the stsd
	  entries arbitrarily and cause qtdemux to try to allocate up to 2GB of
	  memory (half of a 32 bit max int).
	  This patch fixes this by sanity checking the stsd input against the
	  size of the entire stsd atom.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/670>

2020-07-13 10:37:19 +0100  Justin Chadwell <justin.chadwell@pexip.com>

	* gst/isomp4/qtdemux.c:
	* tests/check/elements/qtdemux.c:
	  qtdemux: fix crashes when input stream contained no stsd entries
	  During trak parsing, we need to check for the existence of stsd_entries,
	  otherwise, we end up with a NULL pointer to them. It is entirely
	  possible for the stsd to exist, but for it to have no entries, which the
	  previous checks did not take into account.
	  This patch adds a simply check to ensure that all files that do not
	  contain a stsd entry are deemed corrupt, and adds a test case to prevent
	  a regression.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/670>

2020-07-15 12:40:17 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	  docs: update for new pixel formats
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/753
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/754
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/672>

2020-07-10 21:43:14 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/meson.build:
	  rpicamsrc: fix build with older meson versions
	  assert() used to require two arguments.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/668>

2020-07-10 13:08:55 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/meson.build:
	* tests/examples/rpicamsrc/meson.build:
	* tests/examples/rpicamsrc/test_color_balance.c:
	* tests/examples/rpicamsrc/test_orientation.c:
	  examples: hook up rpicamsrc examples
	  webrtc one should probably go into gst-examples.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-07-10 00:42:13 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/rpicamsrc/test_color_balance.c:
	* tests/examples/rpicamsrc/test_orientation.c:
	* tests/examples/rpicamsrc/webrtc-unidirectional-h264.c:
	  examples: fix indentation of rpicamsrc examples
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-07-09 19:08:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* docs/meson.build:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: flesh out docs and add to plugin docs cache
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-07-09 18:04:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: enable video orientation/direction unconditionally
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-07-09 17:37:01 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstrpicam-enums-template.c:
	* sys/rpicamsrc/gstrpicam-enums-template.h:
	  rpicamsrc: remove mkenums template files which are no longer needed
	  They were still being used by the autotools build, but that's gone.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-07-09 17:35:15 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/RaspiCLI.c:
	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiPreview.c:
	* sys/rpicamsrc/RaspiStill.c:
	* sys/rpicamsrc/RaspiStillYUV.c:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.c:
	  rpicamsrc: fix indentation
	  Not touching the Raspi* files.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-07-09 17:31:49 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/meson.build:
	  rpicamsrc: fix and silence some compiler warnings
	  Some are in system headers, and in Raspi files we want
	  to keep modifications to a minimum.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-07-09 16:07:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson_options.txt:
	* sys/meson.build:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.c:
	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.h:
	* sys/rpicamsrc/meson.build:
	  rpicamsrc: hook up to build
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-07-09 11:46:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	  Merge branch 'plugin-move-rpicamsrc'
	  Move rpicamsrc from https://github.com/thaytan/gst-rpicamsrc/
	  It's a useful little element and works well, so might as well
	  make sure it's widely available so people can stop piping
	  raspivid output into fdsrc.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>

2020-05-02 19:27:20 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstrpicam-enums-template.c:
	* sys/rpicamsrc/gstrpicam-enums-template.h:
	  rpicamsrc: sync autotools with glib-mkenum usage in meson build

2020-05-02 18:28:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/meson.build:
	  rpicamsrc: meson: use gnome.glib_mkenums_simple() and fix build as Meson subproject
	  While at it also fix up the type defines, e.g.
	  GST_RPI_CAM_TYPE_RPI_CAM_SRC_EXPOSURE_MODE -> GST_RPI_CAM_SRC_TYPE_EXPOSURE_MODE

2020-05-03 11:09:47 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstplugin.map:
	* sys/rpicamsrc/meson.build:
	  rpicamsrc: meson: drop map file and fix plugin symbol export with newer gstreamer versions
	  Use -fvisibility instead of a map file for symbol export, so that
	  the right symbols get exported with newer gstreamer versions. Older
	  GStreamer versions also still work of course.
	  Fixes blacklisting/plugin-loading issues with GStreamer >= 1.14
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/984,
	  closes https://github.com/thaytan/gst-rpicamsrc/issues/94
	  and https://github.com/thaytan/gst-rpicamsrc/issues/67

2018-07-16 19:49:21 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Attempt to workaround MMAL timeout bug
	  mmal_queue_timedwait() might spuriously return immediately
	  if called at exactly the wrong instant due to an internal
	  off-by-one bug. Attempt to work around that and just retry.

2018-07-16 19:30:26 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Improve timeout error
	  Propagate timeout errors so they're not reported
	  generically

2018-06-21 22:50:28 +1000  Jan Schmidt <jan@centricular.com>

	* tests/examples/rpicamsrc/webrtc-unidirectional-h264.c:
	  rpicamsrc: webrtc example: Add a STUN server to the configuration
	  To let the webrtc example work through NAT firewalls

2018-06-21 22:44:25 +1000  Jan Schmidt <jan@centricular.com>

	* tests/examples/rpicamsrc/webrtc-unidirectional-h264.c:
	  rpicamsrc: webrtc example: Modify HTML to support other ports than 57778

2018-06-21 21:45:32 +1000  Jan Schmidt <jan@centricular.com>

	* tests/examples/rpicamsrc/webrtc-unidirectional-h264.c:
	  rpicamsrc: webrtc example: Remove external fmtp insertion
	  GStreamer 1.14.2 should contain the backport of gst-plugins-bad
	  commit 5c450c5 adding FEC and RTX support, and incidentally
	  the fmtp field in the SDP

2018-06-21 20:33:03 +1000  Jan Schmidt <jan@centricular.com>

	* tests/examples/rpicamsrc/webrtc-unidirectional-h264.c:
	  rpicamsrc: webrtc example: Set the locale
	  Make the date format in the overlay respect the current
	  locale

2018-06-20 15:36:42 +0000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Don't destroy the camera component on startup error
	  Just disable the camera component when it fails to start. The
	  most common reason is that the camera device is already in use,
	  and if we just disable the mmal component correct cleanup
	  will happen later

2018-05-12 21:13:52 +0000  Jan Schmidt <jan@centricular.com>

	* tests/examples/rpicamsrc/webrtc-unidirectional-h264.c:
	  rpicamsrc: Add webrtc streaming example
	  Add an example for testing webrtc streaming from the rpi
	  camera, based on the code from
	  https://bugzilla.gnome.org/show_bug.cgi?id=795404
	  Requires GStreamer 1.14.1 or git master

2018-05-12 19:57:43 +0000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Expose constrained-baseline profile
	  constrained-baseline is a useful profile for streaming to iOS
	  devices, and seems to work in the firmware, so let's publish it

2018-03-28 22:00:10 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.c:
	  rpicamsrc: Add define and increase reported maximum FPS from 90 to 1000

2017-11-14 15:01:21 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Expand frame timeout from 100ms to 500ms
	  rpicamsrc on a normal rpi camera doesn't start up fast enough,
	  and always fails the new 100ms timeout. A better solution
	  might be to have a longer timeout for the first frame, but
	  shorter once frames are running - but this quick fix will at
	  least make rpicamsrc work again.

2017-11-08 09:14:35 +0000  Georgii Staroselskii <georgii.staroselskii@emlid.com>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: RaspiCapture: use mmal_queue_timedwait() for buffer queueing
	  If an external camera was disconnected, there were no feedback in an
	  application. It seems reasonable to wait on mmal_queue no longer than
	  100ms. If it's stuck we just return a FLOW_ERROR and let the application
	  decide what to do later.

2017-11-07 15:14:06 +0000  Georgii Staroselskii <georgii.staroselskii@emlid.com>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: RaspiCapture: handle MMAL_EVENT_ERROR

2017-07-01 00:51:13 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Implement use-stc property to disable STC timestamps
	  If use-stc=false, then rpicamsrc won't apply
	  the camera timestamping to outgoing buffers, instead
	  relying on real-time timestamping by the
	  GStreamer clock. It means slightly less accuracy
	  and more jitter in timestamps, but might help on some
	  CSI inputs with broken timestamping.

2017-05-19 20:55:35 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Fix the descriptions of text annotation colour properties
	  The text annotation colour properties take an integer value
	  corresponding to a VUY colour, not a text string like
	  the copy-pasted description from raspivid suggests.
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/59

2017-01-27 12:58:29 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Implement dynamic bitrate update
	  Use mmal_port_set_parameter_uint32 to update the encoder
	  bitrate.
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/60

2016-10-08 11:12:09 +0000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Set outgoing buffer durations based on negotiated framerate.
	  make sure outgoing buffers have at least some duration set,
	  otherwise it leads to strange situations, like qtmux writing
	  out a file that doesn't include the final frame inside the
	  playable segment, because no-duration = 0 duration there.

2016-10-08 11:10:30 +0000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Destroy mmal pool on shutdown always.
	  Avoid hangs on the next run because we didn't clean up the mmal pool
	  last time we shutdown.

2016-10-03 15:29:49 +0000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Switch back to MJPEG codec for image/jpeg
	  The JPEG codec hangs, not sure why yet. The MJPEG
	  codec doesn't provide a quality setting, and sometimes
	  freezes on shutdown, but otherwise seems more
	  reliable

2016-10-03 14:00:54 +0000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Don't try and set H264 params with JPEG codec

2016-10-03 02:34:50 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: First attempt at implementing MJPEG and raw video support

2016-09-19 12:06:05 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstplugin.map:
	* sys/rpicamsrc/meson.build:
	  rpicamsrc: Add experimental build using the Meson build system
	  Builds in about 10 seconds vs. 77 seconds with autotools.

2016-08-30 17:00:41 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Implement GstVideoDirection interface
	  Instead of implementing a custom property, we implement that interface.

2016-07-21 02:29:57 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: MMAL gives buffers with nal alignment, not AU
	  Fix the output caps, our buffers are not AU aligned, since
	  the SPS / PPS are given in separate packets at the start.

2016-07-08 15:32:21 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Create orientation property
	  Its behavior and choices are analog to the ones present in [gl]videoflip
	  for the method property.

2016-01-03 08:26:23 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: basesrc event handlers should not unref
	  Don't unref the passed event when handling events via
	  the GstBaseSrc src pad event handler - basesrc does
	  the unref. That breaks handling of upstream
	  force-key-unit events by unreffing twice.
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/43

2015-12-17 14:16:10 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Add property getters for preview window position.
	  Add the lines in get_property() for the preview-x/y/w/h properties
	  so the values can be retrieved without causing critical warnings.
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/42

2015-12-02 01:20:10 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Add preview-x/y/w/h properties
	  Expose properties for setting the position of the preview
	  window on the screen

2015-10-21 21:11:36 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Add properties for configuring annotation text size and colour.
	  Map the raspivid setting for annotation text size and colours
	  to properties.

2015-10-08 10:32:32 +0200  ibauer <iljabauer@gmail.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Changed awb-gain-blue use the correct enum PROP_AWB_GAIN_BLUE and not PROP_AWB_GAIN_RED

2015-07-19 01:48:35 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Fix buffer PTS calculation
	  Timestamps from MMAL are in microseconds,
	  so make sure to convert to nanoseconds before
	  using them to adjust the GStreamer buffer time

2015-05-11 11:16:52 +0200  Philippe Normand <philn@igalia.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* tests/examples/rpicamsrc/test_orientation.c:
	  rpicamsrc: Basic orientation interface support
	  The (h,v)flip attributes are now supported through this interface.
	  It should also be possible to support (h,v)center attributes using the
	  ROI properties.

2015-05-11 21:29:58 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Describe awb-mode=off in lowercase
	  Change the descriptions for the awb-gain-blue and awb-gain-red
	  properties to say 'awb-mode=off' instead of 'awb-mode=OFF'
	  See https://github.com/thaytan/gst-rpicamsrc/issues/26

2015-05-11 10:17:18 +0200  Philippe Normand <philn@igalia.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: colorbalance: protect with config_lock mutex

2015-05-05 19:03:43 +0200  Philippe Normand <philn@igalia.com>

	* tests/examples/rpicamsrc/test_color_balance.c:
	  rpicamsrc: add test-color-balance example
	  This small test will display a live video preview of the rpicam with
	  the balance controls being updated once a second. The controls to
	  update can be disabled in the source by setting the CONTROL_* macros
	  values to 0.

2015-04-29 16:36:18 +0200  Philippe Normand <philn@igalia.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Implement GstColorBalance interface
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/24

2015-04-27 22:56:32 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Fix initial config setting.
	  Make sure to update the captsure config before starting
	  capture. Since the capture component now keeps a local
	  copy of the config, it's not updated automatically.

2015-04-27 04:05:42 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Disable bitrate, quantisation and intra-refresh dynamic changes
	  The firmware rejects dynamic changes of those encoder params.

2015-04-27 04:05:04 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.h:
	  rpicamsrc: Send vcos_log_warn via GStreamer debug messages

2015-04-27 02:43:14 +1000  Jan Schmidt <thaytan@noraisin.net>

	* tests/examples/rpicamsrc/dynamicprops.py:
	  rpicamsrc: Add dynamic properties example
	  Python example of adjusting saturation on the fly

2015-04-27 00:54:54 +1000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Update properties dynamically where possible
	  Update camera and encoder properties at runtime
	  where possible
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/19
	  and https://github.com/thaytan/gst-rpicamsrc/issues/23

2015-04-27 00:40:23 +1000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiPreview.c:
	* sys/rpicamsrc/RaspiPreview.h:
	  rpicamsrc: split preview config and state

2015-04-21 02:45:59 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Clear intra-refresh MMAL param struct.
	  Use memset on the stack allocated MMAL_PARAMETER_VIDEO_INTRA_REFRESH_T
	  struct. Apparently mmal_port_parameter_get() doesn't retrieve all
	  parameters, causing random failures when we set the intra-refresh
	  param on the encoder.
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/22 for me.

2015-04-21 01:17:55 +1000  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/RaspiCamControl.h:
	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Merge changes from userland repo
	  Current to b69f807ce59189457662c2144a8e7e12dc776988
	  No integration of stereoscopic support as yet

2015-04-21 00:02:27 +1000  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicam_types.h:
	  rpicamsrc: Map intra-refresh cyclic-rows to the correct MMAL param.

2015-03-10 00:22:40 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Use MMAL PTS and STC to calculate GStreamer timestamps
	  Don't apply timestamps based on output time from the encoder,
	  but use the MMAL STC and capture PTS to generate a GStreamer
	  timestamp that more accurately resembles the input (and would
	  preserve reordering should the encoder ever add B-frames).
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/16

2015-03-07 02:11:25 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	  rpicamsrc: Defer encoder creation until after caps are negotiated
	  This ensures the encoder is created with the profile
	  negotiated with downstream

2015-03-07 01:17:30 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Read and set H.264 profile from negotiated caps

2015-03-06 03:43:07 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicam_types.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Add intra-refresh-type property, and set default keyframe spacing to -1 (auto)
	  This plus other recent commits mostly fix
	  bug https://github.com/thaytan/gst-rpicamsrc/issues/16

2015-03-06 03:05:24 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicam_types.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Add annotation-mode and annotation-text properties

2015-03-06 02:42:00 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: implement sensor-mode property

2015-03-06 01:27:44 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	  rpicamsrc: More conversion to GStreamer logging

2015-03-06 01:15:48 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicam_types.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Implement drc property

2015-03-06 01:09:16 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: add awb-gain-red and awb-gain-blue properties

2015-03-06 00:52:37 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Add camera-number property

2015-03-06 00:45:05 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: add inline-headers and shutter-speed properties

2015-03-06 00:21:31 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Add quantisation-parameter property, support variable bitrate
	  Allow birate=0 and implement the quantisation-parameter property
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/21

2015-03-05 17:01:33 +1100  Jan Schmidt <jan@centricular.com>

	* sys/rpicamsrc/RaspiCLI.c:
	* sys/rpicamsrc/RaspiCLI.h:
	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/RaspiCamControl.h:
	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/RaspiPreview.c:
	* sys/rpicamsrc/RaspiPreview.h:
	  rpicamsrc: Incorporate raspivid changes from upstream
	  Merge all changes for new features from upstream
	  raspberrypi userland, up to commit 0de0b2

2015-01-05 02:21:16 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Add keyframe-interval property to the element

2014-10-30 00:45:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/RaspiCamControl.h:
	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.c:
	  rpicamsrc: deviceprovider: check if camera is detected and supported

2014-10-29 00:43:51 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.c:
	* sys/rpicamsrc/gstrpicamsrcdeviceprovider.h:
	  rpicamsrc: Add GstDeviceProvider for rpi camera module

2014-09-27 14:31:10 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: avoid single-element lists in template caps

2014-10-09 20:38:41 +0000  Vivia Nikolaidou <n.vivia@gmail.com>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Add force-key-unit event support

2014-03-13 00:16:18 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/RaspiPreview.c:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Move all debug output to go via GStreamer logs
	  Fixes https://github.com/thaytan/gst-rpicamsrc/issues/9

2013-10-19 18:52:25 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Update maximum framerate to 90 fps

2013-10-14 02:39:00 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Enable image effects

2013-10-13 18:01:00 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Re-flow element source code with gst-indent

2013-10-13 17:46:07 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicam-enums-template.c:
	* sys/rpicamsrc/gstrpicam-enums-template.h:
	* sys/rpicamsrc/gstrpicam_types.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Implement a bunch of the raspivid command-line params
	  Add properties for controlling various parts of the capture

2013-10-13 01:29:08 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: Tell basesrc to timestamp buffers for us, for now.

2013-10-13 01:20:51 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Initial caps nego and properties.
	  Support caps negotiation for H.264 frame size and framerate.
	  Add bitrate, saturation, brightness, contrast, sharpness properties.

2013-10-12 19:23:03 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/gstrpicamsrc.c:
	  rpicamsrc: First version which generates buffers on the src pad
	  Fixed to 1920x1080 h264 regardless of caps.

2013-10-12 12:42:07 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/RaspiStill.c:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Checkpoint. Version which writes directly to test.out
	  Switch to plain basesrc for parent class

2013-10-11 19:17:05 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/RaspiCamControl.c:
	* sys/rpicamsrc/RaspiCamControl.h:
	* sys/rpicamsrc/RaspiCapture.c:
	* sys/rpicamsrc/RaspiCapture.h:
	* sys/rpicamsrc/RaspiPreview.c:
	* sys/rpicamsrc/RaspiPreview.h:
	* sys/rpicamsrc/RaspiStill.c:
	* sys/rpicamsrc/RaspiStillYUV.c:
	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: checkpoint

2013-10-10 23:47:38 +1100  Jan Schmidt <thaytan@noraisin.net>

	* sys/rpicamsrc/gstrpicamsrc.c:
	* sys/rpicamsrc/gstrpicamsrc.h:
	  rpicamsrc: Initial commit
	  Simple modified gst-template to use BaseCameraSrc
	  Incorporate Broadcom mmal headers

2018-04-19 13:57:26 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: don't fail when seeking past the end of the content
	  Range errors are already turned into EOS when the size is not known.
	  Do the same thing if the request as outside the known content size.
	  This can be triggered by seeking in a queue2:
	  - Ensure that the range containing the end of the file is available.
	  - Seek into this range from a different range.
	  - queue2 creates a seek event with start=<file-size>
	  - this results in a "Requested Range Not Satisfiable" error
	  Fixes #452
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/385>

2019-11-10 21:19:09 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: don't update the size on error
	  Any data corresponding length in the message is not part of the requested
	  file.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/385>

2020-06-18 19:12:46 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtglrenderer.cc:
	  qt/gloverlay: fix using OpenGL after destroying Qml
	  Qml somewhat unhelpfully seems to uncurrent our OpenGL context on its
	  destruction.  Work around that by uncurrenting and recurrenting again.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/632>

2020-07-08 17:02:34 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* scripts/extract-release-date-from-doap-file.py:
	  meson: set release date from .doap file for releases
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/664>

2020-07-07 12:36:01 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't lock object lock twice in prefill mode
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/762
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/663>

2020-07-04 01:02:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audiofx/meson.build:
	* gst/deinterlace/meson.build:
	* gst/videobox/meson.build:
	* gst/videomixer/meson.build:
	* meson.build:
	* scripts/update-orc-dist-files.py:
	  meson: add update-orc-dist target
	  Add target to update backup orc -dist.[ch] files.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/662>

2020-05-26 10:27:35 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Do not renegotiate if only framerate changed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/602>

2020-07-02 09:15:08 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/flac/gstflacenc.c:
	  flacenc: Pass audio info from set_format() to query_total_samples() explicitly
	  This fixes writing of the seek table header.
	  gst_audio_encoder_get_audio_info() will still return old/unset audio
	  info until set_format() has actually returned, which then results in
	  query_total_samples() to always return 0.
	  Thanks to Jacob Kauffmann for debugging this and finding the main cause.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/756
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/661>

2020-07-03 02:03:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* meson.build:
	  Back to development

=== release 1.17.2 ===

2020-07-03 00:27:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.17.2

2020-07-02 07:53:14 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/deinterlace/meson.build:
	* meson.build:
	  deinterlace: Disable nasm support on x32
	  The assembly assumes pointers are 64-bit, so just disable it.
	  Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/757
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/660>

2020-07-01 18:19:09 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/deinterlace/meson.build:
	  deinterlace: Fix build on x32
	  Need to pass `-f elfx32` to nasm in that case.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/757
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/657>

2020-07-01 16:17:19 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Wait for caps on sparse streams
	  Don't set sparse streams to non-waiting at the collectpads
	  level until after capa arrive, as we need caps on all
	  pads before the file headers get written, or else the
	  subtitle track will be silently absent in the final file.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/724
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656>

2020-07-01 16:13:27 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Warn on late caps arrival
	  As well as warning when caps change after the headers
	  were already written, make sure to warn if the *first* caos
	  arrive late too.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656>

2020-06-30 18:37:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Return TRUE from the LATENCY query handling
	  We always answer it successfully no matter what.
	  The default return value in the function is FALSE even if the code below
	  sets it again to FALSE.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/654>

2020-06-29 11:53:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/imagefreeze.c:
	  imagefreeze: Add test for new live mode
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>

2020-06-29 10:10:09 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: Add a live mode
	  Previously imagefreeze would always operate as non-live element and
	  output frames as fast as possible according to the configured segment
	  (via SEEK events) and the negotiated framerate from start to stop or the
	  other way around.
	  With the new live mode (enabled via the is-live property) it would only
	  output frames in PLAYING. Frames would be output according to the
	  negotiated framerate unless it would be too late, in which case it would
	  jump ahead and skip over the requirement amount of frames.
	  This makes it possible to actually use imagefreeze in live pipelines
	  without having to manually ensure somehow that it would start outputting
	  at the current running time and without still risking to fall behind
	  without recovery.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>

2020-06-28 22:26:23 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Correctly answer the LATENCY query
	  We never run as a live element, even if upstream is live, and never
	  output any buffers with latency but immediately generate buffers as
	  fast as we can according to the negotiated framerate.
	  Passing the query upstream would proxy whatever mode of operation
	  upstream has, which has nothing to do with how we produce buffers.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>

2020-06-25 14:15:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Fix threading issues in orphaning mechanism
	  The pool orphaning function was colling internal _stop() virtual function
	  implementation. This is not thread safe, as a private lock inside the buffer
	  pool is supposed to be held. Fix this by keeping delayed _stop() and orphaning
	  the GstV4L2Allocator instead (REQBUFS(0)).
	  Then, protect the orphaned boolean with the object lock for the case a buffer
	  is being released after we have orphaned the buffer. That would otherwise
	  cause a QBUF to happen while the queue is no longer owned by the buffer pool.
	  This boolean is otherwise used and set from the streaming lock, or after
	  threads have been stopped (final cleanup).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/648>

2020-06-26 16:43:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpoool: Fix requeueue after seek when importing
	  When the buffer pool is importing buffer, it will requeue num_allocated on
	  streamon. As this value was not set for DMABuf import and USERPTR, no buffer
	  was queued  back.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/651>

2020-06-26 16:39:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  Revert "v4l2bufferpool: request the maximum number of buffers for USERPTR"
	  This reverts commit 6bf9f4bd77a4c6cce8786893feea7d601a6e6030.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/651>

2020-06-26 16:37:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  Revert "v4l2bufferpool: request the maximum number of buffers for DMABUF"
	  This reverts commit 94e323c10f2d7fa85bf63f357d203ca5305800c6.
	  Fixes #754
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/651>

2020-06-26 14:48:14 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Only resurrect the right amount of buffers
	  On streamon, we need to resurrect (queue back) some buffers, as during
	  flushign seek we'd endup with an empty queued. We initially started with
	  resurrecting as many as we could without blocking, but that miss-behaved with
	  dynamic CREATE_BUFS, causing the pool to grow dramatically. This was limited
	  by the number of allocated buffers, but this still tried to resurrect too many
	  buffers for the first run, as activating the pool will queued buffers.
	  In this patch, we calculte the missing detal in the queue and only try and
	  resurrect that amount of buffers.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/651>

2020-06-26 13:11:04 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Only offer inactive pools and if needed
	  Avoid offering a pool if it's not needed or if it's still active.
	  This works around the fact the we only have one pool in V4L2.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/651>

2020-06-24 21:58:07 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/gstqtglutility.cc:
	* ext/qt/meson.build:
	  qt: Rework how we find the Qt QPA header
	  Instead of querying the Qt include path from the dependency or from
	  qmake, rely on the qt5qml_dep to set the include path to QtGui
	  correctly, and look for the header inside the private includedir.
	  Then we can use that path to include the header directly.
	  Reported in https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/780#note_548092
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/640>

2020-06-24 22:04:55 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  qt: Only check for moc-qt5/moc in PATH if not cross-compiling
	  This is an extra check that's only needed for working around Linux
	  distribution packaging. `moc` is not required in the cross file.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/640>

2020-06-26 13:10:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Don't do REQBUFS(0) on inactive allocator
	  If the allocator is no longer active, it means the memory has already
	  been freed, calling REQBUF(0) would make no sense.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/649>

2020-06-26 11:05:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Avoid set_flushing warning
	  The gst_buffer_pool_set_flushing() warns when that function is called
	  on an inactive pool. Avoid the warning by checking the state, this is
	  similar to what we do in gst_v4l2_object_unlock().
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/649>

2020-06-26 09:53:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix data offset / bytesused size validation
	  The check was too strict causing spurious warning. Now check for <= so that 0
	  sized buffer do not cause a warning.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/649>

2020-06-25 16:46:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Fix negotiation caps leak
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/649>

2020-06-26 19:28:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: flesh out docs for format-location* signals
	  Make explicit that the returned strings need to be g_free()-able.
	  Fixes #753
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/652>

2020-06-25 16:47:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Skip negotiation of profiles/level if no codec
	  The codec structure is optional and not used for fwht test codec. This
	  was leading to a crash dereferencing NULL pointer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/650>

2020-05-03 13:17:46 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/rtpstats.c:
	  rtpstats: guard against division by zero
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/646>

2020-06-17 23:23:58 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtptwcc.c:
	  rtptwcc: fix pruning of ack'ed twcc-packets
	  Fixes #750
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/645>

2020-06-24 21:15:47 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tests/examples/qt/qmloverlay/meson.build:
	* tests/examples/qt/qmlsink-dynamically-added/meson.build:
	* tests/examples/qt/qmlsink/meson.build:
	* tests/examples/qt/qmlsrc/meson.build:
	  meson: Build Qt5 tests with -std=c++11
	  We already do this for the plugin.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/780#note_548179
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/642>

2020-06-25 12:58:48 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Add new properties for setting muxer/sink presets
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/644>

2020-06-24 17:04:51 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/autodetect/gstautodetect.c:
	  autodetect: mark filter-caps property as DOC_SHOW_DEFAULT
	  When generating the cache we inspect the base class through
	  an instance of one of its subclasses. We don't want potential
	  assignments in subclasses initialization to leak into the
	  base class documentation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/641>

2020-06-24 16:45:27 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* ext/vpx/gstvpxenc.c:
	  vpxenc: mark all properties as GST_DOC_SHOW_DEFAULT
	  When generating the cache we inspect the base class through
	  an instance of one of its subclasses. We don't want potential
	  assignments in subclasses initialization to leak into the
	  base class documentation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/641>

2020-06-23 19:04:03 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/equalizer/gstiirequalizer.c:
	  docs: mark GstIirEqualizer as plugin API

2020-06-23 12:47:44 -0400  Thibault Saunier <tsaunier@igalia.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vpx: Fix links to baseclass properties

2020-06-23 02:50:35 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tunerchannel.c:
	  docs: mark more types as plugin API

2020-06-23 00:02:34 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	  plugins_cache: add base classes

2020-06-23 00:02:21 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/meson.build:
	  meson: mark plugins cache target as always stale

2020-06-21 01:34:43 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/gtk/gstgtkbasesink.c:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxenc.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/autodetect/gstautodetect.c:
	  docs: mark more types as plugin API

2020-06-19 22:54:38 -0400  Thibault Saunier <tsaunier@igalia.com>

	* docs/gst_plugins_cache.json:
	  doc: Stop documenting properties from parents

2020-06-21 20:11:06 +0800  He Junyan <junyan.he@hotmail.com>

	* gst/deinterlace/yadif.c:
	  deinterlace: Add the missing ORC_RESTRICT define.
	  ORC_RESTRICT may not be defined in yadif.c and cause build error.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/637>

2019-06-06 09:41:13 +0200  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: make tests more stable
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/410>

2020-06-20 20:42:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	  docs: update plugin cache for new version
	  Some default values include our version string, like
	  user agent strings.

2020-06-20 00:28:11 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development

=== release 1.17.1 ===

2020-06-19 19:18:59 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.17.1

2020-06-19 20:24:12 +0900  Seungha Yang <seungha@centricular.com>

	* gst/deinterlace/meson.build:
	  meson: deinterlace: Check host cpu type for asm build
	  Add host cpu type check as we would enable asm only for x86_64
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/636>

2020-06-19 19:54:08 +0900  Seungha Yang <seungha@centricular.com>

	* meson.build:
	  meson: Fix build error with MSVC caused by ARCH_X86_64 define
	  ARCH_X86_64 define will enable GCC specific code path in dv_types.h
	  while building dv plugin.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/636>

2020-06-19 10:32:45 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* ext/shout2/gstshout2.c:
	  shout2: advertise documentation caps properly
	  shout2send caps depend on what the libshout2
	  version in question supports, but the
	  documentation caps should always be the same.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/635>

2019-05-26 20:20:03 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/meson.build:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux_tags.c:
	* gst/isomp4/qtdemux_tags.h:
	  qtdemux: Split tag reading functions out
	  Move some code out of the enormous qtdemux.c into a separate
	  qtdemux_tags helper, and make some structs available via qtdemux.h
	  to accommodate that.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/634>

2019-05-26 05:05:06 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/meson.build:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_tree.c:
	* gst/isomp4/qtdemux_tree.h:
	  qtdemux: Move some tree parsing files out to a separate file.
	  Reduce a tiny bit of the bulk of qtdemux.c by moving some
	  agnostic helper functions out.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/634>

2019-05-26 01:24:54 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Factor out svmi parsing. Fix bounds checking.
	  Move the SVMI stereoscopic atom parsing out to a helper
	  function to shrink qtdemux_parse_trak a bit.
	  Add a bounds check that the received atom is large enough
	  before parsing it.
	  Add a note to the atom parser that svmi comes from the
	  MPEG-A spec 23000-11.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/634>

2020-06-15 13:05:49 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* ext/pulse/pulsedeviceprovider.c:
	  pulse: fix discovery of newly added devices
	  Fix regression introduced in 7bc5e28d85992b03e5852879b8d4d96043496caf
	  preventing the device provider to send the device-added message for new
	  devices.
	  By early returning the patch was discarding add/remove events.
	  Fix #735
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/626>

2020-06-18 10:47:28 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/qt/qmlsink-dynamically-added/meson.build:
	* tests/examples/qt/qmlsink-dynamically-added/play.pro:
	* tests/examples/qt/qmlsink-dynamically-added/qmlsink-dyn-added.qrc:
	  examples: qmlsink: rename qrc file to avoid naming conflicts with older meson versions
	  Would get "Tried to create target "qt5-qmlsink_qrc", but a
	  target of that name already exists." with older meson versions.
	  Work around that by renaming the qrc file.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/633>

2020-06-17 16:42:16 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	  meson: Check the nasm version with run_command
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/751
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/631>

2020-06-16 19:34:01 +0900  Seungha Yang <seungha@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't return TRUE for unhandled query
	  Expected return value for unhandled query is FALSE
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/629>

2020-06-16 11:52:38 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/meson.build:
	* gst/deinterlace/x86/x86inc.asm:
	* gst/deinterlace/x86/yadif.asm:
	* gst/deinterlace/yadif.c:
	* gst/deinterlace/yadif.h:
	* meson.build:
	* meson_options.txt:
	  deinterlace: Add yadif ASM optimisations
	  Measured to be about 3.4x faster than C
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/621>

2020-06-12 13:21:02 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/yadif.c:
	  deinterlace: Fix invalid read in yadif
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/621>

2020-06-12 12:18:11 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtglrenderer.cc:
	  qt/gloverlay: reset OpenGL state after Qt drawing
	  Reset to the original OpenGL state as required by the GStreamer OpenGL
	  API contract.  Fixes output with a glimagesink element downstream.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/623>

2020-06-12 12:16:49 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtglrenderer.cc:
	  qt/gloverlay: reset current OpenGL context after Qt
	  Qt may replace the drawable with its own which breaks output if Qt is
	  not displaying the resulting video as used with e.g. glimagesink.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/623>

2020-06-12 09:52:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Change a GST_ERROR_OBJECT() back to GST_DEBUG_OBJECT()
	  It was accidentally changed in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/436
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/624>

2020-06-11 20:39:33 +0300  Jordan Petridis <jpetridis@gnome.org>

	* gst/isomp4/gstqtmux.c:
	* sys/v4l2/gstv4l2videodec.c:
	  Use gst_element_class_set_metadata when passing dynamic strings
	  gst_element_class_set_metadata is meant to only be used with
	  static or inlined strings, which isn't the case for the 2 elements
	  here resulting in use-after-free later on.
	  https://gstreamer.freedesktop.org/documentation/gstreamer/gstelement.html?gi-language=c#gst_element_class_set_static_metadata
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/622>

2020-06-10 13:56:22 +0000  Sebastian Dröge <slomo@coaxion.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Revert "rtpjitterbuffer: Avoid deadlock on flush"
	  This reverts commit 54810bf44f27d9c180730f58f16f6e172c7b0bc8
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/620>

2020-06-09 15:12:13 -0400  Thibault Saunier <tsaunier@igalia.com>

	* docs/gst_plugins_cache.json:
	  docs: Update plugins cache

2020-06-09 13:09:20 -0700  U. Artie Eoff <ullysses.a.eoff@intel.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: g_queue_clear_full introduced in glib 2.60
	  Define g_queue_clear_full if glib < 2.60.
	  Fixes #747
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/619>

2020-06-08 11:33:16 -0400  Thibault Saunier <tsaunier@igalia.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Make internal-ssrc as show default for doc

2020-06-08 10:56:02 -0400  Thibault Saunier <tsaunier@igalia.com>

	* docs/gst_plugins_cache.json:
	  docs: Update plugins cache

2020-06-09 15:21:25 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/meson.build:
	  tests: don't pull in all -bad plugin, only allow the one we need
	  Set up our plugin include list for tests in such a way that
	  we don't pull in *all* plugins from -bad but only the one
	  used in the splitmuxsink unit test, i.e. the timecode plugin,
	  so we don't accidentally use other encoders/decoders such as
	  nvenc/dec for example.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/617>

2020-06-08 17:41:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtptimerqueue.c:
	  rtptimerqueue: Fix leak on timer collision
	  While the caller should make sure this does not happen, make sure timer
	  collision are not silently ignored and leaked.
	  Fixes #726
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/616>

2020-03-27 15:48:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Keep JBUF lock while processing timers
	  Until now, do_expected_timeout() was shortly dropping the JBUF_LOCK in order
	  to push RTX event event without causing deadlock. As a side effect, some
	  CPU hung would happen as the timerqueue would get filled while looping over
	  the due timers. To mitigate this, we were processing the lost timer first and
	  placing into a queue the remainign to be processed later.
	  In the gap caused by an unlock, we could endup receiving one of the seqnum
	  present in the pending timers. In that case, the timer would not be found and
	  a new one was created. When we then update the expected timer, the seqnum
	  would already exist and the updated timer would be lost.
	  In this patch we remove the unlock from do_expected_timeout() and place all
	  pending RTX event into a queue (instead of pending timer). Then, as soon as
	  we have selected a timer to wait (or if there is no timer to wait for) we send
	  all the upstream RTX events. As we no longer unlock, we no longer need to pop
	  more then one timer from the queue, and we do so with the lock held, which
	  blocks any new colliding timers from being created.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/616>

2020-06-08 09:33:10 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* tests/check/elements/vp9enc.c:
	  tests: vp9enc: enforce I420 format
	  Test was not enforcing a video format on videotestsrc. I420 was picked
	  as it was the first format in GST_VIDEO_FORMATS_ALL which will no longer
	  be true (gst-plugins-base!689).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/615>

2020-05-30 08:55:19 +0200  Edward Hervey <edward@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Avoid deadlock on flush
	  When a GST_EVENT_FLUSH_START reaches the jitterbuffer, there is a chance that
	  our task is currently blocking waiting for a timer.
	  There was two problems:
	  * That wait wasn't checking for flushing situations
	  * The flushing handling wasn't waking up that conditional (to check whether it
	  should abort)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/608>

2020-06-06 00:42:25 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/aalib/gstaasink.c:
	* ext/aalib/gstaatv.c:
	* ext/dv/gstdvdec.c:
	* ext/flac/gstflacenc.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/lame/gstlamemp3enc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libcaca/gstcacatv.c:
	* ext/shout2/gstshout2.c:
	* ext/speex/gstspeexenc.c:
	* ext/twolame/gsttwolamemp2enc.c:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxenc.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/alpha/gstalpha.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstripple.c:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp9pay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideomedian.c:
	* gst/videomixer/videomixer2.c:
	* sys/v4l2/gstv4l2object.c:
	  plugins: uddate gst_type_mark_as_plugin_api() calls

2020-06-05 11:49:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Initialize uninitialized variable correctly
	  `last_out` would be used uninitialized if the element has no `set-active`
	  signal. Initialize it to -1 as that's what the "default" value is
	  further below.
	  CID 1455443
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/727
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/613>

2015-11-26 17:52:29 +0100  Mikhail Fludkov <misha@pexip.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvpxenc.c:
	* ext/vpx/gstvpxenc.h:
	* tests/check/elements/vp8enc.c:
	* tests/check/elements/vp9enc.c:
	  vpxenc: Add new bit-per-pixel property to select a better "default" bitrate
	  As part of this also change the default bitrate value to 0. The default
	  value was 256000 previously. In reality, if the property was not set the
	  bitrate value would be scaled according to the resolution which is not
	  very intuitive behavior. It is better to use 0 for this purpose. Now
	  together with newly introduced property "bits-per-pixel" 0 means to
	  assign the bitrate according to resolution/framerate.
	  The default bitrates are now
	  - 1.2Mbps for VP8 720p@30fps
	  - 0.8Mbps for VP9 720p@30fps
	  and scaled accordingly for different resolutions/framerates.
	  Previously the default bitrate was also not scaled according to the
	  framerate but only took the resolution into account.
	  This also fixes the side effect of setting bitrate to 0. Previously
	  encoder would not produce any data at all.
	  Addition from Sebastian Dröge <sebastian@centricular.com> to assume
	  30fps if no framerate is given in the caps instead of not calculating
	  any bitrate at all.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/611>

2020-06-03 18:35:58 -0400  Thibault Saunier <tsaunier@igalia.com>

	* docs/meson.build:
	  doc: Require hotdoc >= 0.11.0

2020-06-02 14:58:47 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  doc: Fix wrong link to GString in rtpjitterbuffer

2020-05-27 16:01:22 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	  docs: Update gst_plugins_cache.json

2020-05-30 01:29:03 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/aalib/gstaasink.c:
	* ext/aalib/gstaatv.c:
	* ext/dv/gstdvdec.c:
	* ext/flac/gstflacenc.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/lame/gstlamemp3enc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libcaca/gstcacatv.c:
	* ext/shout2/gstshout2.c:
	* ext/speex/gstspeexenc.c:
	* ext/twolame/gsttwolamemp2enc.c:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxenc.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/alpha/gstalpha.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstripple.c:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp9pay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideomedian.c:
	* gst/videomixer/videomixer2.c:
	* sys/v4l2/gstv4l2object.c:
	  plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types

2018-02-28 15:46:51 +0100  Stian Selnes <stian@pexip.com>

	* ext/vpx/gstvpxdec.c:
	* tests/check/elements/vp8dec.c:
	  vpxdec: Check that output width and height != 0
	  For VP8 it's possible to signal width or height to be 0, but it does
	  not make sense to do so. For VP9 it's impossible. Hence, we most
	  likely have a corrupt stream. Trying to negotiate caps downstream with
	  either width or height as 0 will fail with something like
	  gst_video_decoder_negotiate_default: assertion 'GST_VIDEO_INFO_WIDTH (&state->info) != 0' failed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/610>

2020-05-29 00:45:03 +0900  Seungha Yang <seungha@centricular.com>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	  speex: Fix crash on Windows caused by cross-CRT issue
	  Use speex_header_free() to free memory which was allocated by
	  library. Cross-CRT issue should not happen on 1.17 Cerbero build
	  but might happen custom build or so.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/606>

2020-05-27 22:33:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Use the correct type for storing the max-rtcp-rtp-time-diff property
	  It's an integer property and rtpbin also expects an integer. Passing it
	  as a GstClockTime (guint64) to g_object_set() will cause problems, and
	  on big endian MIPS apparently causes crashes.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/737
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/605>

2020-05-27 12:42:38 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/meson.build:
	  tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
	  If core is built as a subproject (e.g. as in gst-build), make sure to use
	  the gst-plugin-scanner from the built subproject. Without this, gstreamer
	  might accidentally use the gst-plugin-scanner from the install prefix if
	  that exists, which in turn might drag in gst library versions we didn't
	  mean to drag in. Those gst library versions might then be older than
	  what our current build needs, and might cause our newly-built plugins
	  to get blacklisted in the test registry because they rely on a symbol
	  that the wrongly-pulled in gst lib doesn't have.
	  This should fix running of unit tests in gst-build when invoking
	  meson test or ninja test from outside the devenv for the case where
	  there is an older or different-version gst-plugin-scanner installed
	  in the install prefix.
	  In case no gst-plugin-scanner is installed in the install prefix, this
	  will fix "GStreamer-WARNING: External plugin loader failed. This most
	  likely means that the plugin loader helper binary was not found or
	  could not be run. You might need to set the GST_PLUGIN_SCANNER
	  environment variable if your setup is unusual." warnings when running
	  the unit tests.
	  In the case where we find GStreamer core via pkg-config we use
	  a newly-added pkg-config var "pluginscannerdir" to get the right
	  directory. This has the benefit of working transparently for both
	  installed and uninstalled pkg-config files/setups.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/603>

2020-05-25 20:11:31 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Error out when failling to receive message response
	  And let it rety twice.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/717
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/601>

2020-05-21 17:12:55 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: videodec: Fix broken template caps
	  The profiles and levels were applied to the common caps instead of the copy.
	  That had the side effect of setting profiles/level from one CODEC onto
	  another. Leaving to encoder not being registered or not-negotiated errors.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/599>

2020-05-21 17:09:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2codec.c:
	* sys/v4l2/gstv4l2codec.h:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: codec: Fix GValue leak
	  The levels and profiles probe function returned a dynamically allocated GValue
	  that was leaked. Simplify this by using a stack allocated GValue and a boolean
	  return value.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/599>

2020-05-21 16:39:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2codec.c:
	  v4l2codec: Remove uneeded factorisation
	  There is only one user of that function and the split only increase
	  complexicity.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/599>

2020-05-20 17:30:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Ignore non-increasing sequence number
	  With older kernel, older driver or just broken drivers, the sequence number
	  may not be increasing. This simply ignore the sequence in this case. This
	  would otherwise miss-leading large amount of lost frame being reported.
	  Fixes #729
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/598>

2020-05-18 13:17:14 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtoverlay.cc:
	* ext/qt/gstqtoverlay.h:
	* tests/examples/qt/qmloverlay/main.cpp:
	  qtoverlay: add the root item as a property
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/595>

2020-05-20 13:17:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Send gap events if one of the streams falls behind the other by more than 3s
	  Same mechanism and threshold as in other demuxers.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/597>

2020-05-20 12:53:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.h:
	  flvdemux: Remove unused audio_linked/video_linked booleans
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/597>

2020-05-20 10:46:45 +0200  Edward Hervey <edward@centricular.com>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Answer bitrate queries from upstream
	  If upstream (such as queue2 in urisourcebin) asks for our bitrate, check if we
	  have stored audio/video bitrates, and use them.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/596>

2020-05-20 10:45:16 +0200  Edward Hervey <edward@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Handle empty metadata strings
	  g_utf8_validate() errors out on empty string. But empty strings are valid,
	  so only check if they're not
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/596>

2020-05-20 10:44:19 +0200  Edward Hervey <edward@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Set ACCEPT_TEMPLATE flag on sinkpad
	  A demuxer can accept any caps matching its sinkpad template caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/596>

2020-05-15 19:20:45 +0300  Raul Tambre <raul@tambre.ee>

	* ext/qt/qtglrenderer.cc:
	  qtglrenderer.cc: Fix compiling
	  46bfb7d247aef880c15300dad63eb2bbf6dc4928 fixed a format warning without checking if it actually compiled.
	  toUtf8() returns QByteArray so we need to assign it to a temporary variable to be able to get the raw string data from it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/593>

2020-05-15 06:07:25 +0000  Raul Tambre <raul@tambre.ee>

	* ext/qt/qtglrenderer.cc:
	  qtglrenderer.cc: Fix -Wformat-security warning
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/592>

2020-05-12 04:35:37 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	* ext/taglib/meson.build:
	* meson.build:
	* sys/osxvideo/meson.build:
	  meson: Pass native: false to add_languages()
	  This is needed for cross-compiling without a build machine compiler
	  available. The option was added in 0.54, but we only need this in
	  Cerbero and it doesn't break older versions so it should be ok.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/589>

2020-05-12 04:33:43 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	* ext/taglib/meson.build:
	* meson.build:
	  meson: Make C++ compiler detection not be automagic
	  It is now controlled by the qt5 and/or taglib options. We won't
	  silently fail to build taglib now.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/587>

2020-05-12 04:32:01 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/gtk/meson.build:
	* ext/qt/meson.build:
	* meson.build:
	* tests/examples/gtk/meson.build:
	  meson: Fix gstgl checks for qt and gtk
	  Also rename from build_ to have_, which is more accurate.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/587>

2020-05-12 04:30:13 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	* tests/examples/qt/meson.build:
	* tests/examples/qt/qmloverlay/meson.build:
	* tests/examples/qt/qmlsink-dynamically-added/meson.build:
	* tests/examples/qt/qmlsink/meson.build:
	* tests/examples/qt/qmlsrc/meson.build:
	  meson: Revamp qt5qml plugin and example build code
	  Stricter and simpler. For example, now we properly error out when
	  gstreamer-gl-1.0 was not found when the qt5 plugin is enabled or when
	  a C++ compiler is not enabled.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/587>

2020-05-09 03:09:03 +1000  Jan Schmidt <jan@centricular.com>

	* gst/deinterlace/yadif.c:
	  deinterlace: Split out NULL checks in yadif
	  Separate out explicit NULL checks for fields we depend on so
	  that coverity can hopefully verify dependencies better.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/585>

2020-05-09 03:07:33 +1000  Jan Schmidt <jan@centricular.com>

	* gst/deinterlace/tvtime/greedy.c:
	  deinterlace: Handle NV12/NV21 for the greedyl mode.
	  Don't fall back on the default interpolate_scanline function, which
	  blindly tries to copy from the next field, which can be NULL in
	  mixed progressive/interlaced streams
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/585>

2020-05-05 16:59:56 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/yadif.c:
	  deinterlace: Support packed formats for YADIF
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>

2020-05-06 11:04:18 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: Call the planar functions for the Y plane of nv12/nv21
	  In some algorithms (like yadif), the Y plane has to be handled different
	  than the UV plane. Therefore, the planar_y functions are now called for
	  the Y plane, and the nv12/nv21 functions are handling only the UV/VU
	  planes respectively.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>

2020-01-03 02:34:59 +1100  Jan Schmidt <jan@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/meson.build:
	* gst/deinterlace/yadif.c:
	* gst/deinterlace/yadif.h:
	  deinterlace: Add C implementation of YADIF
	  Import the YADIF deinterlacer from ffmpeg and modify
	  it to match the simple deinterlace scanlines structure.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>

2020-01-03 02:33:25 +1100  Jan Schmidt <jan@centricular.com>

	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	  deinterlace: Allow for 5 fields for interpolation
	  Add an extra field to the simple deinterlace implementation,
	  so that methods can potentially use 5 fields - the current
	  field, and 2 before and 2 after.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>

2020-05-07 01:17:25 +1000  Jan Schmidt <jan@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Force renegotiation when changing mode
	  Switching the deinterlacing mode on-the-fly from disabled to
	  auto used to work, but was broken by commit #1f21747c some
	  years ago.
	  Force re-negotiation with downstream when the mode or
	  fields properties are changed, otherwise deinterlace
	  never switches out of the passthrough mode.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/584>

2020-04-23 15:32:58 +0800  nian.yan <nian.yan@transwarp.io>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: remove meta copy in jpegenc
	  GstVideoEncoder takes care of the Meta copy, so there is no need in
	  jpegenc
	  Fixes http://gstreamer-devel.966125.n4.nabble.com/jpegenc-copy-GstMeta-twice-tt4693981.html
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/576>

2020-05-05 17:47:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: Handle flushing correctly
	  First of all get rid of the atomic seeking boolean, which was only ever
	  set and never read. Replace it with a flushing boolean that is used in
	  the loop function to distinguish no buffer because of flushing and no
	  buffer because of an error as otherwise we could end up in a
	  GST_FLOW_ERROR case during flushing.
	  Also only reset the state of imagefreeze in flush-stop when all
	  processing is stopped instead of doing it as part of flush-start.
	  And last, get a reference to the imagefreeze buffer in the loop function
	  in the very beginning and work from that as otherwise it could in theory
	  be replaced or set to NULL in the meantime as we release and re-take the
	  mutex a couple of times during the loop function.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/580>

2020-05-06 06:48:24 +0200  Edward Hervey <edward@centricular.com>

	* gst/videobox/gstvideobox.c:
	  videbox: Use MIN instead of CLAMP for uint
	  an unsigned int is always positive.
	  CID #206207
	  CID #206208
	  CID #206209
	  CID #206210
	  CID #206211
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/583>

2020-05-06 06:35:27 +0200  Edward Hervey <edward@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Avoid potential double-free
	  stream->name was being freed (without being NULL-ed) before we were certain it
	  would be set again.
	  CID #1456071
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/582>

2020-05-05 17:30:48 +0200  Edward Hervey <edward@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Don't leak frame in error case
	  CID #1455494
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/581>

2020-05-05 15:19:49 +0200  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  slitmuxsrc: Properly stop the loop if not part reader is present
	  Previously this would end up in a refcounting loop of hell.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/578>

2020-03-31 14:32:19 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Add skip-backwards-streams property
	  Backwards timestamps confuse librtmp, even if they're only backwards
	  relative to the other stream. If the timestamp of a stream is going
	  backwards related to the other stream, this property allows the muxer to
	  skip a few buffers until it reaches the timestamp of the other stream.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/572>

2020-03-31 14:10:35 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/flv/gstflvmux.c:
	  flvmux: Allow requesting streamable pads after header is written
	  Allows us to request pads after writing header for streamable flv's.
	  For non-streamable it doesn't make sense to request a new pad after
	  writing the header, because the headers have been written already and we
	  can't add the new stream. But for streamable, any clients that connect
	  after the new pad has been added will be able to see both streams.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/572>

2020-04-27 18:11:32 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qt/x11: also pass the window for gstgl -> qt context
	  Removes this warning from Qt:
	  QGLXContext: Multiple configs for FBConfig ID -1
	  QSGContext::initialize: depth buffer support missing, expect rendering errors
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/575>

2020-04-27 15:34:15 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtglrenderer.cc:
	* ext/qt/qtglrenderer.h:
	  qt: perform surface creation in the main thread
	  As is required when creating a QWindow instance set out in the Qt
	  documentation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/575>

2020-04-22 15:32:31 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Add 'mp3 ' fourcc that VLC seems to produce now
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/574>

2020-04-22 14:09:37 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: Properly free internal packets queue in finalize()
	  As we override the GLib item with our own structure, we cannot use any
	  function from GList or GQueue that would try to free the RTPJitterBufferItem.
	  In this patch, we move away from g_queue_new() which forces using
	  g_queue_free(). This this function could use g_slice_free() if there is any items
	  left in the queue. Passing the wrong size to GSLice may cause data corruption
	  and crash.
	  A better approach would be to use a proper intrusive linked list
	  implementation but that's left as an exercise for the next person
	  running into crashes caused by this.
	  Be ware that this regression was introduced 6 years ago in the following
	  commit [0], the call to flush() looked useless, as there was a g_queue_free()
	  afterward.
	  Signed-off-by: Nicolas Dufresne <nicolas.dufresne@collabora.com>
	  [0] https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/commit/479c7642fd953edf1291a0ed4a3d53618418019c
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/573>

2020-04-20 19:43:57 +0900  Seungha Yang <seungha@centricular.com>

	* tests/check/elements/splitmuxsink.c:
	* tests/check/elements/splitmuxsinktimecode.c:
	* tests/check/meson.build:
	  tests: splitmuxsink: Add more timecode based split test
	  ... and split test cases to run tests in parallel

2020-04-10 23:52:45 +0900  Seungha Yang <seungha@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Enhancement for timecode based split
	  The calculated threshold for timecode might be varying depending on
	  "max-size-timecode" and framerate.
	  For instance, with framerate 29.97 (30000/1001) and
	  "max-size-timecode=00:02:00;02", every fragment will have identical
	  number of frames 3598. However, when "max-size-timecode=00:02:00;00",
	  calculated next keyframe via gst_video_time_code_add_interval()
	  can be different per fragment, but this is the nature of timecode.
	  To compensate such timecode drift, we should keep track of expected
	  timecode of next fragment based on observed timecode.

2020-04-11 00:35:16 +0900  Seungha Yang <seungha@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Post error when requested timecode interval is invalid
	  In case we cannot rely on max-size-timecode for split decision,
	  post error instead of crashing

2020-04-16 16:47:50 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: don't use RTX packets in rate-calc and reset-logic
	  The problem was this:
	  Due to the highly irregular arrival of RTX-packet the max-misorder variable
	  could be pushed very low. (-10).
	  If you then at some point get a big in the sequence-numbers (62 in the
	  test) you end up sending RTX-requests for some of those packets, and then
	  if the sender answers those requests, you are going to get a bunch of
	  RTX-packets arriving. (-13 and then 5 more packets in the test)
	  Now, if max-misorder is pushed very low at this point, these RTX-packets
	  will trigger the handle_big_gap_buffer() logic, and because they arriving
	  so neatly in order, (as they would, since they have been requested like
	  that), the gst_rtp_jitter_buffer_reset() will be called, and two things
	  will happen:
	  1. priv->next_seqnum will be set to the first RTX packet
	  2. the 5 RTX-packet will be pushed into the chain() function
	  However, at this point, these RTX-packets are no longer valid, the
	  jitterbuffer has already pushed lost-events for these, so they will now
	  be dropped on the floor, and never make it to the waiting loop-function.
	  And, since we now have a priv->next_seqnum that will never arrive
	  in the loop-function, the jitterbuffer is now stalled forever, and will
	  not push out another buffer.
	  The proposed fixes:
	  1. Don't use RTX in calculation of the packet-rate.
	  2. Don't use RTX in large-gap logic, as they are likely to be dropped.

2020-04-15 12:36:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Increase internal bitstream pool size
	  This patch will now set the maximum of buffers to 32, allowing to grow the
	  pool for drivers that supports that and will respect the minimum buffers
	  reported by the driver. This was made to fix a stall with the virtio CODEC
	  driver.
	  Fixes #672

2020-04-15 17:50:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Do split-at-running-time splitting based on the time of the start of the GOP
	  If the start of the GOP is >= the requested running time, put it into a
	  new fragment. That is, split-at-running-time would always ensure that a
	  split happens as early as possible after the given running time.
	  Previously it was comparing against the current incoming timestamp,
	  which does not tell us what we actually want to know as it has no direct
	  relation to the GOP start/end.

2020-04-15 13:21:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix off-by-one in running time comparison for split-at-running-time
	  If we get a keyframe exactly at the requested running time we would only
	  split on the next keyframe afterwards due to wrong usage of > vs. >=.

2020-04-09 12:23:44 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Properly set segments seqnums after seeks

2020-04-08 19:49:00 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't write an empty string as a tag
	  To stop warnings like:
	  GStreamer-WARNING **: 19:47:48.186: Trying to set empty string on
	  taglist field 'encoder'. Please file a bug.

2020-04-08 12:34:40 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: request the maximum number of buffers for USERPTR
	  This is to match what we now do for DMABuf importation.

2019-11-20 15:32:29 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: request the maximum number of buffers for DMABUF
	  There are often only two buffers queued in the kernel so no new buffers are
	  requested.
	  With every qbuf, the kernel receives a new DMABUF for the specified index.
	  This most likely differs from the last DMABUF and the old cached entry is
	  released. This results in a lot of map/unmap overhead if the kernel driver
	  needs a mapping for the buffer.
	  With a larger queue, it's quite likely, that both old and new DMABUFs are
	  also mapped for another index. So the map/unmap is skipped, because the
	  mapping is reference counted.
	  The corresponding allocated buffers don't contain any actual memory, so
	  allocating them is quite cheep. So the log message is updated to clarify
	  this.

2020-04-08 09:45:17 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Avoid stack overflow recursing waiting for response
	  Instead of recursing, simply implement a loop with gotos, the same
	  way it was done before 812175288769d647ed6388755aed386378d9210c
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/710

2020-04-06 16:25:59 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Add property for enforcing the creation of chunks in single-stream files
	  This is disabled by default as it unnecessarily creates bigger headers
	  but it is something that is required by some applications and most
	  notably the Apple ProRes spec.

2020-04-03 00:16:10 +1100  Jan Schmidt <jan@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix invalid padlist accesses.
	  Request pads can released at any time, so make sure to hold
	  the object lock when iterating the element sinkpads list where
	  that's safe, or to use other safe pad iteration patterns in
	  other places.
	  When choosing a best pad, return a reference to the pad to make sure it
	  stays alive for output in the aggregator srcpad task.
	  Should fix a spurious valgrind error in the CI flvmux tests and some
	  other potential problems if the request sink pads are released while
	  the element is running..
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/714

2018-10-22 15:41:56 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Add option to create a timecode trak in non-mov flavors
	  Even if timecode trak is officially unsupported in non-mov flavors,
	  some software still supports it, e.g. Final Cut Pro X:
	  https://developer.apple.com/library/archive/technotes/tn2174/_index.html
	  The user might still expect to see the timecode information in the
	  non-mov file despite it being officially unsupported , because other
	  software e.g. QuickTime will create a timecode trak even in mp4 files.
	  Furthermore, software that supports timecode trak in non-mov flavors
	  will also display the file duration in "timecode units" instead of real
	  clock time, which is not necessarily the same for 29.97 fps and friends.
	  This might confuse users, who see a different duration for the same
	  framerate and amount of frames depending on whether the container is mp4
	  or mov.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/512

2020-01-16 09:30:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpL8depay.c:
	  rtpLXXdepay: Set the UNPOSITIONED flag on the audio-info when configuring an unpositioned layout
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/688

2020-04-01 13:19:46 +0200  Kristofer Björkström <kristofb@axis.com>

	* gst/rtp/gstrtpjpegpay.c:
	* tests/check/elements/rtpjpeg.c:
	* tests/check/meson.build:
	  rtpjpeg: Use gst_memory_map() instead of gst_buffer_map()
	  gst_buffer_map () results in memcopying when a GstBuffer contains
	  more than one GstMemory.
	  This has quite an impact on performance on systems with limited amount
	  of resources. With this patch the whole GstBuffer will not be mapped at
	  once, instead each individual GstMemory will be iterated and mapped
	  separately.

2020-04-01 13:17:03 +0200  Kristofer Björkström <kristofb@axis.com>

	* gst/rtp/gstbuffermemory.c:
	* gst/rtp/gstbuffermemory.h:
	  buffermemory: keep track of buffer size and current offset
	  Added the possibility to get current offset and the total size of the
	  buffer.

2020-04-03 10:29:18 +0200  Havard Graff <havard@pexip.com>

	* gst/rtp/gstrtpopuspay.c:
	* tests/check/elements/rtpopus.c:
	* tests/check/meson.build:
	  rtpopuspay: make depay ! pay work
	  There is a use-case for a server to re-payload opus going through it.
	  Problem was that the payloader requires channels in the caps, but
	  this is not something the depayloader can parse out of the stream, meaning
	  caps-negotiation would fail.
	  Removing the requirement of channels in the template-caps fixes this.

2020-04-03 16:49:25 +0900  Seungha Yang <seungha@centricular.com>

	* tests/check/elements/splitmuxsink.c:
	* tests/check/elements/splitmuxsrc.c:
	* tests/check/meson.build:
	  tests: Split splitmux test case
	  Since we are adding more and more tests into splitmux,
	  we need to split it to avoid CI timeout.

2020-04-03 13:45:56 +0900  Seungha Yang <seungha@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* tests/check/elements/splitmux.c:
	  splitmuxsink: Don't send too many force key unit event
	  splitmuxsink should requst keyframe depending on configured
	  threshold and previously requested time in order to avoid too many
	  keyframe request.

2020-03-20 21:32:36 +1100  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska: Check the return value of gst_segment_do_seek()
	  gst_segment_do_seek() can fail.

2018-06-08 13:12:01 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Send instant-rate-change event if requested in the SEEK event
	  Handle an instant rate change seek immediately by reflecting
	  it downstream as an instant-rate-change event, and do no
	  further seek handling.

2018-05-15 18:26:16 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Send instant-rate-change event if requested in the SEEK event
	  Short-circuit instant rate change events by generating
	  a downstream instant-rate-change event and doing no further
	  seek processing.

2020-03-10 23:16:00 +0900  Seungha Yang <seungha@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: Update for video-hdr struct change
	  See the change of -base https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/594

2020-03-31 15:51:27 -0400  Aaron Boxer <boxerab@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: make warning messages more meaningful

2020-03-27 19:24:03 +0100  Nicolas Pernas Maradei <nicopernas@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: rename RTCP thread
	  RTP session starts a new thread for RTCP and names it
	  "rtpsession-rtcp-thread" which happens to be longer than the maximum 16B
	  allowed by pthread_setname_np and causes the naming to fail.
	  See docs for more details.
	  This commit simply shortens the thread's name so it can actually be set.

2020-03-30 22:26:33 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: create specific API for appending buffers, events etc
	  To avoid specifying a bunch of mystic variables.

2020-02-10 17:33:54 +0100  Havard Graff <havard@pexip.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: various test-improvements
	  Mainly generalize all the latest tests that have found various stalls
	  in the jitterbuffer, so that they only consist of a series of packets
	  with various seqnum/rtptime/rtx combinations, arriving at a specific time.
	  This means future tests can be more easily written to prove certain
	  behavior does not cause stalls.
	  Also fix the warning on windows:
	  warning C4244: 'initializing': conversion from 'double' to 'gint', possible loss of data

2020-03-27 14:07:04 +0100  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix waiting timer/queue code
	  Changing the types from boolean to guint due to the ++ operand used on
	  them, and only call JBUF_SIGNAL_QUEUE after settling down,
	  or else you end up signaling the waiting code in chain() for every buffer
	  pushed out.

2020-03-23 19:55:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Error out instead of crashing if reserved-max-duration is 0 or no samples could be created in prefill mode

2020-03-12 15:16:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: Use GST_VIDEO_OVERLAY_COMPOSITION_BLEND_FORMATS for the supported formats
	  We don't do any blending by ourselves since a while now.
	  Note that this is a regression in "supported" formats: previously
	  ARGB64 was supported, for example, but in practice it caused blending to
	  not take place at all.

2020-03-24 00:23:24 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multifile/gstsplitmuxsrc.h:
	  splitmuxsrc: Fix some deadlock conditions and a crash
	  When switching the splitmuxsrc state back to NULL quickly, it
	  can encounter deadlocks shutting down the part readers that
	  are still starting up, or encounter a crash if the splitmuxsrc
	  cleaned up the parts before the async callback could run.
	  Taking the state lock to post async-start / async-done messages can
	  deadlock if the state change function is trying to shut down the
	  element, so use some finer grained locks for that.

2020-03-24 00:18:54 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/splitmux.c:
	  splitmux: Make the unit test faster
	  The playback test is considerably faster if it runs with the
	  appsink set to sync=false

2020-03-25 22:14:36 +0900  Seungha Yang <seungha@centricular.com>

	* tests/check/elements/splitmux.c:
	* tests/check/meson.build:
	  tests: splitmux: Add test for timecode based split

2020-03-25 21:20:07 +0900  Seungha Yang <seungha@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Split fragment only if queued time is larger than threshold
	  The queued time includes the duration of the last queued frame
	  (i.e., new keyframe) so the condition check should not be inclusive.
	  Note that the new fragment will be cut excluding the last frame
	  and therefore if the condition is inclusive way,
	  the fragment might have one frame shorter duration for all keyframe
	  stream such as jpeg or all-inter video streams.

2020-03-25 21:01:00 +0900  Seungha Yang <seungha@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't need to trace next timecode for split decision
	  Since the commit 94bb76b6b9c48981d3ad42a8c4370b9658db4229, splitmuxsink
	  will split fragments based on queued time and the threshold of that.
	  So don't need to store the next timecode for split decision.

2018-08-08 09:27:19 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2: add alternate interlace mode
	  When using this mode each frame is split in two fields, each one being
	  transferred using its own buffer.
	  This is implemented with the V4L2_FIELD_ALTERNATE field format in v4l2.
	  This mode is enabled using a caps filter such as
	  "v4l2src ! video/x-raw\(format:Interlaced\)"
	  Here are the main changes related to this feature:
	  - use the INTERLACED caps feature with this mode.
	  - in this mode both fields of a given frame have the same sequence/offset
	  so adjust the algorithm checking for lost field/frame accordingly.
	  - double pool's min number of buffers as each frame requires 2 buffers.
	  Fix #504
	  Co-authored-by: Zeeshan Ali <zeenix@collabora.co.uk>

2020-02-05 13:03:51 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: display field when setting or trying format
	  Ease debugging interlacing pipelines.

2020-01-30 12:35:02 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: pass v4l2object to GST_V4L2_MIN_BUFFERS()
	  Will be used to double the number of buffers in alternate interlace
	  mode.

2020-01-30 12:09:12 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: use GST_VIDEO_INFO_FIELD_HEIGHT()
	  Use GST_VIDEO_INFO_FIELD_HEIGHT() instead of GST_VIDEO_INFO_HEIGHT()
	  when we actually want the field height rather than the frame height.
	  So far both are equals but that won't longer be the case when
	  implementing alternate interlace mode.

2020-03-24 22:08:27 +0900  Seungha Yang <seungha@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Mark some split decision related properties as MUTABLE_READY
	  The change of various criteria for split decision while muxing is on progress
	  wouldn't work well as expected.

2020-03-24 13:45:00 +0900  Seungha Yang <seungha@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Take account queued time and max-size-timecode for split decision
	  Not only the requested keyframe time, the queued size should be
	  a criterion for the split decision of timecode based mode
	  (same as max-size-time based split case).

2020-03-24 12:55:27 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtoverlay.cc:
	  qmlgloverlay: fix usage without an qmlglsink in the pipeline
	  Without a qmlglsink, we need to retrieve the window system display
	  ourselves rather than relying solely on qmlglsink to have priority on
	  the choice of display.

2020-03-23 21:32:04 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/rtpmanager/rtptwcc.c:
	* gst/videocrop/gstvideocrop.c:
	* tests/check/elements/rtpbin.c:
	* tests/check/elements/rtpsession.c:
	  Fix usage of C99
	  It's 2020, way too early for that, let's stick to C89 for now.

2020-03-23 16:34:46 +0900  Seungha Yang <seungha@centricular.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2bufferpool: Use unique name for v4l2bufferpool object
	  Assign unique sequence number to an object name for better debugging

2020-03-23 14:02:22 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtglrenderer.cc:
	  qmlgloverlay: don't leak resources freed on a different GL thread
	  deleting a QOpenGLFrameBufferObject needs to occur on the same thread it
	  was created on in order to actually free the relevant resources
	  immediately.  Otherwise, they will be queued for deletion and not freed
	  until the associated QOpenGLContext is destroyed.

2020-03-20 09:14:01 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qt: reorganize include defines

2020-03-19 23:17:21 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtptimerqueue.c:
	* gst/rtpmanager/rtptimerqueue.h:
	* tests/check/elements/rtptimerqueue.c:
	  rtptimerqueue: remove ->num from the timer
	  This concept was only used by the "multi"-lost timer, and since that
	  one is not around any longer, the "num" concept is superfluous.

2020-03-19 23:37:26 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: remove the concept of "already-lost"
	  This is a concept that only applies when a buffer arrives in the chain
	  function, and it has already been scheduled as part of a "multi"-lost
	  timer.
	  However, "multi"-lost timers are now a thing of the past, making this
	  whole concept superflous, and this buffer is now simply counted as "late",
	  having already been pushed out (albeit as a lost-event).

2020-03-19 23:12:04 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: immediately insert a lost-event on multiple lost packets
	  There is a problem with the code today, where a single timer will
	  be scheduled for a series of lost packets, and then if the first packet
	  in that series arrives, it will cause a rescheduling of that timer, going
	  from a "multi"-timer to a single-timer, causing a lot of the packets
	  in that timer to be unaccounted for, and creating a situation in where
	  the jitterbuffer will never again push out another packet.
	  This patch solves the problem by instead of scheduling those lost packets
	  as another timer, it instead asks to have that lost-event pushed straight
	  out.
	  This very much goes with the intent of the code here: These packets are
	  so desperately late that no cure exists, and we might as well get the
	  lost-event out of the way and get on with it.
	  This change has some interesting knock-on effect being presented in
	  later commits. It completely removes the concept of "already-lost", so
	  that is why that test has been disabled in this commit, to be
	  removed later.

2020-03-19 23:03:50 +0100  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: rework large-gap tests
	  Make sure to set the time the buffer is supposed to arrive at, so
	  as not to trigger an artificial situation.

2020-03-19 12:17:22 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refactor lost_timeout code
	  Split it up in code related to the timer, (do_lost_timeout) and code
	  to insert a lost-item/event and update private jitterbuffer-variables.

2019-10-18 17:43:36 +0200  Havard Graff <havard@pexip.com>

	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/rtptimerqueue.c:
	* tests/check/meson.build:
	  test/check: split out rtptimerqueue-tests in a separate file

2020-02-05 09:56:23 +0100  Dmitriy Purgin <dmitriy.purgin@sequality.at>

	* ext/qt/qtplugin.pro:
	  gstqmlgl: Link to opengl32.lib on MinGW

2020-03-19 23:51:47 +0900  Seungha Yang <seungha@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix build warning
	  gstqtmux.c(644): warning C4133: '=':
	  incompatible types - from 'gboolean (__cdecl *)(GstAggregator *,GstAggregatorPad *,GstEvent *)'
	  to 'GstFlowReturn (__cdecl *)(GstAggregator *,GstAggregatorPad *,GstEvent *)'

2020-03-19 23:05:49 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Reset cleanly for reuse
	  Reset the splitmuxsink completely when changing states so that
	  it can be reused.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1241

2020-02-17 22:37:10 -0600  Zebediah Figura <z.figura12@gmail.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstmpegaudioparse.h:
	  mpegaudioparse: Use a constant bit rate to convert between time and bytes if possible.
	  This should result in no worse accuracy than the base parse element, and may
	  result in better accuracy. In particular, the number of bytes processed at any
	  given point, as accumulated by baseparse, can be only accurate to
	  (1 / # of frames) bytes per second, and if we try to seek immediately after
	  pausing the pipeline to a large offset, this small inaccuracy can propagate to
	  something noticeable.
	  The use case that prompted this patch is a 45-minute MPEG-1 layer 3 file, which
	  has a constant bit rate but no seek tables. Trying to seek the pipeline
	  immediately after pauisng it, without the ACCURATE flag, to a location 41
	  minutes in, yields a location that is, even with <https://gitlab.freedesktop.org/gstreamer/gstreamer/merge_requests/374>,
	  still audibly incorrect. This patch yields a much closer position, no longer
	  audibly incorrect, and likely within a frame of the most correct position.

2020-03-04 22:10:40 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: fix renegotiation check
	  By the time sink_event is called, the pad's current caps have
	  already been updated. To address this, implement sink_event_pre_queue,
	  and check if the pad can be renegotiated there.
	  Fixes #707

2020-03-12 20:34:47 +0900  Seungha Yang <seungha@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* tests/check/elements/splitmux.c:
	  splitmuxsink: Decouple keyframe request and the decision for fragmentation
	  Split the decision for keyframe request and fragmentation in order to
	  ensure periodic keyframe request.

2020-02-26 18:29:06 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.cc:
	* ext/qt/gstqtoverlay.cc:
	* ext/qt/qtglrenderer.cc:
	* ext/qt/qtglrenderer.h:
	* ext/qt/qtitem.cc:
	* tests/examples/qt/qmloverlay/main.cpp:
	* tests/examples/qt/qmloverlay/overlay2.qml:
	* tests/examples/qt/qmloverlay/qmloverlay.qrc:
	  qt: add a qml overlay filter element [part 2]
	  It takes a qml scene description and renders it using a possible input
	  stream.
	  Currently supported on GLX and WGL.
	  Follow up to (as that MR had an old version of the commit):
	  - https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/475
	  - 4778d7166a02caf793df4f845dc35b6933d87c81: qt: add a qml overlay filter element

2020-02-26 18:29:06 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstplugin.cc:
	* ext/qt/gstqtglutility.cc:
	* ext/qt/gstqtglutility.h:
	* ext/qt/gstqtoverlay.cc:
	* ext/qt/gstqtoverlay.h:
	* ext/qt/meson.build:
	* ext/qt/qtglrenderer.cc:
	* ext/qt/qtglrenderer.h:
	* tests/examples/qt/meson.build:
	* tests/examples/qt/qmloverlay/main.cpp:
	* tests/examples/qt/qmloverlay/main.qml:
	* tests/examples/qt/qmloverlay/meson.build:
	* tests/examples/qt/qmloverlay/overlay.qml:
	* tests/examples/qt/qmloverlay/qmloverlay.qrc:
	  qt: add a qml overlay filter element
	  It takes a qml scene description and renders it using a possible input
	  stream.
	  Currently supported on GLX and WGL.

2020-02-25 21:47:14 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqsgtexture.cc:
	* ext/qt/qtitem.cc:
	  qt: don't always activate/deactivate our GstGLContext
	  Techincally it is enough to activate at the beginning and then forget.

2020-02-04 19:43:52 +1100  Matthew Waters <matthew@centricular.com>

	* tests/examples/qt/meson.build:
	* tests/examples/qt/qmlsink-dynamically-added/.gitignore:
	* tests/examples/qt/qmlsink-dynamically-added/main.cpp:
	* tests/examples/qt/qmlsink-dynamically-added/main.qml:
	* tests/examples/qt/qmlsink-dynamically-added/meson.build:
	* tests/examples/qt/qmlsink-dynamically-added/play.pro:
	* tests/examples/qt/qmlsink-dynamically-added/qmlsink.qrc:
	  test/qml: add an dynamically adding qmlglsink element
	  The example shows how to add qmlglsink to an already running pipeline
	  with pre-existing OpenGL elements.

2020-02-04 19:40:45 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtsink.cc:
	  qmlglsink: propagate the context up the the application
	  Allows the application to be notified of the OpenGL context creation.

2020-02-03 15:59:34 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  qtitem: fix leak of caps

2020-03-15 19:28:18 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/wavpack/gstwavpackdec.h:
	* ext/wavpack/gstwavpackenc.h:
	  wavpack: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:26:18 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp8enc.h:
	* ext/vpx/gstvp9dec.h:
	* ext/vpx/gstvp9enc.h:
	* ext/vpx/gstvpxdec.h:
	* ext/vpx/gstvpxenc.h:
	  vpx: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:22:00 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/twolame/gsttwolamemp2enc.h:
	  twolame: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:20:49 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/taglib/gstapev2mux.h:
	* ext/taglib/gstid3v2mux.h:
	  taglib: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:18:39 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/speex/gstspeexdec.h:
	* ext/speex/gstspeexenc.h:
	  speex: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:16:22 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/soup/gstsouphttpclientsink.h:
	  soup: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:14:17 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:11:52 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/raw1394/gst1394clock.h:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gstdv1394src.h:
	* ext/raw1394/gsthdv1394src.h:
	  raw1394: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:06:50 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/qt/gstqtsink.h:
	* ext/qt/gstqtsrc.h:
	  qt: Use G_DECLARE_FINAL_TYPE

2020-03-15 19:00:18 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/pulse/pulsedeviceprovider.h:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.h:
	  pulse: Use G_DECLARE_FINAL_TYPE

2020-03-15 18:54:33 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/mpg123/gstmpg123audiodec.h:
	  mpg123: Use G_DECLARE_FINAL_TYPE

2020-03-15 18:52:57 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/libpng/gstpng.h:
	* ext/libpng/gstpngdec.h:
	* ext/libpng/gstpngenc.h:
	  libpng: Use G_DECLARE_FINAL_TYPE

2020-03-15 18:49:53 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/libcaca/gstcacasink.h:
	* ext/libcaca/gstcacatv.h:
	  libcaca: Use G_DECLARE_FINAL_TYPE

2020-03-15 18:40:28 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/lame/gstlamemp3enc.h:
	  lame: Use G_DECLARE_FINAL_TYPE

2020-03-14 17:52:38 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.h:
	  jack: Use G_DECLARE_FINAL_TYPE

2020-03-14 17:43:50 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/gtk/gstgtkbasesink.h:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gstgtksink.h:
	  gtk: Use G_DECLARE_FINAL_TYPE

2020-03-13 18:47:49 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.h:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	* ext/gdk_pixbuf/gstgdkpixbufsink.h:
	  gdk_pixbuf: Use G_DECLARE_FINAL_TYPE

2020-03-13 18:42:38 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.h:
	* ext/flac/gstflactag.h:
	  flax: Use G_DECLARE_FINAL_TYPE

2020-03-13 18:39:38 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/dv/gstdvdec.h:
	* ext/dv/gstdvdemux.h:
	  dv: Use G_DECLARE_FINAL_TYPE

2020-03-12 19:24:57 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/cairo/gstcairooverlay.h:
	  cairo: Use G_DECLARE_FINAL_TYPE

2020-03-12 19:20:42 +0100  Niels De Graef <nielsdegraef@gmail.com>

	* ext/aalib/gstaasink.h:
	* ext/aalib/gstaatv.h:
	  aalib: Use G_DECLARE_FINAL_TYPE

2020-03-12 16:55:44 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: add minimal vp8/vp9 rtp payloading/depayloading test

2018-10-19 16:17:17 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp9pay.c:
	  rtpvp8pay, rtpvp9pay: fix caps leak in set_caps()

2020-03-12 11:22:56 +0100  Edward Hervey <edward@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't leak peer caps

2020-02-11 16:19:15 -0300  Thibault Saunier <tsaunier@igalia.com>

	* docs/gst_plugins_cache.json:
	* gst/multifile/gstimagesequencesrc.c:
	* gst/multifile/gstimagesequencesrc.h:
	* gst/multifile/gstmultifile.c:
	* gst/multifile/meson.build:
	  imagesequencesrc: Cleanup and add some features
	  * Implement the GstURIHandlerInterface
	  * Rework the locking
	  * Implement backward seeking handling
	  * Generate documentation

2016-04-10 02:25:32 +0000  Fabian Orccon <fabian.orccon@pucp.pe>

	* gst/multifile/gstimagesequencesrc.c:
	* gst/multifile/gstimagesequencesrc.h:
	  Add an imagesequencesrc element to stream sequence of images
	  See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/121

2020-03-05 08:55:44 -0800  Gordon Hart <gordon.hart2@gmail.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: decrease gst_v4l2src_create log verbosity
	  Lower the verbosity of the 'sync' log message emitted
	  each buffer from gst_v4l2src_create down to LOG(6)
	  from INFO(4). This brings the logging behavior of
	  v4l2src closer to the GStreamer guidelines, which
	  recommend the INFO level be reserved for rare or
	  one-off messages.

2020-03-10 17:19:46 +0800  yychao <yychao@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add support for AC4
	  The caps received from qtdemux for AC-4 content are audio/x-gst-fourcc-ac_4
	  Based on patch by: Savinderjit Kaur
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/413

2020-03-10 21:07:12 +1100  Matthew Waters <matthew@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: handle reconfigure events on the srcpad

2020-03-05 22:47:16 +1100  Matthew Waters <matthew@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: properly ignore setting caps failures
	  Ignore the return value of gst_pad_set_caps() so that setcaps will set a
	  framerate that is usable.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/705

2020-03-05 22:45:32 +1100  Matthew Waters <matthew@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: don't fail sending sticky events downstream
	  They will be repropagated anyway.

2020-03-09 23:31:09 +0100  Markus Ebner <info@ebner-markus.de>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: Add support for Y41B and Y42B

2020-03-09 23:25:03 +0100  Markus Ebner <info@ebner-markus.de>

	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	  videocrop: Add support for Y444
	  - Refactored the planar transform method to support all video formats
	  that are stored planar, independent of the used subsampling
	  - Added support for Y444

2020-03-09 23:23:50 +0100  Markus Ebner <info@ebner-markus.de>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: Use G_VALUE_INIT to initialize GValues

2020-02-28 19:35:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Configure JPEG chroma-siting for YUV formats

2020-02-06 09:23:24 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtp/gstbuffermemory.c:
	* gst/rtp/gstbuffermemory.h:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/meson.build:
	* tests/check/elements/rtph264.c:
	  rtph26x: Use gst_memory_map() instead of gst_buffer_map() in avc mode
	  gst_buffer_map () results in memcopying when a GstBuffer contains
	  more than one GstMemory and when AVC (length-prefixed) alignment is used.
	  This has quite an impact on performance on systems with limited amount of
	  resources. With this patch the whole GstBuffer will not be mapped at once,
	  instead each individual GstMemory will be iterated and mapped separately.

2019-11-26 15:08:20 +0100  Milian Wolff <milian.wolff@kdab.com>

	* ext/qt/gstqtgl.h:
	  qmlgl: ensure Qt defines GLsync to fix compile on some platforms
	  By explictly including QtGui/qopengl.h we force the code path that
	  defines GLsync in the Qt-specific way. Without that, some platforms
	  failed to compile the qmlgl plugin, since neither Qt nor gstreamer
	  defined GLsync then, leading to e.g.:
	  ```
	  make[4]: Entering directory '/.../gst-plugins-good-1.16.1/ext/qt'
	  CXX      libgstqmlgl_la-qtitem.lo
	  In file included from gstqtgl.h:32,
	  from qtitem.h:27,
	  from qtitem.cc:28:
	  /.../usr/include/gstreamer-1.0/
	  gst/gl/gstglfuncs.h:93:17: error: expected identifier before ‘*’ token
	  ret (GSTGLAPI *name) args;
	  ^
	  /.../usr/include/gstreamer-1.0/
	  gst/gl/glprototypes/sync.h:27:1: note: in expansion of macro
	  ‘GST_GL_EXT_FUNCTION’
	  GST_GL_EXT_FUNCTION (GLsync, FenceSync,
	  ^~~~~~~~~~~~~~~~~~~
	  ```

2020-03-02 13:50:55 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtptwcc.c:
	* gst/rtpmanager/rtptwcc.h:
	  rtptwcc: make RTPTWCCManager a GObject

2020-03-04 11:17:16 +0100  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix stalling when resetting timers
	  When calling gst_rtp_jitter_buffer_reset you pass in a seqnum.
	  This is considered the starting-point for a new stream.
	  However, the old behavior would unref this buffer, basically lying to
	  the thread that is pushing out buffers saying that it can expect
	  this buffer, when it would never arrive. The resulting effect being no
	  more buffer pushed out of the jitterbuffer, and it would buffer
	  incoming data indefinitely.
	  By instead inserting the buffer in the gap_packets queue, the _reset()
	  function will take responsibility for using that as the first buffer
	  of the new stream.
	  Fixes #703

2020-02-21 02:14:11 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	* gst/multifile/gstsplitmuxsrc.c:
	* tests/check/elements/splitmux.c:
	  splitmux: Avoid negative DTS
	  In order to concatenate fragments, splitmuxsrc offsets
	  the start of each fragment PTS to 0 to align it with the
	  previous file. This means that DTS can go negative for
	  the first fragment, with really bad results.
	  Add a fixed offset to outgoing timestamp ranges to
	  avoid that.

2020-03-04 03:43:51 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Remove warning in the log for mono video
	  Vanilla mono video was generating a spurious warning into the debug log
	  that's just misleading. Handle mono caps explicitly to avoid the warning.

2020-01-27 12:29:18 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: add alternate support
	  In this mode each field is carried using its own buffer.
	  Allow deinterlace to negotiate caps with the Interlaced feature and
	  adjust the algorithm fetching lines.
	  Fix #620

2020-02-03 13:08:39 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: add wrapper to get field lines from history
	  No semantic change so far, will be used to implement alternate support.

2020-02-04 16:48:21 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: stop checking line index boundaries
	  The LINE2() macro already prevents out of bound indexes using CLAMP_HI()
	  and CLAMP_LOW().

2020-01-20 12:30:12 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: fix video info on output frames
	  Output frames used to have their interlace mode set to the same one as
	  the input. This breaks their field and comp heights when deinterlacing
	  an alternate stream.

2020-01-14 14:51:07 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: use output caps to compute buffer size
	  In interlace-mode=alternate the input buffers have half the size of the
	  output ones as each field has its own buffer.

2020-02-29 08:10:56 -0500  Jennifer Berringer <berringerjennifer@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix broken reordering of flac metadata
	  Each FLAC metadata block starts with a flag denoting whether it is the
	  last metadata block. The existing flacparse code moves any existing
	  VORBISCOMMENT block to immediately follow the STREAMINFO block without
	  changing any block's last-metadata-block flag. If no VORBISCOMMENT block
	  exists, it created one with the last-metadata-block flag set to true.
	  This results in gstflacdec sometimes giving bad headers to libflac when
	  trying to play perfectly valid FLAC files depending on the file's
	  metadata ordering. Depending on the contents of the other metadata
	  blocks, current versions of libflac may or may not return
	  FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER when given this broken
	  metadata. This is most noticeable with files that have a large cover art
	  image attached where VORBISCOMMENT is the very last metadata block with
	  no PADDING afterwards.
	  This patch changes that behavior so that:
	  1. For FLAC files that already have a VORBISCOMMENT block, the metadata
	  order is preserved.
	  2. For FLAC files that do not have a VORBISCOMMENT block, the generated
	  dummy VORBISCOMMENT is placed immediately after STREAMINFO and
	  inherits the last-metadata-block flag from STREAMINFO.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/484

2020-02-27 14:50:51 +0900  Yeongjin Jeong <yeongjin.jeong@navercorp.com>

	* tests/check/elements/flvmux.c:
	  tests: flvmux: Instead of using the testclock, just send eos event for drain
	  When using the testclock for determining clock in test, it is sometimes observed
	  that the clock entry is not registered in time by the aggregator. So deadlock occurs
	  between the aggregator and the test thread.

2020-02-28 14:23:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Try to infer useful header values for raw audio if the sound sample descriptions contain zero values

2020-02-28 14:00:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Also use the enda atom for determining endianess of in32, fl32 and fl64 formats
	  Previously it was only used for in24.

2020-02-28 13:59:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix up header information for various fixed-format raw audio formats
	  Sometimes the headers contain useless, wrong or zero values for e.g. the
	  sample size with these formats. There's only a single valid value for
	  them so let's set these instead.

2020-02-28 13:59:06 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't print "unhandled type" warnings for various other raw audio fourccs

2020-02-28 13:57:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Add some more raw audio fourccs to the header instead of duplicating them

2020-02-25 21:14:54 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Don't use glib format modifiers with sscanf
	  We do not have a way to know the format modifiers to use with string
	  functions provided by the system. G_GUINT64_FORMAT and other string
	  modifiers only work for glib string formatting functions. We cannot
	  use them for string functions provided by the stdlib. See:
	  https://developer.gnome.org/glib/stable/glib-Basic-Types.html#glib-Basic-Types.description
	  ```
	  ../gst/rtpmanager/gstrtpjitterbuffer.c: In function 'gst_jitter_buffer_sink_parse_caps':
	  ../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: unknown conversion type character 'l' in format [-Werror=format=]
	  || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
	  ^~~~~~~~~~
	  In file included from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/gtypes.h:32,
	  from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/galloca.h:32,
	  from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib.h:30,
	  from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/gst.h:27,
	  from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/rtp/gstrtpbuffer.h:27,
	  from ../gst/rtpmanager/gstrtpjitterbuffer.c:108:
	  /home/nirbheek/cerbero/build/dist/windows_x86/lib/glib-2.0/include/glibconfig.h:69:28: note: format string is defined here
	  #define G_GUINT64_FORMAT "llu"
	  ^
	  ../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: too many arguments for format [-Werror=format-extra-args]
	  || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
	  ^~~~~~~~~~
	  ```
	  See also: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/379

2020-02-24 15:25:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Add support for 8k resolutions in prefill mode with ProRes

2020-02-25 11:06:43 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtptimerqueue.c:
	  rtpjitterbuffer: Include string.h for memcpy() / memset()
	  Usually something else is pulling it in somehow already, but not on
	  Windows.

2020-02-24 13:06:27 +0000  Håvard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: fix crash when no extension-header present for twcc

2020-02-21 09:34:30 +0100  Johan Bjäreholt <johanbj@axis.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Fix incorrect rounding of timestamps
	  Previously we saved the buffer_timestamp straight into
	  mux->cluster_time. Since the cluster time saved into the file does not
	  have as high precision as GstClockTime depending on the timecodescale
	  the rounding of relative_timestamp was invalid as mux->cluster_time
	  which it was calculated relative to was not equal to the cluster time
	  written to the matroska file.
	  Example of "mkvinfo -v" of how it looks before and after this change in
	  an scenario where previously timestamps got out of order because of this
	  issue.
	  Notice the timestamp of the SimpleBlock right before and right after the
	  Cluster now being in order. The consequence of this however is that the
	  cluster timestamp is not necessarily the same as the timestamp of the
	  first buffer in the cluster however (in case it's rounded up).
	  Before
	  | + SimpleBlock (track number 1, 1 frame(s), timecode 126.922s = 00:02:06.922)
	  |  + Frame with size 432
	  | + SimpleBlock (track number 2, 1 frame(s), timecode 126.933s = 00:02:06.933)
	  |  + Frame with size 329
	  | + SimpleBlock (track number 2, 1 frame(s), timecode 126.955s = 00:02:06.955)
	  |  + Frame with size 333
	  |+ Cluster
	  | + Cluster timecode: 126.954s
	  | + Cluster previous size: 97344
	  | + SimpleBlock (key, track number 1, 1 frame(s), timecode 126.954s = 00:02:06.954)
	  |  + Frame with size 61239
	  | + SimpleBlock (track number 2, 1 frame(s), timecode 126.975s = 00:02:06.975)
	  |  + Frame with size 338
	  After
	  | + SimpleBlock (track number 1, 1 frame(s), timecode 135.456s = 00:02:15.456)
	  |  + Frame with size 2260
	  | + SimpleBlock (track number 2, 1 frame(s), timecode 135.468s = 00:02:15.468)
	  |  + Frame with size 332
	  | + SimpleBlock (track number 2, 1 frame(s), timecode 135.490s = 00:02:15.490)
	  |  + Frame with size 335
	  |+ Cluster
	  | + Cluster timecode: 135.489s
	  | + Cluster previous size: 158758
	  | + SimpleBlock (key, track number 1, 1 frame(s), timecode 135.490s = 00:02:15.490)
	  |  + Frame with size 88070
	  | + SimpleBlock (track number 2, 1 frame(s), timecode 135.511s = 00:02:15.511)
	  |  + Frame with size 336

2020-02-19 15:59:19 +1100  Jake Barnes <jake@vivi.io>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Fix cookies property
	  Disable session sharing and cookie jar when cookies property is set.
	  The cookie jar actually replaces or removes any existing Cookie header
	  set on the message, so the cookies property was effectively being
	  ignored. There doesn't appear to be a way to inject the cookies into the
	  jar without having to specify matching domains etc., so it's not
	  possible to simulate the old behaviour of unconditionally sending the
	  cookies with all messages, besides simply disabling the cookie jar.

2020-02-20 09:06:10 +0100  Stefano Buora <sbuora@marchnetworks.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove useless function calls
	  Comparing gst_rtspsrc_loop_interleaved and gst_rtspsrc_loop_udp, and investigating on timeout issues, it sounds like a piece of code has been originally copied from udp to the interleaved one. The timeout variable is never used inside the interleaved one. No side effect has been seen in the removed function calls.
	  The debug message removed is pointless as the timeout used is "src->tcp_timeout" that is fixed.
	  The presence of the two timeout drove my team in investigating if the reference to the tcp_timeout was correct (it is). Hence we removed the misleading reference to the local timeout variable.

2020-02-20 13:43:13 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix typo setting max-dropout/misorder-time
	  we were setting the max-dropout-time to the value of the
	  max-misorder-time which by default has a factor of 30 difference in
	  value.

2020-02-19 20:27:54 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Parse VP Codec Configuration Box
	  The VP Codec Configuration Box (vpcC) contains vp9 profile and
	  colorimetry information. Especially the profile information might
	  be useful for downstream to select capable decoder element.

2020-02-18 18:36:36 +0900  Yeongjin Jeong <yeongjin.jeong@navercorp.com>

	* tests/check/elements/flvmux.c:
	  tests: flvmux: Add test for rollover timestamp
	  The timestamps that exceed uint32 maximum value should be handled to rollover.

2020-02-18 14:58:00 +0900  Yeongjin Jeong <yeongjin.jeong@navercorp.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Support rollover in timestamp
	  For live streams, if we keep the stream for a long time, the timestamp
	  will be larger than max_uint32. In that case, timestamp should be handled
	  as a rollover timestamp rather than a backward timestamp.

2020-02-17 15:03:28 +0100  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: don't use the timer-object after JBUF_UNLOCK
	  It could have been freed (rtp_timer_free) in the meantime.

2019-06-29 18:06:11 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/meson.build:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	* gst/rtpmanager/rtptwcc.c:
	* gst/rtpmanager/rtptwcc.h:
	* tests/check/elements/rtpsession.c:
	  rtpmanager: Google Transport-Wide Congestion Control RTP Extension
	  Generating and parsing the RTCP-messages described in:
	  https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01

2020-02-14 10:08:05 +0000  Håvard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	* tests/check/elements/rtpfunnel.c:
	  rtpfunnel: various cleanups
	  * Organize GstRtpFunnelPad and GstRtpFunnel separately
	  * Use G_GNUC_UNUSED instead of (void) casts
	  * Don't call an event "caps"
	  * Use semicolons after GST_END_TEST (helps gst-indent)

2020-01-29 23:51:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Merge sample tables for raw audio streams with one container sample per audio sample
	  Instead of having chunks with one sample per raw audio sample, have
	  chunks with a single sample that contains lots of raw audio samples. If
	  necessary these are still split again later when reading the stream.
	  With this we are allocating a lot less memory for the parsed sample
	  tables and can play files that previously triggered our limit of 200MB
	  for the sample table. For example, one file here would previously
	  allocate 3.5GB for the sample table and now only allocates 70KB.

2020-01-13 11:55:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add a minimum buffer size for raw audio to not output one buffer per frame
	  Outputting 48000 buffers per second is not a good idea performance-wise.
	  If a container sample is less than 1024 raw audio frames, combine
	  multiple samples to get at least 1024 raw audio samples as long as
	  they're stored contiguous in the file.
	  For the other direction, if a container sample contains more than 4096
	  samples there is already code for splitting them up.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692750

2020-02-11 21:52:41 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix requested range
	  When the server replies with a range "now-", it is presumed to
	  be a "live" stream and we should request a similar range.
	  This was the case prior to my refactoring to make use of
	  gst_rtsp_range_to_string in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7,
	  this commit restores the behaviour for that case.

2017-07-13 13:49:07 +0200  Mikhail Fludkov <misha@pexip.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	* tests/check/elements/rtpptdemux.c:
	* tests/check/meson.build:
	  rtpptdemux: set payload to caps inside gst_rtp_pt_demux_get_caps
	  Refactoring to remove duplicate code and add test

2017-03-16 20:57:54 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: Fix debug to use GST_DEBUG_OBJECT

2016-09-14 16:49:26 +0200  Mikhail Fludkov <misha@pexip.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: use max-streams on rtpssrcdemux
	  The proper way of capping on max-streams is to do it in rtpssrcdemux.
	  This patch uses the newly introduced property on rtpssrcdemux. Previous
	  behavior would not prevent rtpssrcdemux spawning new pads for every new
	  ssrc and potentialy causing performance trouble during teardown.

2017-01-18 14:32:03 +0000  John Bassett <john.bassett@pexip.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* tests/check/elements/rtpssrcdemux.c:
	  rtpssrcdemux: Handle RTCP APP packets
	  Fix crash when processing RTCP APP packets.

2017-01-12 16:05:59 +0000  John Bassett <john.bassett@pexip.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* tests/check/elements/rtpssrcdemux.c:
	  rtpssrcdemux: Bad RTP/RTCP packet is not fatal
	  When used for processing bundled media streams within rtpbin the rtpssrcdemux element may
	  receive bad RTP and RTCP packets, these should not be treated as a fatal error.

2016-09-14 16:41:02 +0200  Mikhail Fludkov <misha@pexip.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	* tests/check/elements/rtpssrcdemux.c:
	  rtpssrcdemux: introduce max-streams property
	  The property is useful against atacks when the sender changes SSRC for
	  every RTP packet. The property with the same name introduced in rtpbin
	  was not enough, because we still can end up with thousands of pads
	  allocated in rtpssrcdemux.

2020-02-10 14:22:47 +0100  Havard Graff <havard@pexip.com>

	* tests/check/elements/rtpssrcdemux.c:
	  rtpssrcdemux: fix test warnings

2020-02-07 10:03:49 +0100  Alexander Lapajne <alexander.lapajne@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix for segmentation fault when handling set/get_parameter requests
	  gstrtspsrc uses a queue, set_get_param_q, to store set param and get
	  param requests. The requests are put on the queue by calling
	  get_parameters() and set_parameter(). A thread which executs in
	  gst_rtspsrc_thread() then pops requests from the queue and processes
	  them. The crash occured because the queue became empty and a NULL
	  request object was then used. The reason that the queue became empty
	  is that it was popped even when the thread was NOT processing a get
	  parameter or set parameter command. The fix is to make sure that the
	  queue is ONLY popped when the command being processed is a set
	  parameter or get parameter command.

2019-09-27 16:52:06 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsource.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Add test for packet rate maths

2019-09-10 19:03:02 +0100  olivier.crete@collabora.com <Olivier Crête>

	* gst/rtpmanager/rtpstats.c:
	  rtpstats: Base the packet rate average on the packet rate itself
	  Do this so that the average update speed is in time instead of varying
	  based on the actual packet arrival rate.

2019-09-10 18:59:02 +0100  olivier.crete@collabora.com <Olivier Crête>

	* gst/rtpmanager/rtpstats.c:
	  rtpstats: Don't save the ts & seqnum if the avg is not updated
	  This makes it update correctly when you have more than one packet per
	  frame.

2020-02-05 12:48:45 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: map GST_VIDEO_FORMAT_BGR15
	  The GstVideoFormat to v4l2 conversion was missing for BGR15.

2020-02-05 12:00:00 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix crash on invalid caps
	  gst_v4l2_object_set_format_full() was returning FALSE without setting
	  an error. Caller code (gst_v4l2src_fixate()) was then derefing a
	  NULL pointer when trying to handle the error.

2020-01-27 16:00:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Include actual sink element in the fragment-opened/closed messages
	  If not configuring the sinks via the "location" property this can be
	  useful to know for which sink the fragment was actually opened/closed,
	  especially if finalization of the fragments is happening asynchronously.

2020-01-29 12:05:07 +0100  Juergen Werner <pogojotz@gmx.net>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix scaling from RTP-time to NTP-time
	  The scaling was inverse.

2020-01-27 23:59:05 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	* tests/check/elements/rtprtx.c:
	  rtprtxsend: allow generic input caps
	  When connected to an upstream rtpfunnel element, payload-type,
	  ssrc and clock-rate will not be present in the received caps.
	  rtprtxsend can already deal with only the clock rate being
	  present there, a new property is exposed to allow users to
	  provide a payload-type -> clock-rate map, this enables the
	  use of the max-size-time property for bundled streams.

2020-01-27 15:17:27 -0800  Julien Isorce <jisorce@oblong.com>

	* ext/vpx/gstvpxenc.c:
	  vp8enc/vp8enc: set 1 for the default value of VP8E_SET_STATIC_THRESHOLD
	  In Google webrtc, the setting VP8E_SET_STATIC_THRESHOLD is set to 1
	  (except when the content is known to be static very often in which
	  case it is set to 100, i.e. when sharing screen with Google Hangouts).
	  The cpu usage drops a lot when using 1 for above setting because it
	  allows the encoder to skip static/low content blocks. The current
	  0 default value uses too much cpu and confuses the user regarding
	  the cpu usage expectations. User expects vp8enc to use low cpu by
	  default.
	  Documentation of VP8E_SET_STATIC_THRESHOLD:
	  https://github.com/webmproject/libvpx/blob/master/vpx/vp8cx.h#L188
	  chromium/webrtc:
	  https://chromium.googlesource.com/external/webrtc/+/b484ec0082948ae086c2ba4142b4d2bf8bc4dd4b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc#822
	  Closes #58

2020-01-27 17:16:02 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Check return value of gst_buffer_map()
	  Without this check, the element will crash instead of returning an
	  error.

2020-01-27 15:52:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Check the correct sink class for the existence of the "location" property

2020-01-13 11:58:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Always prefer information from v1/v2 sound sample description over sample description entry
	  ffmpeg is doing the same and various files in the wild have bogus
	  information in the sample description if the same information is also
	  duplicated afterwards in the v1/v2 sound sample desription.
	  Previously we only did this for non-raw audio due to
	  https://bugzilla.gnome.org/show_bug.cgi?id=374914
	  but this specific file is already worked around differently. It still
	  works after this change.
	  Also remove ad-hoc GST_READ_DOUBLE_BE re-implementation and move the
	  switch for legacy audio formats after reading all the sample
	  descriptions as we want to override the values from there.

2020-01-13 20:02:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avimux: Add support for >2 raw audio channels
	  For this case write a WAVEFORMATEXTENSIBLE header and also reorder the
	  raw audio channels to the AVI channel order if needed.

2020-01-13 20:07:01 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Fix writing of the channel mask with >2 channels
	  The channel position is an enum but the conversion code assumed it's a
	  mask. Convert accordingly.

2020-01-10 16:30:33 +0100  Kristofer Björkström <kristofb@axis.com>

	* gst/rtp/gstrtph265pay.c:
	* tests/check/elements/rtph265.c:
	  rtph265pay: TID for NALU type 48 was always set to 7
	  A typo bug: | instead of & resulted in TID alwasy being set to 7
	  for the aggregated NALU of type 48

2020-01-10 14:54:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: Add support for replacing the output buffer
	  By default imagefreeze will still reject new buffers after the first one
	  and immediately return GST_FLOW_EOS but the new allow-replace property
	  allows to change this.
	  Whenever updating the buffer we now also keep track of the configured
	  caps of the buffer and from the source pad task negotiate correctly
	  based on the potentially updated caps.
	  Only the very first time negotiation of a framerate with downstream is
	  performed, afterwards only the caps themselves apart from the framerate
	  are updated.

2020-01-09 18:43:02 +0000  Alicia Boya García <ntrrgc@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix race on pad reconnection
	  Elements emitting frames through several srcpads should use a
	  flow combiner to aggregate the chain returns and therefore only return
	  GST_FLOW_NOT_LINKED to upstream when all the downstream pads have
	  received GST_FLOW_NOT_LINKED.
	  In addition to that, in order to handle pads being relinked downstream,
	  the flow combiner should be reset in response to RECONFIGURE events.
	  This ensures that a both srcpads process a chain operation before a
	  GST_FLOW_NOT_LINKED can be propagated upstream (which would usually stop
	  the pipeline).
	  Otherwise, in a configuration with two srcpads, only one linked at a
	  time, after the relink the element could chain data through the now
	  unlinked pad and the flow combiner would resolve as GST_FLOW_NOT_LINKED
	  (stopping the pipeline) just because the now linked pad has not been
	  chained yet to update the flow combiner.
	  This patch adds handling of RECONFIGURE events to qtdemux. Also, since
	  this event handling causes the flow combiner to be used from a thread
	  other than the qtdemux streaming thread, usages of the flow combiner
	  has been guarded by the object lock.

2020-01-07 01:20:24 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix assertion failure on set_property()
	  GValue might have null object.
	  (gst-inspect-1.0:10304): GStreamer-CRITICAL ...
	  gst_object_ref_sink: assertion 'object != NULL' failed

2020-01-03 15:16:02 +0100  Daniel Molkentin <daniel@molkentin.de>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: allow properties to be animated by GstController

2019-12-24 08:24:51 -0500  Aaron Boxer <aaron.boxer@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: improved handling of control concatenation with base
	  Also, `control_url` variable has been renamed to `control_path`,
	  as it is actually a path.

2019-12-06 12:34:15 -0500  Aaron Boxer <aaron.boxer@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: append aggregate control string to base URL before query string
	  Appending control string to end of query changes meaning of query string
	  Fixes #650

2019-12-28 23:01:19 +0000  Eric Marks <bigmarkslp@gmail.com>

	* ext/aalib/gstaasink.c:
	* ext/aalib/gstaatv.c:
	* ext/aalib/gstaatv.h:
	* ext/aalib/meson.build:
	* ext/libcaca/gstcacasink.c:
	* ext/libcaca/gstcacatv.c:
	* ext/libcaca/gstcacatv.h:
	* ext/libcaca/meson.build:
	  aasink & cacasink: add filter aatv & cacatv
	  Add transform filter capabilities to aasink and cacasink in the form of new elements aatv and cacatv.

2019-06-06 11:03:34 +0200  Niels De Graef <niels.degraef@barco.com>

	* gst/alpha/gstalpha.h:
	* gst/alpha/gstalphacolor.h:
	  alpha: Cleanup using G_DECLARE_FINAL_TYPE
	  We started depending on GLib 2.44, so we can clean up all the GObject
	  boilerplate macros.

2019-12-18 16:07:18 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* ext/shout2/gstshout2.c:
	* gst/multipart/multipartmux.c:
	* sys/ximage/gstximagesrc.c:
	  good: use of g_value_dup_string
	  Use helper method to get string from GValue.

2019-12-19 23:48:09 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	* tests/check/elements/rtpbin.c:
	  rtpbin: fix shutdown crash in rtpbin
	  The key is to make sure the jitterbuffer is set to NULL *before* the
	  ptdemux.
	  The race that existed would basically happen when ptdemux had reached
	  READY, and the jitterbuffer would then push a buffer, triggering a new
	  pad with a new payloadtype being added and ghosted to the rtpbin itself.
	  However, the srcpad of the ptdemux would now be inactive, and all the
	  sticky-event pushed on it would be swallowed, not allowing any to reach
	  the ghost-pad. Then the buffer in-flight would come to the ghostpad,
	  and we would assert that a buffer arrived before the necessary
	  events.
	  By simply re-ordering the state-changes, we ensure that there will be
	  no buffer racing into the ptdemux while its state is being changed,
	  and the problem disappears completely.
	  Notice also that there is not point in disconnecting the signals on the
	  ptdemux before this point, since we need the push-thread to settle
	  down before we can do this in a non-racy way.

2019-09-12 14:22:10 -0600  Aaron Boxer <aaron.boxer@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid seek DISCONT when only rate changes in same direction
	  Not setting DISCONT avoids a noticable delay when seeking
	  with only rate changing, in the same direction as current
	  rate.

2019-12-10 18:13:11 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Remove deprecated GTimeVal
	  GTimeVal won't work past 2038

2019-12-10 17:13:45 -0500  Olivier Crête <olivier.crete@collabora.com>

	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: Remove deprecated GTimeVal

2019-12-18 12:19:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavimux.c:
	  avimux: Add support for S24LE and S32LE raw audio
	  avidemux already handles this correctly.

2019-12-16 21:07:08 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavimux.c:
	  avimux: Allow muxing v210 video into AVI
	  avidemux already handles this.

2019-12-16 18:43:44 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't replace video codec data when we receive a PAR
	  Receiving a pixel-aspect-ratio should trigger a caps change, but not
	  replace the existing video codec tag

2019-12-12 20:20:35 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: protect access to GstElement.sinkpads

2019-12-03 15:30:06 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* tests/check/elements/qtmux.c:
	  qtmux: port to GstAggregator

2019-12-16 13:03:51 +0100  Joakim Johansson <joakimj@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  gstrtspsrc: Add missing lock on free set_get_param_q
	  Otherwise is it possible to get a crash in gst_rtspsrc_set_parameter.

2019-12-12 18:53:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Increment fragment_id even if no fragment location was provided
	  Applications might handle locations and generally configuration of the
	  sink by themselves instead of having splitmuxsink set the location on
	  the sink. Nonetheless it makes sense to increment the fragment_id that
	  is passed to the signal so that applications know which fragment is
	  requested.

2019-12-12 10:59:35 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvmux.c:
	  flvmux: Use the last DTS for the metadata timestamp
	  This avoids creating a timestamp regression during a stream.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/429

2019-12-11 17:30:50 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: send GAP events for lagging audio and video streams too
	  The logic is taken straight from matroskademux, see
	  77403d0afee635f2de6c2e53a23e1f50ad0d00fa

2019-12-10 23:48:35 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/flv/gstflvmux.c:
	* meson.build:
	  flvmux: Use thread-safe gmtime_r if available
	  gmtime on *nix is not thread-safe.

2019-12-05 14:58:40 +0000  Stéphane Cerveau <scerveau@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: provides a start-index property
	  Allow to change the fragment-id start index.

2019-12-03 11:36:07 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* ext/qt/meson.build:
	  qmlglsink: fix build on EGL platform without X11 headers
	  If Mesa is built without X11 headers, building against Mesa EGL headers
	  requires a dependency on egl.pc, to define MESA_EGL_NO_X11_HEADERS.
	  This fixes a build error when compiling ext/qt/gstqtglutility.cc:
	  In file included from /usr/include/EGL/egl.h:39,
	  from /usr/include/gstreamer-1.0/gst/gl/egl/gstegl.h:44,
	  from ../gst-plugins-good-1.16.1/ext/qt/gstqtglutility.cc:43:
	  /usr/include/EGL/eglplatform.h:124:10: fatal error: X11/Xlib.h: No such file or directory

2019-12-04 01:03:49 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpjpegdepay.c:
	  rtpjpegdepay: outputs framed jpeg
	  Add parsed=true to output caps, as we always output
	  whole frames, timestamped and all. Means also that
	  the output can be decoded by avdec_mjpeg wihout
	  plugging an extra parser (which has no rank).

2019-12-03 13:47:22 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvmux.c:
	  flvmux: Correct metadata handling in file and stream mode
	  In file mode, only push one onMetaData at the start of the stream.
	  In stream mode, always push complete onMetaData. They get replaced, not
	  merged.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/418

2019-12-03 13:46:09 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvmux.c:
	  flvmux: Don't calculate duration in streamable mode
	  There's no header to rewrite, so the duration is left unused.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/418

2016-11-30 15:55:01 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtp/gstrtpL16depay.c:
	  rtpL16depay: don't crash if data is not modulo channels*width

2019-12-02 19:00:45 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* pkgconfig/gstreamer-plugins-good-uninstalled.pc.in:
	* pkgconfig/meson.build:
	  pkgconfig: remove gst-plugins-good-1.0-uninstalled.pc
	  This was never installed and it was only used by the uninstalled
	  autotools dev environment to locate the -good plugins for use
	  in unit tests in gstreamer modules higher up the stack.
	  It is no longer needed now that we no longer have an autotools build.

2017-10-10 15:45:28 +0200  Håvard Graff <havard.graff@gmail.com>

	* pkgconfig/meson.build:
	  meson.build: use join_paths() on prefix
	  So that "/" are correct on Windows.

2017-06-30 09:48:58 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: use baseclass allocator for buffers
	  That way we get some of the meta -> rtp-extension goodies.

2019-11-29 20:46:26 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* ext/vpx/gstvp9dec.c:
	  vp9dec: Fix broken 4:4:4 8bits decoding
	  VPX_IMG_FMT_I444 pixel format with sRGB colorspace means
	  GBR data.
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/651

2019-10-18 17:45:43 +0200  Havard Graff <havard@pexip.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: add test for requesting FIR after having requested PLI

2019-11-26 15:00:18 +0100  Havard Graff <havard@pexip.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: make test more stable

2019-11-29 14:23:49 +0100  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/gstrtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: add locking for clear-pt-map
	  ...or it will segfault from time to time...

2018-05-31 10:29:43 +0200  Linus Svensson <linussn@axis.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	* gst/matroska/matroska-read-common.c:
	  matroskamux: Add property to set DateUTC
	  Add a property that makes it possible for an application to set the
	  DateUTC header field in matroska files. This is useful for live feeds,
	  where the DateUTC header can be set to a UTC timestamp, matching the
	  beginning of the file.
	  Needs gstreamer!323
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/481

2018-05-31 11:20:36 +0200  Linus Svensson <linussn@axis.com>

	* gst/matroska/ebml-ids.h:
	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-mux.c:
	  matroskamux: Use nanosecond precision for DateUTC
	  DateUTC is specified with nanosecond precision in matroska, make use of
	  that.

2018-10-17 02:28:13 +0000  Nicolas Dufresne <nicolas@ndufresne.ca>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Queue number of allocated buffers to capture
	  Before we do streamon, we queue all capture buffers by calling
	  resurrect. When the driver supports CREATE_BUFS, this would lead
	  to buffers being allocated till the maximum of 32 is reached.
	  Instead, we now save the number of allocated buffers and queue this
	  amount.

2019-11-19 14:23:48 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Pass the right size to gst_collect_pads_add_pad
	  We were lucky that GstMatroskamuxPad is larger than GstMatroskaPad.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/393

2019-11-18 13:27:42 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Workaround bad TRY_FMT colorimetry implementation
	  libv4l2 reset the colorpace to 0 and does not do any request to the
	  driver. This yields an invalid colorspace which currently cause a
	  negotiation failure. This workaround by ignoring bad values during the
	  TRY_FMT step.

2019-11-04 17:18:30 +0800  aogun <gun.ao@magewell.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix wrong offset of adts channel

2019-10-07 12:45:00 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't take lock during posting message
	  An application might try to access splitmuxsink from sync message handler
	  by g_object_{get,set} which takes lock also. In general, we don't
	  take lock around message handler.

2019-09-12 15:21:24 -0400  Scott Kanowitz <skanowitz@echo360.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Fix incorrect logic in EOI tag detection
	  This change fixes the reversed logic in the EOI tag detection
	  code.

2019-08-26 08:03:24 +0200  Niels De Graef <nielsdegraef@gmail.com>

	* ext/cairo/gstcairooverlay.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/shout2/gstshout2.c:
	* gst/rtp/rtpstorage.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tunerchannel.c:
	  Don't pass default GLib marshallers for signals
	  By passing `NULL` to `g_signal_new` instead of a marshaller, GLib will
	  actually internally optimize the signal (if the marshaller is available
	  in GLib itself) by also setting the valist marshaller. This makes the
	  signal emission a bit more performant than the regular marshalling,
	  which still needs to box into `GValue` and call libffi in case of a
	  generic marshaller.
	  Note that for custom marshallers, one would use
	  `g_signal_set_va_marshaller()` with the valist marshaller instead.

2019-11-14 17:33:08 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Check the exit condition after executing timers
	  The do_expected_timeout() function may release the JBUF_LOCK, so we need
	  to check if nothing wanted the timer thread to exit after this call.
	  The side effect was that we may endup going back into waiting for a timer
	  which will cause arbitrary delay on tear down (or deadlock when test
	  clock is used).
	  Fixes #653

2019-11-14 17:20:51 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Check exit condition immediately after JBUF_WAIT
	  JBUF_WAIT_QUEUE drops the JBUF_LOCK, which means the stop condition
	  for the chain function may have changed (change_state to NULL). Check
	  this immediately after the wait so that we don't delay shutting down.

2019-11-12 17:28:22 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: Also update the coordinate when in-place
	  This update is needed when the output caps is not changed (e.g. we are
	  moving a viewport around).
	  Fixes #669

2019-11-11 13:19:08 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: Don't always re-run the allocation query
	  When in-place, running an allocation is not useful since videocrop
	  is not implicated in the allocation. So only force the allocation
	  query for the case it was in passthrough. This is needed since the
	  change in the crop region will likely pull us out of this mode. For the
	  case we where neither in passthrough or in-place, the allocation query
	  is already ran by the baseclass, so nothing special is needed.
	  This fixes performance issues when changing the crop region per frame.
	  This was reproduced using videocrop2-test.

2019-11-11 13:18:52 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: Cleanup spurious assignment
	  These are just writing the same thing a second time.

2018-11-07 09:00:02 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: don't overwrite the last valid line
	  If the the height is not a multiple of the macro block size then the memory
	  of the last line is reused for all extra lines. This is no problem if the
	  last line is duplicated properly. However, if the extra lines are not
	  initialized properly during encoding, then the last visible line is
	  overwritten with undefined data.
	  Use a extra buffer to avoid this problem.

2019-11-07 12:28:58 +0100  Stéphane Cerveau <scerveau@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: add fakesink support
	  fakesink does not support "location" property and was generating
	  a warning.

2018-12-12 19:07:39 +0300  Sergey Nazaryev <sergey.nazaryev@cogentembedded.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: don't lose scope_id

2019-11-05 21:41:55 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/vpx/meson.build:
	  vpx: Error out if enabled and no features found
	  Seee: https://gitlab.freedesktop.org/gstreamer/cerbero/issues/200

2019-05-25 21:19:21 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: update match_buffer_layout() debug messages
	  It's no longer used only to try importing buffers.

2019-05-23 10:49:39 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: try matching buffer layout from downstream
	  Ask v4l2 to produce buffers matching the buffer layout requested
	  downstream.

2019-05-21 10:31:46 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: factor out gst_v4l2_object_match_buffer_layout()
	  No semantic change.

2019-10-20 12:17:25 +0200  Havard Graff <havard@pexip.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: make sure not to drop packets based on skew
	  One of the jitterbuffers functions is to try and make sense of weird
	  network behavior.
	  It is quite unhelpful for the jitterbuffer to start dropping packets
	  itself when what you are trying to achieve is better network resilience.
	  In the case of a skew, this could often mean the sender has restarted
	  in some fashion, and then dropping the very first buffer of this "new"
	  stream could often mean missing valuable information, like in the case
	  of video and I-frames.
	  This patch simply reverts back to the old behavior, prior to https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/commit/8d955fc32b552b2db933c67f3cfa31d987f36b81
	  and includes the simplest test I could write to demonstrate the behavior,
	  where a single packet arrives "perfectly", then a 50ms gap happens,
	  and then two more packets arrive in perfect order after that.
	  # Conflicts:
	  #	tests/check/elements/rtpjitterbuffer.c

2019-04-17 12:40:22 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: use alignments from upstream when importing on sink
	  Try configuring the v4l2 output with the alignments from upstream when
	  importing its buffers. This allows us to support importing with
	  non-standard strides and/or heights if supported by the driver.

2019-04-17 12:25:14 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: add support for vertical padding when importing buffers
	  We were already supporting horizontal padding by setting bytesperline to
	  the buffer stride but not vertical one.
	  We are now updating the format height with the padded height and crop to
	  the actual video resolution if needed.

2019-04-17 11:46:10 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: fix debug message if driver rejects stride
	  The 'want' and 'got' strides were inversed.

2019-04-15 11:43:41 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: improve logs when importing buffers
	  Log strides and offsets from upstream.
	  Also fix a typo.

2019-10-29 14:05:48 +0000  James Cowgill <james.cowgill@thinci.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: ensure pool exists before orphaning it
	  In commit e2ff87732d0b ("v4l2videodec: support orphaning") support for
	  orphaning the capture buffer pool was added when the format is
	  renegotiated. However, the commit forgot to check that a pool existed
	  before doing this. This is needed because it's possible for the format
	  to be renegotiated before a capture pool is allocated, which would
	  result in trying to orphan a NULL pool and lead to a NULL pointer
	  dereference.
	  Fix this by checking a pool exists first. If the pool doesn't exist,
	  there are no buffers to be reclaimed, so skip the allocation query in
	  that case.

2019-10-25 22:03:18 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtwindow.cc:
	  qmlglsrc: read from the back buffer when use-default-fbo = TRUE
	  glReadBuffer(GL_COLOR_ATTACHMENT0) on the default framebuffer (0) is
	  invalid GL API usage and would result in a GL error being thrown.

2019-10-25 21:47:01 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtsrc.cc:
	  qmlglsrc: fix vertical flip matrix
	  Some time ago libgstgl defined the majorness of matrices it uses.
	  The majorness used by qmlglsrc was incompatible with the libgstgl.

2019-07-30 12:07:18 +0200  Patricia Muscalu <patricia@axis.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Fix memory leak while pushing fragmented data
	  The memory leak occurs in the case when the buffer has been
	  added to the fragment_buffers array of the current pad and
	  never been sent because of the push failure of the previous
	  buffers: moof or mdat header or fragmented buffer(s).

2019-10-11 14:20:15 +0200  Edward Hervey <edward@centricular.com>

	* gst/debugutils/cpureport.c:
	* gst/debugutils/cpureport.h:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/progressreport.h:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/atoms.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-mux.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/udp/gstmultiudpsink.c:
	* sys/v4l2/gstv4l2src.c:
	  good: Avoid usage of deprecated API
	  GTimeval and related functions are now deprecated in glib.
	  Replacement APIs have been present since 2.26

2019-07-15 07:46:56 +0200  Javier Celaya <javier.celaya@flexvdi.com>

	* sys/osxaudio/meson.build:
	  osxaudio: misspelled dependency
	  When building osxaudio, the required 'AudioToolbox' dependency is
	  misspelled as 'AudioToolBox', which crashes the build with error:
	  ld: framework not found AudioToolBox

2019-06-09 00:43:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* .gitignore:
	* .gitmodules:
	* Makefile.am:
	* README:
	* autogen.sh:
	* common:
	* configure.ac:
	* docs/.gitignore:
	* ext/Makefile.am:
	* ext/aalib/Makefile.am:
	* ext/cairo/Makefile.am:
	* ext/dv/Makefile.am:
	* ext/flac/Makefile.am:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/gtk/Makefile.am:
	* ext/jack/.gitignore:
	* ext/jack/Makefile.am:
	* ext/jpeg/Makefile.am:
	* ext/lame/Makefile.am:
	* ext/libcaca/Makefile.am:
	* ext/libpng/Makefile.am:
	* ext/mpg123/Makefile.am:
	* ext/pulse/Makefile.am:
	* ext/qt/.gitignore:
	* ext/qt/Makefile.am:
	* ext/raw1394/.gitignore:
	* ext/raw1394/Makefile.am:
	* ext/shout2/Makefile.am:
	* ext/soup/Makefile.am:
	* ext/speex/Makefile.am:
	* ext/taglib/.gitignore:
	* ext/taglib/Makefile.am:
	* ext/twolame/Makefile.am:
	* ext/vpx/Makefile.am:
	* ext/wavpack/Makefile.am:
	* gst/Makefile.am:
	* gst/alpha/Makefile.am:
	* gst/apetag/Makefile.am:
	* gst/audiofx/.gitignore:
	* gst/audiofx/Makefile.am:
	* gst/audioparsers/Makefile.am:
	* gst/auparse/.gitignore:
	* gst/auparse/Makefile.am:
	* gst/autodetect/Makefile.am:
	* gst/avi/.gitignore:
	* gst/avi/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debugutils/Makefile.am:
	* gst/deinterlace/Makefile.am:
	* gst/dtmf/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/equalizer/.gitignore:
	* gst/equalizer/Makefile.am:
	* gst/flv/Makefile.am:
	* gst/flx/Makefile.am:
	* gst/goom/.gitignore:
	* gst/goom/Makefile.am:
	* gst/goom2k1/.gitignore:
	* gst/goom2k1/Makefile.am:
	* gst/icydemux/Makefile.am:
	* gst/id3demux/Makefile.am:
	* gst/imagefreeze/Makefile.am:
	* gst/interleave/Makefile.am:
	* gst/isomp4/Makefile.am:
	* gst/law/Makefile.am:
	* gst/level/.gitignore:
	* gst/level/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/monoscope/.gitignore:
	* gst/monoscope/Makefile.am:
	* gst/multifile/Makefile.am:
	* gst/multipart/Makefile.am:
	* gst/replaygain/Makefile.am:
	* gst/rtp/Makefile.am:
	* gst/rtpmanager/Makefile.am:
	* gst/rtsp/.gitignore:
	* gst/rtsp/Makefile.am:
	* gst/shapewipe/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/spectrum/.gitignore:
	* gst/spectrum/Makefile.am:
	* gst/udp/Makefile.am:
	* gst/videobox/Makefile.am:
	* gst/videocrop/Makefile.am:
	* gst/videofilter/.gitignore:
	* gst/videofilter/Makefile.am:
	* gst/videomixer/Makefile.am:
	* gst/wavenc/Makefile.am:
	* gst/wavparse/.gitignore:
	* gst/wavparse/Makefile.am:
	* gst/y4m/Makefile.am:
	* m4/.gitignore:
	* m4/Makefile.am:
	* m4/README:
	* m4/a52.m4:
	* m4/aalib.m4:
	* m4/as-ffmpeg.m4:
	* m4/as-liblame.m4:
	* m4/as-slurp-ffmpeg.m4:
	* m4/check-libheader.m4:
	* m4/freetype2.m4:
	* m4/glib.m4:
	* m4/gst-alsa.m4:
	* m4/gst-artsc.m4:
	* m4/gst-fionread.m4:
	* m4/gst-ivorbis.m4:
	* m4/gst-matroska.m4:
	* m4/gst-sdl.m4:
	* m4/gst-shout2.m4:
	* m4/gst-sid.m4:
	* m4/gtk.m4:
	* m4/libfame.m4:
	* m4/ogg.m4:
	* m4/vorbis.m4:
	* pkgconfig/.gitignore:
	* pkgconfig/Makefile.am:
	* po/.gitignore:
	* po/Makevars:
	* po/POTFILES:
	* sys/Makefile.am:
	* sys/directsound/Makefile.am:
	* sys/oss/.gitignore:
	* sys/oss/Makefile.am:
	* sys/oss4/Makefile.am:
	* sys/osxaudio/Makefile.am:
	* sys/osxvideo/Makefile.am:
	* sys/v4l2/Makefile.am:
	* sys/waveform/Makefile.am:
	* sys/ximage/Makefile.am:
	* tests/Makefile.am:
	* tests/check/.gitignore:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/generic/.gitignore:
	* tests/check/pipelines/.gitignore:
	* tests/examples/Makefile.am:
	* tests/examples/audiofx/.gitignore:
	* tests/examples/audiofx/Makefile.am:
	* tests/examples/cairo/.gitignore:
	* tests/examples/cairo/Makefile.am:
	* tests/examples/equalizer/.gitignore:
	* tests/examples/equalizer/Makefile.am:
	* tests/examples/gtk/.gitignore:
	* tests/examples/gtk/Makefile.am:
	* tests/examples/jack/Makefile.am:
	* tests/examples/level/.gitignore:
	* tests/examples/level/Makefile.am:
	* tests/examples/qt/qmlsink/.gitignore:
	* tests/examples/qt/qmlsrc/.gitignore:
	* tests/examples/rtp/.gitignore:
	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtsp/Makefile.am:
	* tests/examples/shapewipe/.gitignore:
	* tests/examples/shapewipe/Makefile.am:
	* tests/examples/spectrum/.gitignore:
	* tests/examples/spectrum/Makefile.am:
	* tests/examples/v4l2/.gitignore:
	* tests/examples/v4l2/Makefile.am:
	* tests/files/Makefile.am:
	* tests/icles/.gitignore:
	* tests/icles/Makefile.am:
	  Remove autotools build system

2019-10-13 12:46:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: fix wrong type cast
	  Follow-up to commit 1b752c0f !361

2019-09-25 12:36:32 +0000  HuQian <huqian123hq@hotmail.com>

	* sys/v4l2/gstv4l2object.c:
	  is a typo here? gstv4l2object.c

2019-10-11 12:27:12 +0000  Kevin Song <kevinbing.song@gmail.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Check stop in flush() to avoid race condition.
	  Backward playback will drain and flush every frame. Stop playback
	  when backward playback have race condition between exit thread and
	  streaming thread flush. Add one check to avoid it.
	  Fixes #639

2019-10-11 10:33:20 +0800  Fuwei Tang <fuweix.tang@intel.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: fix type conversion errors

2019-09-02 08:27:35 -0400  Aaron Boxer <aaron.boxer@collabora.com>

	* NEWS:
	* docs/gst_plugins_cache.json:
	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflactag.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* ext/gtk/gstgtkbasesink.c:
	* ext/jack/gstjackaudioclient.c:
	* ext/jpeg/Makefile.am:
	* ext/pulse/pulsesink.c:
	* ext/qt/qtwindow.cc:
	* ext/raw1394/gstdv1394src.h:
	* ext/taglib/gstid3v2mux.cc:
	* ext/wavpack/gstwavpackenc.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempoplugin.c:
	* gst/autodetect/gstautodetect.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/tvtime/sse.h:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopBottom.inc:
	* gst/deinterlace/tvtime/tomsmocomp/StrangeBob.inc:
	* gst/deinterlace/tvtime/tomsmocomp/WierdBob.inc:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstquark.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstindex.c:
	* gst/interleave/deinterleave.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_lang.c:
	* gst/level/gstlevel.c:
	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/monoscope/monoscope.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multifile/patternspec.c:
	* gst/replaygain/replaygain.h:
	* gst/rtp/README:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpredenc.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/rtpstorage.c:
	* gst/rtp/rtpulpfeccommon.c:
	* gst/rtp/rtpulpfeccommon.h:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpptdemux.h:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtsp/README:
	* gst/rtsp/gstrtspsrc.c:
	* gst/spectrum/gstspectrum.h:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videomixer/README:
	* gst/videomixer/videomixer2.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* hooks/pre-commit.hook:
	* m4/aalib.m4:
	* m4/freetype2.m4:
	* m4/glib.m4:
	* m4/gst-fionread.m4:
	* m4/gst-matroska.m4:
	* m4/gst-sdl.m4:
	* m4/gst-shout2.m4:
	* m4/gtk.m4:
	* m4/libfame.m4:
	* m4/ogg.m4:
	* m4/vorbis.m4:
	* sys/oss4/oss4-audio.c:
	* sys/oss4/oss4-soundcard.h:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	* sys/ximage/ximageutil.h:
	* tests/check/elements/jpegdec.c:
	* tests/check/elements/level.c:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rgvolume.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtpbin.c:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/rtpred.c:
	* tests/check/elements/rtprtx.c:
	* tests/check/elements/rtpsession.c:
	* tests/check/elements/rtpstorage.c:
	* tests/check/elements/splitmux.c:
	* tests/check/pipelines/simple-launch-lines.c:
	* tests/examples/cairo/cairo_overlay.c:
	* tests/examples/gtk/glliveshader.c:
	* tests/examples/rtp/client-rtpaux.c:
	* tests/examples/v4l2/camctrl.c:
	  documentation: fix a number of typos

2019-10-04 20:31:56 +0000  Simon Arnling Bååth <simon.baath@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  gstrtpjitterbuffer: Custom messages when dropping packets
	  This commit adds custom element messages for when gstrtpjitterbuffer
	  drops an incoming rtp packets due to for example arriving too late.
	  Applications can listen to these messages on the bus which enables
	  actions to be taken when packets are dropped due to for example high
	  network jitter.
	  Two properties has been added, one to enable posting drop messages and
	  one to set a minimum time between each message to enable throttling the
	  posting of messages as high drop rates.

2019-09-03 16:46:30 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Specify REDIRECT information in error message
	  There are in the wild (mp4) streams that basically contain no tracks
	  but do have a redirect info[0], in which case, we won't be able
	  to expose any pad (there are no tracks) so we can't post anything but
	  an error on the bus, as:
	  - it can't send EOS downstream, it has no pad,
	  - posting an EOS message will be useless as PAUSED state can't be
	  reached and there is no sink in the pipeline meaning GstBin will
	  simply ignore it
	  The approach here is to to add details to the ERROR message with a
	  `redirect-location` field which elements like playbin handle and use right
	  away.
	  [0]: http://movietrailers.apple.com/movies/paramount/terminator-dark-fate/terminator-dark-fate-trailer-2_480p.mov

2019-09-26 18:39:48 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Cancel timers instead of just unlocking loop thread
	  When the queue is full (and adding more packets would risk a seqnum
	  roll-over), the best approach is to just start pushing out packets
	  from the other side.  Just pushing out the packets results in the
	  timers being left hanging with old seqnums, so it's safer to just
	  execute them immediately in this case. It does limit the timer space
	  to the time it takes to receiver about 32k packets, but without
	  extended sequence number, this is the best RTP can do.
	  This also results in the test no longer needed to have timeouts or
	  timers as pushing packets in drives everything.
	  Fixes #619

2019-09-27 14:04:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Optimize offset update
	  As we are applying the same offset over all timers, there timer
	  ordering won't change, so we can safely skip time-reordering.

2019-09-27 16:21:22 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtptimerqueue.c:
	  rtptimerqueue: Optimize reschedule optations
	  This basically add ability to choose between inserting from head, tail
	  or in-place in order to try and minimize the distance to walk through in
	  the timer queue. This removes an overhead we had seen on high drop rate.

2019-09-27 14:04:03 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix a typo in comment

2019-07-02 15:52:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Don't use stats timer on the timers queue
	  The timer passed to update_timers may be from the stats timer. At the
	  moment, we could endup rescheduling (reusing) that timer onto the normal
	  timer queue, unschedul it as if it was from the normal timer queue or
	  duplicate it into the stats timer queue again. This was protected before
	  as the with the fact the stats timer didn't have a valid idx.

2019-06-21 14:08:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Update timers on ts-offset changes
	  As the offset is already applied now, we need to update and reschedule
	  all timers each time the offset is changed. I'm not sure who expect this
	  to be retro-actively applied, but there was a unit test for it.

2019-06-20 15:59:48 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: No need to wake the timer thread on head changes
	  If the jitterbuffer head change, there is no need to systematically
	  wakeup the timer thread. The timer thread will be waken up on if
	  an earlier timeout has been pushed. This prevent some more spurious
	  wakeup when the system is loaded. As a side effect, cranking the clock
	  may set the clock at an earlier position.

2019-06-18 19:07:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtptimerqueue.h:
	  rtpjittterbuffer: Port timers array to RtpTimerQueue
	  In this patch we now make use of the new RtpTimerQueue instead of the
	  old GArray. This required a lot of changes all over the place, some of
	  the important changes are that `timer->timeout` is no longer a PTS but
	  the actual timeout. This was required to get the RtpTimerQueue sorting
	  right. The applied offset is saved as `timer->offset`, this allow
	  retreiving back the PTS when needed.
	  The clockid updates only happens once per incoming packet. If the
	  currently schedule timer is before the earliest timer in the queue, we
	  no longer wakeup the thread. This way, if other timers get setup in the
	  meantime, this will reduce the number of wakup.
	  The timer loop code has been mostly rewritten, though the behaviour of
	  running the lost timers first has been kept (even though there is no
	  test to show what would be the side effect of doing this differently).
	  Fixes #608

2019-06-14 14:29:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjittterbuffer: Port from TimerQueue to RtpTimerQueue

2019-06-13 17:08:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtptimerqueue.h:
	  rtpjitterbuffer: Port use the new RtpTimer structure
	  First iteration toward porting to the new timer queue.

2019-06-12 09:59:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/meson.build:
	* gst/rtpmanager/rtptimerqueue.c:
	* gst/rtpmanager/rtptimerqueue.h:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/meson.build:
	  rtptimerqueue: Consolidate a data structure for timers
	  Implement a single timer queue for all timers. The goal is to always use
	  ordered queues for storing timers. This way, extracting timers for
	  execution becomes O(1). This also allow separating the clock wait
	  scheduling from the timer itself and ensure that we only wake up the
	  timer thread when strictly needed.
	  The knew data structure is still O(n) on insertions and reschedule,
	  but we now use proximity optimization so that normal cases should be
	  really fast. The GList structure is also embeded intot he RtpTimer
	  structure to reduce the number of allocations.

2019-06-10 16:46:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: jitterbuffer: Demacroify some helpers
	  There is no reason for these to be macros anymore. This makes the
	  test helper much more readable.

2019-06-06 14:44:27 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: Move item structure outside of the element
	  This moves the RtpJitterBufferStructure type, alloc, free into
	  rtpjitterbuffer.c/h implementation. jitterbuffer.c strictly rely on
	  the fact this structure is compatible with GList, and so it make more
	  sense to keep encapsulate it. Also, anything that could possibly
	  reduce the amount of code in the element is a win.
	  In order to support that move, a function pointer to free the data
	  was added. This also allow making the free function option when
	  flushing the jitterbuffer.

2019-06-06 13:09:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Constify timer pointers where possible
	  This helps understanding which function modify the Timerdata
	  and which one does not. This is not always obvious from thelper
	  name considering recalculate_timer() does not.

2019-09-27 08:46:22 +0200  Philipp Zabel <philipp.zabel@gmail.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2mpeg2codec.c:
	* sys/v4l2/gstv4l2mpeg2codec.h:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/meson.build:
	  v4l2: Add MPEG-2 profile and level support
	  Add support for V4L2 MPEG-2 decoders reporting supported profiles and
	  levels.

2019-09-23 14:34:20 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: add support for ABGR, xBGR, RGBA, and RGBx formats
	  Map them to the new V4L2_PIX_FMT_{BGRA32,BGRX32,RGBA32,RGBX32} pixel
	  formats.

2019-09-23 14:10:15 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	  v4l2: update kernel headers to latest from media tree
	  Update to the latest installed headers (output of make headers_install)
	  from the media tree, keeping the slight modifications to the includes.
	  This includes typo fixes in enum v4l2_mpeg_video_multi_slice_mode,
	  MPEG-2 level and profile enums, new FWHT and H.264 Qp controls, new
	  RGB(A) formats, and new continuous bytestream and dynamic resolution
	  format flags.

2017-12-19 18:23:16 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: add request-jitterbuffer signal
	  This can be used to pass the threadsharing jitterbuffer from
	  gst-plugins-rs for example.

2019-09-23 18:46:16 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/qtdemux.c:
	  build: fix werror build with newer gcc
	  In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:55,
	  from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/tag/tag.h:25,
	  from ../gst/isomp4/qtdemux.c:56:
	  In function ‘qtdemux_inspect_transformation_matrix’,
	  inlined from ‘qtdemux_parse_trak’ at ../gst/isomp4/qtdemux.c:10676:5,
	  inlined from ‘qtdemux_parse_tree’ at ../gst/isomp4/qtdemux.c:14210:5:
	  ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstinfo.h:645:5: error: ‘%s’ directive argument is null [-Werror=format-overflow=]
	  645 |     gst_debug_log ((cat), (level), __FILE__, GST_FUNCTION, __LINE__, \
	  |     ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	  646 |         (GObject *) (object), __VA_ARGS__);    \
	  |         ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	  ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstinfo.h:1062:35: note: in expansion of macro ‘GST_CAT_LEVEL_LOG’
	  1062 | #define GST_DEBUG_OBJECT(obj,...) GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_DEBUG,   obj,  __VA_ARGS__)
	  |                                   ^~~~~~~~~~~~~~~~~
	  ../gst/isomp4/qtdemux.c:10294:5: note: in expansion of macro ‘GST_DEBUG_OBJECT’
	  10294 |     GST_DEBUG_OBJECT (qtdemux, "Transformation matrix rotation %s",
	  |     ^~~~~~~~~~~~~~~~
	  ../gst/isomp4/qtdemux.c: In function ‘qtdemux_parse_tree’:
	  ../gst/isomp4/qtdemux.c:10294:64: note: format string is defined here
	  10294 |     GST_DEBUG_OBJECT (qtdemux, "Transformation matrix rotation %s",
	  |                                                                ^~

2019-09-18 18:31:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: Use the new helper functions for mapping the colr atom values to colorimetry

2019-09-18 18:29:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use the new helper functions for mapping the colr atom values to colorimetry

2019-09-10 22:44:20 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	  docs: update plugin cache

2019-09-10 22:43:49 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/smpte/barboxwipes.c:
	  smpte: don't register transition types twice

2019-09-08 20:43:17 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/alpha/gstalpha.c:
	  alpha: Fix one_over_kc calculation
	  On arm/aarch64, converting from float directly to unsigned int uses
	  a different opcode and negative numbers result in 0. Cast to
	  signed int first.

2019-07-31 16:17:36 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* tests/check/elements/splitmux.c:
	  splitmux: Add muxer-pad-map property
	  Add a property which explicitly maps splitmuxsink pads to the
	  muxer pads they should connect to, overriding the implicit logic
	  that tries to match pads but yields arbitrary names.

2019-07-26 02:21:59 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: In async mode, retain previous muxer pad names.
	  When running in async-finalize mode, request new pads from the muxer
	  using the same names as old pads, instead of letting the muxer assign
	  new ones based on the pad template name.

2019-07-26 02:13:31 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Mark split-* signals as action signals. Doc fixes.
	  Add the G_SIGNAL_ACTION flag to the split-* signals on splitmuxsink,
	  and make some improvements to their docstrings

2019-08-29 22:11:02 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix incompatible type warning with MSVC
	  gstqtmux.c(5582): warning C4133: 'function':
	  incompatible types - from 'GstVideoMultiviewFlags *' to 'guint *'

2019-09-02 16:33:05 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix git diff indentation

2019-08-30 22:42:58 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: normalize variable to boolean

2019-08-29 21:29:34 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: clip output segment on accurate seeks
	  The output segment is only used in ONVIF mode.
	  The previous behaviour was to output a segment computed from
	  the Range response sent by the server.
	  In ONVIF mode, servers will start serving from the appropriate
	  synchronization point (keyframe), and the Range in response will
	  start at that position.
	  This means rtspsrc can now perform truly accurate seeks in that
	  mode, by clipping the output segment to the values requested in
	  the seek. The decoder will then discard out of segment buffers
	  and playback will start without artefacts at the exact requested
	  position, similar to the behaviour of a demuxer when an accurate
	  seek is requested.

2019-08-30 14:00:26 +1000  Matthew Waters <matthew@centricular.com>

	* ext/vpx/gstvpxenc.c:
	  vpx: fix macos werror build
	  ../ext/vpx/gstvpxenc.c:1723:49: error: format specifies type 'long' but the argument has type 'vpx_codec_pts_t' (aka 'long long') [-Werror,-Wformat]
	  ", gst frame pts: %" G_GINT64_FORMAT, pkt->data.frame.pts, pts);
	  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
	  /Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:1065:96: note: expanded from macro 'GST_TRACE_OBJECT'
	  #define GST_TRACE_OBJECT(obj,...)       GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_TRACE,   obj,  __VA_ARGS__)
	  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~
	  /Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:646:31: note: expanded from macro 'GST_CAT_LEVEL_LOG'
	  (GObject *) (object), __VA_ARGS__);                             \
	  ^~~~~~~~~~~
	  ../ext/vpx/gstvpxenc.c:1723:70: error: format specifies type 'long' but the argument has type 'vpx_codec_pts_t' (aka 'long long') [-Werror,-Wformat]
	  ", gst frame pts: %" G_GINT64_FORMAT, pkt->data.frame.pts, pts);
	  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~
	  /Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:1065:96: note: expanded from macro 'GST_TRACE_OBJECT'
	  #define GST_TRACE_OBJECT(obj,...)       GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_TRACE,   obj,  __VA_ARGS__)
	  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~
	  /Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:646:31: note: expanded from macro 'GST_CAT_LEVEL_LOG'
	  (GObject *) (object), __VA_ARGS__);                             \
	  ^~~~~~~~~~~

2019-08-30 13:37:59 +1000  Matthew Waters <matthew@centricular.com>

	* sys/osxvideo/cocoawindow.m:
	  osxvideosink: call superclass in reshape
	  Fixes macos werror build
	  ../sys/osxvideo/cocoawindow.m:437:1: error: method possibly missing a [super reshape] call [-Werror,-Wobjc-missing-super-calls]
	  }
	  ^

2019-08-23 18:56:01 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/lame/gstlamemp3enc.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux.c:
	* gst/shapewipe/gstshapewipe.c:
	  docstrings: port ulinks to markdown links

2019-08-10 12:33:46 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	  replaygain: fix up doc links to defunct replaygain.org website
	  Fixes #624

2019-08-22 00:18:51 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Fix incompatible type build warning
	  gstsouphttpsrc.c(2191): warning C4133:
	  '=': incompatible types - from 'guint (__cdecl *)(GType)' to 'GstURIType (__cdecl *)(GType)'

2019-08-19 11:07:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/gstvpxdec.c:
	* ext/vpx/meson.build:
	  vpx: bump libvpx requirement to 1.5.0
	  Was released in Nov 2015.

2019-08-19 11:03:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/meson.build:
	  vpx: avoid confusing meson configure output when checking for vpx versions
	  Used to print:
	  |Run-time dependency vpx found: YES 1.7.0
	  |Message: libvpx provides VP8 encoder interface (vpx_codec_vp8_cx_algo)
	  |Message: libvpx provides VP8 decoder interface (vpx_codec_vp8_dx_algo)
	  |Message: libvpx provides VP9 encoder interface (vpx_codec_vp9_cx_algo)
	  |Message: libvpx provides VP9 decoder interface (vpx_codec_vp9_dx_algo)
	  |Dependency vpx found: YES (cached)
	  |Dependency vpx found: NO found '1.7.0' but need: '>=1.8.0'
	  |Run-time dependency vpx found: NO (tried pkgconfig and cmake)
	  We can check the version of the found dep in a way that
	  doesn't produce this confusing output.

2019-08-19 07:30:17 +0000  Amr Mahdi <amramahdi@gmail.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix push mode ignoring audio with a size smaller than segment buffer
	  In push mode (streaming), if the audio size is smaller than segment buffer size, it would be ignored.
	  This happens because when the plugin receives an EOS signal while a single audio chunk that is less than the segment buffer size is buffered, it does not
	  flush this chunk. The fix is to flush the data chunk when it receives an EOS signal and has a single (first) chunk buffered.
	  How to reproduce:
	  1. Run gst-launch with tcp source
	  ```
	  gst-launch-1.0  tcpserversrc port=3000 !  wavparse ignore-length=0 ! audioconvert ! filesink location=bug.wav
	  ```
	  2. Send a wav file with unspecified data chunk length (0). Attached a test file
	  ```
	  cat test.wav | nc localhost 3000
	  ```
	  3. Compare the length of the source file and output file
	  ```
	  ls -l test.wav bug.wav
	  -rw-rw-r-- 1 amr amr    0 Aug 15 11:07 bug.wav
	  -rwxrwxr-x 1 amr amr 3564 Aug 15 11:06 test.wav
	  ```
	  The expected length of the result of the gst-lauch pipeline should be the same as the test file minus the headers (44), which is ```3564 - 44 = 3520``` but the actual output length is ```0```
	  After the fix:
	  ```
	  ls -l test.wav fix.wav
	  -rw-rw-r-- 1 amr amr 3520 Aug 15 11:09 fix.wav
	  -rwxrwxr-x 1 amr amr 3564 Aug 15 11:06 test.wav
	  ```

2019-08-12 18:56:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	  rtpvp8depay: Add property for waiting until the next keyframe after packet loss
	  If VP8 is not encoded with error resilience enabled then any packet loss
	  causes very bad artefacts when decoding and waiting for the next
	  keyframe instead improves user experience considerably.

2019-08-06 22:27:40 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/types-compat.h:
	  v4l2: Fix type compatibility issue with glibc 2.30
	  From now on, we will use linux/types.h on Linux, and use typedef of the
	  various flavour of BSD.
	  Fixes #635

2019-08-07 18:29:25 -0400  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/gst-plugins-good.supp:
	  valgrind: suppress Cond error coming from gnutls
	  taken from https://salsa.debian.org/debian/flatpak/commit/fb4a8dda211c4bc036781f2b0d706266e95ce068

2019-07-10 22:07:05 +0300  Mart Raudsepp <mart.raudsepp@collabora.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroska: Provide audio lead-in for some lossy formats
	  Various audio formats require an audio lead-in to decode it properly.
	  Most parsers would take care of it, but when a container like matroska is
	  involved, the demuxer handles the seeking and without its own lead-in
	  handling would never even pass the lead-in data to the parser.
	  This commit provides an initial implementation of that for audio/mpeg,
	  audio/x-ac3 and audio/x-eac3 by calculating the worst case lead-in time
	  needed from known samplerate, potential lead-in frames need and the
	  maximum blocksize possible for the format (as we don't parse that out
	  exactly in matroskademux) and seeking that much earlier in case of
	  accurate seeks. This is especially important for NLE use-cases with GES.
	  If accurate seeking to a position that happens to have a video keyframe,
	  it'll go back to the previous keyframe than needed, but with typical
	  video files that's the best we can do anyway without falling back to
	  scanning the clusters, as typically only keyframes are indexed in
	  Cueing Data.
	  If the media doesn't have a CUE, then we bisect for the cluster to seek
	  to with the same modified time as well in case of accurate seeking,
	  ensuring sufficient lead-in. This code path is typically hit only with
	  (suboptimal) audio-only matroska files, e.g. when created with ffmpeg,
	  which doesn't add a CUE for audio-only mkv muxing.

2019-03-11 15:15:12 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for invalid packets in buffer list
	  Upstream elements can send all kinds of data in a buffer list, so cover
	  the case of an invalid RTP packet mixed with valid RTP packets.

2019-03-11 15:12:03 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for multiplexed RTP and RTCP
	  RTP and RTCP packets can be muxed together on the same channel (see
	  RFC5761) and can arrive in the same buffer list.
	  The GStreamer rtpsession element support RFC5761, so add a test to cover
	  this case for buffer lists too.

2019-03-11 15:09:27 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for different timestamps in buffer list
	  Buffers with different timestamps (e.g. packets belonging to different
	  frames) can arrive together in the same buffer list,
	  Add a test to cover this case.

2019-03-12 15:24:26 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add function to check timestamp

2019-04-02 18:02:19 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test about reordered or duplicated seqnums

2019-04-02 17:52:54 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for lange jump in seqnums with recovery

2019-04-02 17:50:35 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for large jump in sequence numbers

2019-04-02 17:47:27 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for wrapping sequence numbers

2019-03-11 15:07:08 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for permissible gap in sequence numbers

2019-03-11 15:03:31 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for the case of failed probation
	  When a new source fails to pass the probation period (i.e. new packets
	  have non-consecutive sequence numbers), then no buffer shall be pushed
	  downstream. Add a test to validate this case.

2019-03-12 15:23:16 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add function to check sequence number

2019-04-03 14:46:35 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add test to verify that receiving stats are correct
	  Add a test to verify that stats about received packets are correct when
	  using buffer lists in the rtpsession receive path.
	  Split get_session_source_stats() in two to be able to get stats from
	  a GstRtpSession object directly.

2019-02-27 16:17:57 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add a test for buffer lists on the recv path

2019-02-27 17:03:44 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: add support for buffer lists on the recv path
	  The send path in rtpsession processes the buffer list along the way,
	  sharing info and stats between packets in the same list, because it
	  assumes that all packets in a buffer list are from the same frame.
	  However, in the receiving path packets can arrive in all sorts of
	  arrangements:
	  - different sources,
	  - different frames (different timestamps),
	  - different types (multiplexed RTP and RTCP, invalid RTP packets).
	  so a more general approach should be used to correctly support buffer
	  lists in the receive path.
	  It turns out that it's simpler and more robust to process buffers
	  individually inside the rtpsession element even if they come in a buffer
	  list, and then reassemble a new buffer list when pushing the buffers
	  downstream.
	  This avoids complicating the existing code to make all functions
	  buffer-list-aware with the risk of introducing regressions,
	  To support buffer lists in the receive path and reduce the "push
	  overhead" in the pipeline, a new private field named processed_list is
	  added to GstRtpSessionPrivate, it is set in the chain_list handler and
	  used in the process_rtp callback; this is to achieve the following:
	  - iterate over the incoming buffer list;
	  - process the packets one by one;
	  - add the valid ones to a new buffer list;
	  - push the new buffer list downstream.
	  The processed_list field is reset before pushing a buffer list to be on
	  the safe side in case a single buffer was to be pushed by upstream
	  at some later point.
	  NOTE:
	  The proposed modifications do not change the behavior of the send path.
	  The process_rtp callback is called in rtpsource.c by the push_rtp
	  callback (via source_push_rtp) only when the source is not internal.
	  So even though push_rtp is also called in the send path, it won't end up
	  using process_rtp in this case because the source would be internal in
	  the send path.
	  The reasoning from above may suggest a future refactoring: push_rtp
	  might be split to better differentiate the send and receive path.

2019-08-07 10:01:34 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-parse.c:
	  matroska: Handle interlaced field order

2019-08-07 12:09:46 +0000  Amr Mahdi <amramahdi@gmail.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix ignoring of last chunk in push mode
	  In push mode (streaming), if the last audio payload chunk is less than the segment rate buffer size, it would be ignored since the plugin waits until it has at least segment rate bufer size of audio.
	  The fix is to introduce a flushing flag that indicates that no more audio will be available so that the plugin can recognize this condition and flush the data is has even if it is less
	  than the desired segment rate buffer size.

2019-08-06 16:27:37 +0200  Robert Tiemann <rtie@gmx.de>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Log any error returned by soup_session_send()

2019-08-07 11:42:21 +0900  luke.lin <luke.lin@realtek.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: enlarge the maximal atom size
	  For 8K content, frame size is over 25MB, and cause the negotiation failure.
	  Enlarge the limitation of QTDEMUX_MAX_ATOM_SIZE to 32MB.

2019-07-27 04:05:01 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: expose and implement is-live property
	  This is useful to support the ONVIF case: when is-live is set to
	  FALSE and onvif-rate-control is no, the client can control the
	  rate of delivery and arrange for the server to block and still
	  keep sending when unblocked, without requiring back and forth
	  PAUSE / PLAY requests. This enables, amongst other things, fast
	  frame stepping on the client side.
	  When is-live is FALSE, we don't use a manager at all. This case
	  was actually already pretty well handled by the current code. The
	  standard manager, rtpbin, is simply no longer needed in this case.
	  Applications can instantiate a downloadbuffer after rtspsrc if
	  needed.

2019-07-27 04:03:44 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: reset_time when flush stopping

2019-07-12 22:33:08 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: expose and implement onvif-mode property
	  Refactor the code for parsing and generating the Range, taking
	  advantage of existing API in GstRtspTimeRange.
	  Only use the TCP protocol in that mode, as per the specification.
	  Generate an accurate segment when in that mode, and signal to the
	  depayloader that it should not generate its own segment, through
	  the "onvif-mode" field in the caps, see
	  <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/merge_requests/328>
	  for more information.
	  Translate trickmode seek flags to their ONVIF representation
	  Expose an onvif-rate-control property

2019-07-01 20:38:20 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: improve handling of rate in seeks

2019-07-31 21:55:16 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	  rtpfunnel: forward correct segment when switching pad
	  Forwarding a single segment event from the pad that first gets
	  chained is incorrect: when that first event was sent by an element
	  such as x264enc, with its offset start, we end pushing out of segment
	  buffers for the other pad(s).
	  Instead, everytime the active pad changes, forward the appropriate
	  segment event.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1028

2019-08-05 19:35:36 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use new GstRTSPMessage API to set message body from a buffer directly

2019-04-04 13:17:34 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: fix receiver source stats to consider previously queued packets
	  When it is not clear yet if a packet relative to a source should be
	  pushed, the packet is put into a queue, this happens in two cases:
	  - the source is still in probation;
	  - there is a large jump in seqnum, and it is not clear what
	  the cause is, future packets will help making a guess.
	  In either case stats about received packets are not updated at all; and
	  even if they were, when init_seq() is called it resets all receiver
	  stats, effectively loosing any possible stat about previously received
	  packets.
	  Fix this by taking into account the queued packets and update the stats
	  when calling init_seq().

2019-04-09 10:46:39 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: clarify meaning of the octets-sent and octets-received stats
	  The octets-send and octets-received stats count the payload bytes
	  excluding RTP and lower level headers, clarify that in the
	  documentation.

2019-04-04 13:16:36 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: expose field bytes_received in RTPSourceStats
	  Since commit c971d1a9a (rtpsource: refactor bitrate estimation,
	  2010-03-02) bytes_received filed in RTPSourceStats is set but then never
	  used again, expose it so that it can be used  by user code to verify how
	  many bytes have been received.

2019-06-21 17:46:36 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpmanager: consider UDP and IP headers in bandwidth calculation
	  According to RFC3550 lower-level headers should be considered for
	  bandwidth calculation.
	  See https://tools.ietf.org/html/rfc3550#section-6.2 paragraph 4:
	  Bandwidth calculations for control and data traffic include
	  lower-layer transport and network protocols (e.g., UDP and IP) since
	  that is what the resource reservation system would need to know.
	  Fix the source data to accommodate that.
	  Assume UDPv4 over IP for now, this is a simplification but it's good
	  enough for now.
	  While at it define a constant and use that instead of a magic number.
	  NOTE: this change basically reverts the logic of commit 529f443a6
	  (rtpsource: use payload size to estimate bitrate, 2010-03-02)

2019-08-01 15:02:23 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use empty-array safe way to cleanup GPtrArray
	  Fix assertion fail
	  GLib-CRITICAL **: g_ptr_array_remove_range: assertion 'index_ < rarray->len' failed

2019-08-01 14:28:04 +0000  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	  rtpmp4vpay: config-interval -1 send at idr
	  adjust/port from rtph264pay and allow sending the configuration data at
	  every IDR
	  The payloader was stripping the configuration data when the
	  config-interval was set to 0. The code was written in such a way !(a >
	  0) that it stripped the config when it was set at -1 (send config_data
	  as soon as possible).
	  This resulted in some MPEG4 streams where no GOP/VOP-I was detected to
	  be sent out without configuration.

2019-07-27 14:21:34 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Ignore crc32 element while peeking at cluster.

2019-07-25 21:21:26 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	  gtkglsink: fix crash when widget is resized after element destruction
	  Prevent _size_changed_cb() to be called after gtkglsink has been finalized.
	  Fix #632

2019-07-26 02:45:51 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix reverse playback EOS conditions
	  In reverse playback, we don't want to rely on the position of the current
	  keyframe to decide a stream is EOS: the last GOP we push will start with
	  a keyframe, which position is likely to be outside of the segment.
	  Instead, let the normal seek_to_previous_keyframe mechanism do its job,
	  it works just fine.

2019-07-23 01:42:02 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix key unit seek corner case
	  If a key unit seek is performed with a time position that matches
	  the offset of a keyframe, but not its actual PTS, we need to
	  adjust the segment nevertheless.
	  For example consider the following case:
	  * stream starts with a keyframe at 0 nanosecond, lasting 40 milliseconds
	  * user does a key unit seek at 20 milliseconds
	  * we don't adjust the segment as the time position is "over" a keyframe
	  * we push a segment that starts at 20 milliseconds
	  * we push a buffer with PTS == 0
	  * an element downstream (eg rtponviftimestamp) tries to calculate the
	  stream time of the buffer, fails to do so and drops it

2019-07-25 15:08:54 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Don't dereference NULL input state if we have no caps in TIME segments
	  Simply assume that the JPEG frame is not going to be interlaced instead
	  of crashing.

2019-07-22 10:28:50 +0200  Knut Andre Tidemann <knutandre.tidemann@zenitel.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtp: opuspay: fix memory leak in gst_rtp_opus_pay_setcaps.
	  The src caps were never dereferenced, causing a memory leak.

2019-07-12 20:51:44 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: implement support for trickmode interval
	  When the seek event contains a (newly-added) trickmode interval,
	  and TRICKMODE_KEY_UNITS was requested, only let through keyframes
	  separated with the required interval

2019-07-17 19:12:19 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* docs/meson.build:
	  meson: Don't generate doc cache when no plugins are enabled
	  Fixes gst-build with -Dauto-features=disabled

2019-07-15 23:24:05 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: Port to color_{primaries,transfer,matrix}_to_iso
	  ... and remove duplicated code.

2019-05-25 22:08:05 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* tests/check/elements/splitmux.c:
	  splitmuxsink: add the ability to mux auxilliary video streams
	  The primary video stream is used to select fragment cut points
	  at keyframe boundaries. Auxilliary video streams may be
	  broken up at any packet - so fragments may not start with a keyframe
	  for those streams.

2019-06-11 23:17:30 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Add video_%d pad template.
	  splitmuxsrc actually supports multiple video pads. Make that clear,
	  especially since it was already creating pads named "video_0" etc.

2019-07-09 23:12:45 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix conditions for end of segment in reverse playback
	  The time_position field of the stream is offset by the media_start
	  of its QtDemuxSegment compared to the start of the GstSegment of
	  the demuxer, take it into account when making comparisons.

2019-07-09 23:06:12 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix mismatched transfer characteristic
	  TransferCharacteristics(18) should be ARIB STD-B67 (HLG)
	  See https://www.webmproject.org/docs/container/#TransferCharacteristics
	  Also map more color primaries indexes which have been handled by matroska-mux.

2019-07-09 19:49:57 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: Remove misleading comments
	  gst_pad_template_new() does not take ownership of the caps

2019-01-23 18:27:06 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpsession.c:
	  rtp session: Add test for collision loopback detection
	  Ignore further collisions if the remote SSRC change with ours, it's
	  probably because someone is sending us back the packets we send out.

2019-01-23 18:14:23 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession tests: Add test for third-party collision detection
	  Add tests to validate the code that ignores the same packets coming
	  from 2 different sources (an third-party collision).

2019-01-23 17:19:15 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: Add test for collision on incoming packets
	  Make sure that the collision is properly detected on incoming packets.

2019-01-23 17:09:27 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession test: Verify that on-ssrc-collision message is emitted

2019-01-23 16:58:22 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Also send conflict event when sending packet
	  If the conflict is detected when sending a packet, then also send an
	  upstream event to tell the source to reconfigure itself.
	  Also ignore the collision if we see more than one collision from the same
	  remote source to avoid problems on loops.

2019-04-15 16:32:03 -0700  Song Bing <bing.song@nxp.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: set right buffer count.
	  Set right buffer count to avoid one buffer.

2019-06-27 19:47:41 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	* tests/check/elements/rtph265.c:
	  rtph265pay: Also immediately send packet if it is a suffix NAL
	  Immediately send packet if it contains any suffix NAL, this is required
	  in case they come after the VCL nal to not have to wait until the next frame.

2019-06-27 19:46:01 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Don't drop second byte of NAL header
	  At least keep 2 bytes per NAL even if the second one is 0, the
	  second byte of the NAL header could very well be 0.

2019-06-26 16:42:44 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	  rtph26xpay: Avoid print when there is no bundle at end of packet

2019-06-26 16:25:01 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtph264.c:
	* tests/check/elements/rtph265.c:
	  rtph26xpay: Wait until there is a VCL or suffix NAL to send
	  With unit tests.

2019-06-19 17:16:03 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtph265.c:
	  rtph265pay test: Add unit tests for aggregation

2019-06-18 19:07:38 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtph265.c:
	  rtph265pay: Implement Aggregation packets
	  Align with rtph264pay

2019-06-18 15:03:09 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtph264.c:
	  rtph264pay test: Add unit tests for aggregation

2019-06-18 13:45:15 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Report latency when in maximal aggregation mode

2019-06-17 11:31:53 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	* tests/check/elements/rtph264.c:
	  rtph264pay: Default to not adding latency when aggregating
	  Send the bundle as soon as there is one VCL unit in the packet at
	  the end of an incoming buffer.
	  The DELTA_UNIT flag is not reliable, so ignore it.

2019-06-14 16:54:23 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtp-payloading.c:
	  rtp-payloading test: Fix working to 1.0 buffers instead of groups

2019-06-13 18:07:35 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Replace fragmentation while-loop with for-loop
	  Align with rtph264pay

2019-06-13 17:42:05 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Rename payload_len to max_fragment_size
	  Align to rtph264pay

2019-06-13 17:30:08 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Clean up _payload_nal
	  Move determining whether we need to fragment at all into the
	  fragmenter.
	  Align with rtph264pay

2019-06-13 17:23:26 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Extract sending fragments into _payload_nal_fragment
	  Align with rtph264pay

2019-06-13 16:22:57 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Extract sending a single packet into _payload_nal_single
	  Align with rtph264pay

2019-06-13 16:14:31 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Define and use FU_A_TYPE_ID
	  Align with rtph264pay

2019-06-13 16:08:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Use snake_case variables
	  Align with rtph264pay

2019-06-13 16:04:39 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Clean up whitespace and syntax
	  Align with rtph264pay

2018-07-03 19:39:25 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtph264.c:
	  rtph264pay: Support STAP-A bundling
	  Add a new property "do-aggregate"* to the H.264 RTP payloader which
	  enables STAP-A aggregation as per [RFC-6184][1]. With aggregation enabled,
	  packets are bundled instead of sent immediately, up until the MTU size.
	  Bundles also end at access unit boundaries or when packets have to be
	  fragmented.
	  *: The property-name is kept generic since it might apply more widely,
	  e.g. STAP-B or MTAP.
	  [1]: https://tools.ietf.org/html/rfc6184#section-5.7
	  Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/434

2018-11-05 17:15:39 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Fix delta-unit/discont handling when injecting SPS/PPS
	  Apply the wanted delta-unit and discont to the first packet; following
	  packets for this frame are always delta units and not discont.

2018-11-05 19:03:45 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Replace fragmentation while-loop with for-loop

2018-11-05 18:57:38 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Calculate the right max_fragments

2018-11-05 18:36:35 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Rename payload_len to max_fragment_size

2018-11-05 18:34:40 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Clean up _payload_nal_fragment

2018-11-05 18:06:19 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Clean up _payload_nal
	  Move determining whether we need to fragment at all into the fragmenter.

2018-11-05 18:04:13 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Clean up _payload_nal_single

2018-11-05 17:55:23 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Extract sending fragments into _payload_nal_fragment

2018-11-05 17:49:52 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Extract sending a single packet into _payload_nal_single

2018-11-05 17:10:03 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Define and use FU_A_TYPE_ID

2018-11-05 17:07:06 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Use snake_case variables

2018-11-05 17:04:14 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Clean up whitespace and syntax

2019-06-06 16:05:31 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: Unlock output if the queue is full

2019-06-29 23:17:28 -0600  Thomas Bluemel <tbluemel@control4.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Ignore unsolicited rtx packets.
	  If an rtx packet arrives that hasn't been requested (it might
	  have been requested from prior to a reset), ignore it so that
	  it doesn't inadvertently trigger a clock skew.

2019-06-29 23:16:44 -0600  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Add unit test for unsolicited rtx affecting skew

2019-06-13 15:45:28 -0600  Thomas Bluemel <tbluemel@control4.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	* tests/check/elements/rtpbin.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Only calculate skew or reset if no gap.
	  In the case of reordered packets, calculating skew would cause
	  pts values to be off. Only calculate skew when packets come
	  in as expected. Also, late RTX packets should not trigger
	  clock skew adjustments.
	  Fixes #612

2019-07-02 21:21:05 +0300  Mart Raudsepp <mart.raudsepp@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Provide a 30 frames lead-in for MP3
	  mpegaudioparse suggests MP3 needs 10 or 30 frames of lead-in (depending on
	  mpegaudioversion, which we don't know here), thus provide at least 30 frames
	  lead-in for such cases as a followup to commit cbfa4531ee5ef.

2019-05-24 10:31:39 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: max-dropout-time gets cast to int32
	  So any value over MAXINT32 gets considered as negative and is silently ignored.

2019-07-02 13:00:32 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: do_seek can never be called with a NULL event

2019-07-01 22:38:41 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only adjust segment time when adjusting segment start
	  We ended up setting segment.time to segment.position when doing
	  reverse playback, which is obviously wrong.

2019-07-01 13:54:13 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: unref the event in element seek handler

2019-06-29 00:25:26 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle seek event on the element
	  Without this, the user has to wait for rtspsrc to have sent a PLAY
	  request and exposed its pads before seeking it.

2019-06-26 18:03:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Add missing socket.h include
	  Without this include, macro like SO_BINDTODEVICE is not visible and
	  associated feature gets out-compiled. This also affects the support for
	  SO_SNDBUF.

2019-06-24 17:35:15 +0200  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvmux.c:
	  flvmux: Clear new_tags if sending metadata in header
	  This avoids sending an additional metadata object right after the
	  headers.

2018-06-13 14:55:29 -0700  Song Bing <bing.song@nxp.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Fix drain() function return type
	  Return right type for drain() function.

2019-06-24 14:28:39 +0300  Mart Raudsepp <mart.raudsepp@collabora.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: add back segment clipping to parsers that have lost it
	  The pre_push_frame default clipping behaviour was introduced in 2010
	  with commit 30be03004e82 and modified with commit 4163969a2422 in 2011,
	  when most parsers didn't implement a pre_push_frame yet. Not having it
	  meant that clipping was done by default. Those that did implement a
	  pre_push_frame (flacparse and mpegaudioparse) at the time, had the flag
	  adjusted as part of the 2011 refactor work.
	  All other parsers got a pre_push_frame vfunc implementation only in
	  2013, but seem to have forgot to keep the clipping behaviour, as
	  was done automatically when a pre_push_frame implementation doesn't
	  exist for the parser. aacparse lost it with commit 91d4abcea in
	  July 2013; the others in Dec 2013 as part of AUDIO_CODEC tag posting
	  in commits 6f89b430e, d2ab5199b, 29f2cae12, 753d3c23a and 292780574.

2019-06-24 09:42:31 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2codec.c:
	  v4l2: fix compiler warning due to c99-ism

2019-06-19 14:28:28 +0200  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* tests/check/elements/flvmux.c:
	  test: flvmux: Test changing caps with one sinkpad
	  These tests segfault without the preceding crash fix.

2019-06-19 14:08:06 +0200  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* tests/check/elements/flvmux.c:
	  test: flvmux: Use gst_harness_sink_push_many
	  And check its return value.

2019-06-19 12:31:46 +0200  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvmux.c:
	  flvmux: Simplify an if-else chain
	  Merge the identical branches and turn the condition around to make it
	  easier to read.

2019-06-19 12:28:22 +0200  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvmux.c:
	  flvmux: Avoid crash when changing caps without both streams
	  mux->video_pad and mux->audio_pad can be NULL if the corresponding pad
	  has not been requested.

2019-06-12 15:57:48 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Send caps anyway if caps are pending in the adapter but are different from the new ones
	  Otherwise it can happen that we receive a caps event, then another caps
	  event and only then buffers. We would then send out the first caps event
	  in the stream but mark buffers with the caps version of the second caps
	  event.

2019-06-12 14:57:24 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstdepay.h:
	  rtpgstdepay: Only store the current caps and drop old caps immediately
	  Otherwise it can happen that we already collected 7 caps, miss the 8th
	  caps packet (packet loss) and then re-use the 1st caps for the following
	  buffers instead of the 8th caps which will likely cause errors further
	  downstream unless both caps are accidentally the same.
	  Keeping old caps around does not seem to have any value other than
	  potentially causing errors. We would always receive new caps whenever
	  they change (even if they were previous ones) and it's very unlikely
	  that they happen to be exactly the same as the previous ones.
	  Also after having received new caps or a buffer with a next caps
	  version, no buffers with old caps version will arrive anymore.

2019-06-15 02:00:43 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Clear clock master before unreffing
	  Make sure to clear any master clock on the media_clock
	  before unreffing it to release the timer callback that's
	  updating the clock and keeping it reffed.

2019-06-16 11:07:31 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-ids.c:
	  matroska: Initialise a video_context field to satisfy valgrind
	  Clear the mastering_display_info_present field explicitly
	  after reallocating the track context into a video context
	  to avoid uninitialised warnings in valgrind

2019-06-14 17:34:31 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/multifile/gstmultifilesink.c:
	  docs: Fix link to strings
	  We can't link to #gchar* this way.

2019-06-14 00:17:22 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  jitterbuffer: unset DTS on output buffers

2019-05-22 21:40:52 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: set the same seqnum on flush_start / flush_stop
	  It's currently not made mandatory by aggregator, but it might
	  eventually be, and is more consistent behaviour
	  See https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/977

2019-06-13 11:55:04 +0200  Mikhail Fludkov <misha@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: late packets shouldn't affect PTS of the following packet
	  If, say, a rtx-packet arrives really late, this can have a dramatic
	  effect on the jitterbuffer clock-skew logic, having it being reset
	  and losing track of the current dts-to-pts calculations, directly affecting
	  the packets that arrive later.
	  This is demonstrated in the test, where a RTX packet is pushed in really
	  late, and without this patch the last packet will have its PTS affected
	  by this, where as a late RTX packet should be redundant information, and
	  not affect anything.

2019-06-12 10:47:39 +0200  Mikhail Fludkov <misha@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix rtx delay calulation when large packet spacing

2016-11-24 18:18:01 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix delay for EXPECTED timers added by gaps
	  This patch corrects the delay set on EXPECTED timers that are added when
	  processing gaps. Previously the delay could be too small so that
	  'timout + delay' was much less than 'now', causing the following retries
	  to be scheduled too early. (They were sent earlier than
	  rtx-retry-timeout after the previous timeout.)

2018-11-20 16:11:12 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpstats.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: don't try and calculate packet-rate if seqnum are jumping
	  Turns out that the "big-gap"-logic of the jitterbuffer has been horribly
	  broken.
	  For people using lost-events, an RTP-stream with a gap in sequencenumbers,
	  would produce exactly that many lost-events immediately.
	  So if your sequence-numbers jumped 20000, you would get 20000 lost-events
	  in your pipeline...
	  The test that looks after this logic "test_push_big_gap", basically
	  incremented the DTS of the buffer equal to the gap that was introduced,
	  so that in fact this would be more of a "large pause" test, than an
	  actual gap/discontinuity in the sequencenumbers.
	  Once the test was modified to not increment DTS (buffer arrival time) with
	  a similar gap, all sorts of crazy started happening, including adding
	  thousands of timers, and the logic that should have kicked in, the
	  "handle_big_gap_buffer"-logic, was not called at all, why?
	  Because the number max_dropout is calculated using the packet-rate, and
	  the packet-rate logic would, in this particular test, report that
	  the new packet rate was over 400000 packets per second!!!
	  I believe the right fix is to don't try and update the packet-rate if
	  there is any jumps in the sequence-numbers, and only do these calculations
	  for nice, sequential streams.

2019-06-12 11:16:22 +0200  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix unused variables

2019-06-12 02:42:42 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Protect initial pad configuration with the object lock
	  gst_splitmux_src_activate_part() configures the pad information
	  before starting the pad task, but occasionally the changes it makes
	  to the pad are not seen in the pad task because they're not
	  protected by the right locking. Use the pad's object lock to
	  protect those variables.

2019-06-12 01:42:20 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Restart pad task on a reconfigure
	  On a reconfigure event, restart streaming on the pad so
	  that switching tracks in playbin works cleanly

2019-06-11 18:40:09 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multifile/gstsplitmuxsrc.h:
	  splitmuxsrc: Use an RW lock instead of a mutex to protect the pad list
	  Fix a deadlock around the pads list by using an RW lock to
	  allow simultaneous readers. The pad list doesn't really changes
	  except at startup and shutdown.

2019-06-11 23:18:24 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Ignore duplicate seeks
	  Use the seqnum to ignore duplicated seek events.

2019-05-29 09:20:07 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Improve debug output
	  Make the debug output less confusing by not mentioning a src
	  pad when doing calculations on the sink pad side.
	  Improve debug around why a GOP is considered overflowing a fragment

2019-05-29 09:20:07 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Give internal queues useful names
	  Makes debug output more useful

2019-06-05 23:13:33 +0300  Mart Raudsepp <mart.raudsepp@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Provide a 2 frames lead-in for audio decoders
	  AAC and various other audio codecs need a couple frames of lead-in to
	  decode it properly. The parser elements like aacparse take care of it
	  via gst_base_parse_set_frame_rate, but when inside a container, the
	  demuxer is doing the seek segment handling and never gives lead-in
	  data downstream.
	  Handle this similar to going back to a keyframe with video, in the
	  same place. Without a lead-in, the start of the segment is silence,
	  when it shouldn't, which becomes especially evident in NLE use cases.

2019-05-28 20:14:49 +0300  Mart Raudsepp <mart.raudsepp@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: remove indent exception and reindent
	  As the indent disabling isn't playing along for a following fix,
	  remove the indent disabling and reindent in a way that doesn't
	  look too stupid.

2019-03-08 14:43:20 +0000  Philippe Normand <philn@igalia.com>

	* sys/v4l2/gstv4l2h264codec.c:
	  v4l2: Fix H.264 level 3 string representation
	  The string_to_level function handles "3" so the level_to_string function should
	  do the same, to prevent caps negotiation issues.

2019-03-04 11:05:29 +0000  Philippe Normand <philn@igalia.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2codec.c:
	* sys/v4l2/gstv4l2codec.h:
	* sys/v4l2/gstv4l2fwhtenc.c:
	* sys/v4l2/gstv4l2h263enc.c:
	* sys/v4l2/gstv4l2h264codec.c:
	* sys/v4l2/gstv4l2h264codec.h:
	* sys/v4l2/gstv4l2h264enc.c:
	* sys/v4l2/gstv4l2h264enc.h:
	* sys/v4l2/gstv4l2h265codec.c:
	* sys/v4l2/gstv4l2h265codec.h:
	* sys/v4l2/gstv4l2h265enc.c:
	* sys/v4l2/gstv4l2h265enc.h:
	* sys/v4l2/gstv4l2jpegenc.c:
	* sys/v4l2/gstv4l2mpeg4codec.c:
	* sys/v4l2/gstv4l2mpeg4codec.h:
	* sys/v4l2/gstv4l2mpeg4enc.c:
	* sys/v4l2/gstv4l2mpeg4enc.h:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videodec.h:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/gstv4l2videoenc.h:
	* sys/v4l2/gstv4l2vp8codec.c:
	* sys/v4l2/gstv4l2vp8codec.h:
	* sys/v4l2/gstv4l2vp8enc.c:
	* sys/v4l2/gstv4l2vp8enc.h:
	* sys/v4l2/gstv4l2vp9codec.c:
	* sys/v4l2/gstv4l2vp9codec.h:
	* sys/v4l2/gstv4l2vp9enc.c:
	* sys/v4l2/gstv4l2vp9enc.h:
	* sys/v4l2/meson.build:
	  v4l2: Profile and level probing support for encoders and decoders
	  There used to be some profile/level support in encoders. This code was moved to
	  GstV4l2Codecs and is now also used for decoders. The caps templates for the
	  H.264, H.265, MPEG4, VP8 and VP9 encoders and decoders should now reflect the
	  profiles and levels advertised by the kernel.

2019-06-03 16:21:12 -0400  Aaron Boxer <aaron.boxer@collabora.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix typo in property description

2019-06-04 13:39:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/gst-plugins-good.supp:
	  supp: Ignore leaks caused by shout/sethostent
	  sethostent() seems to be using a global state and we endup with leaks from
	  that API when called through shout_init(). We had the option to only
	  ignore the shout case, but the impression is that if we have shout and
	  another sethostend user, as it's a global state, we may endup with a
	  different stack trace for the same leak. So in the end, we just ignore
	  memory allocated by sethostent in general.

2019-04-30 17:28:25 -0400  Thibault Saunier <tsaunier@igalia.com>

	* ext/pulse/pulsedeviceprovider.c:
	  pulse-device: Hide the alsa device provider if we provide alsa devices

2019-05-21 15:25:03 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* tests/check/elements/rtpssrcdemux.c:
	  rtpssrcdemux: Avoid taking streamlock out-of-band
	  In this change we now protect the internal srcpads list using the
	  stream lock and limit usage of the internal stream lock to
	  preventing data flowing on the other src pad type while creating
	  and signalling the new pad.
	  This fixes a deadlock with RTPBin shutdown lock. These two locks would
	  end up being taken in two different order, which caused a deadlock. More
	  generally, we should not rely on a streamlock when handling out-of-band
	  data, so as a side effect, we should not take a stream lock when
	  iterating internal links.

2019-05-27 18:08:54 +0900  Damian Hobson-Garcia <dhobsong@igel.co.jp>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Orphan buffer pool on object_stop if supported
	  Use V4L2 buffer orphaning, on recent kernels so that
	  the device can be restarted immediately with
	  a new buffer pool during renogatiation.

2019-05-30 13:12:31 +0900  Damian Hobson-Garcia <dhobsong@igel.co.jp>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Free orphaned allocator resources when buffers are released
	  Allocator resources cannot be freed when a buffer pool is orphaned
	  while its buffers are in use. They should, however, be freed once those
	  buffers are no longer needed. This patch disposes of any buffers
	  belonging to an orphaned pool as they are released, and makes sure
	  that the allocator is cleaned up when the last buffer is returned.

2019-05-30 11:13:07 +0900  Damian Hobson-Garcia <dhobsong@igel.co.jp>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: return TRUE when buffer pool orphaning succeeds
	  When trying to orphan a buffer pool, successfully return and unref
	  the pool when the pool is either successfully stopped or orphaned.
	  Indicate failure and leave the pool untouched otherwise.

2019-05-31 23:04:11 +0200  Niels De Graef <niels.degraef@barco.com>

	* configure.ac:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudpsrc.c:
	* meson.build:
	  meson: Bump minimal GLib version to 2.44
	  This means we can use some newer features and get rid of some
	  boilerplate code using the G_DECLARE_* macros.
	  As discussed on IRC, 2.44 is old enough by now to start depending on it.

2018-09-05 21:10:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Use size of first closed caption buffer in prefill mode
	  It must be accurate for all samples to work in Final Cut properly, so
	  the best we can do is to assume that all samples are the same as the
	  first. Bigger samples are truncated, smaller samples are padded.

2019-05-29 22:06:58 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/meson.build:
	* ext/lame/gstlamemp3enc.c:
	* ext/mpg123/gstmpg123audiodec.c:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/twolame/gsttwolamemp2enc.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/level/gstlevel.c:
	* gst/rtp/gstrtpL8depay.c:
	* gst/rtp/gstrtpL8pay.c:
	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpredenc.c:
	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/spectrum/gstspectrum.c:
	* sys/v4l2/gstv4l2object.c:
	  doc: remove xml from comments

2019-05-29 11:02:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	  docs: update plugins cache
	  And add gtk+ and qt plugins

2019-05-29 10:58:40 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/dv/meson.build:
	* ext/gtk/meson.build:
	* ext/qt/meson.build:
	* sys/osxaudio/meson.build:
	* sys/osxvideo/meson.build:
	* sys/waveform/meson.build:
	  dv, gtk, qt, osxaudio, osxvideo, waveform: add to plugins list
	  Makes sure the paths for these plugins are included in the
	  uninstalled plugin paths list. And also for the docs.
	  Fixes #604

2019-04-18 15:31:00 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: Add new property to offset all streams to start at zero
	  This takes the timestamp of the earliest stream and offsets it so that
	  it starts at 0. Some software (VLC, ffmpeg-based) does not properly
	  handle Matroska files that start at timestamps much bigger than zero.
	  Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/449

2019-05-28 14:13:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gdepay.h:
	  rtpmp4gdepay: don't spam debug log for broken ADTS-in-RTP AAC
	  Print warning only once.

2019-05-22 18:06:04 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Only set running time on finalizing sink element when in async-finalize mode
	  There is only a single sink element in async-finalize mode, and we would
	  keep the running time from previous fragments set in that case. As we
	  don't ever set the running time for the very last fragment on EOS, this
	  would mean that the closing time reported for the very last fragment is
	  the same as the closing time of the previous fragment.

2015-03-26 13:08:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Remove uneeded keep-alive hack
	  The rtsp connection code has been fixed now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=744209

2019-05-26 17:46:06 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Print GstClockTimeDiff as GST_STIME_FORMAT

2019-05-25 19:45:02 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	  doc: update plugin cache

2019-05-25 17:25:02 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: the documentation for GstVideoMixer2Pad is not exposed

2019-05-25 16:56:32 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/spectrum/gstspectrum.c:
	  doc: fix element section documentations
	  Element sections were not rendered anymore after the hotdoc
	  port, fixing this revealed a few incorrect links.

2019-02-19 12:15:19 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Improve RTPStorage action signal documentation
	  This is a tiny clarification as the storage was loosely named "storage".
	  This change clarify that the storage is specificaly used for received RTP
	  packets. This is unlike the storage found in rtprtxsend that stores a
	  backlog of sent RTP packets.

2019-05-05 22:16:36 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: Add BT2020_10, PQ and HLG transfer functions
	  The direct use of newly added transfer functions

2019-05-23 12:38:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/aalib/meson.build:
	  aasink: Generate pkg-config file for the plugin

2019-05-22 11:01:17 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: Fix documentation of max-file-duration property
	  The max-file-duration property works with max-duration mode

2019-05-14 17:36:14 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Always keep at least one NACK on early RTCP
	  We recently added code to remove outdate NACK to avoid using bandwidth
	  for packet that have no chance of arriving on time. Though, this had a
	  side effect, which is that it was to get an early RTCP packet with no
	  feedback into it. This was pretty useless but also had a side effect,
	  which is that the RTX RTT value would never be updated. So we we stared
	  having late RTX request due to high RTT, we'd never manage to recover.
	  This fixes the regression by making sure we keep at least one NACK in
	  this situation. This is really light on the bandwidth and allow for
	  quick recover after the RTT have spiked higher then the jitterbuffer
	  capacity.

2019-05-16 09:14:19 -0400  Thibault Saunier <tsaunier@igalia.com>

	* docs/meson.build:
	  docs: Stop building the doc cache by default
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36

2019-05-13 22:53:59 -0400  Thibault Saunier <tsaunier@igalia.com>

	* docs/gst_plugins_cache.json:
	  docs: Update plugins documentation cache

2019-04-23 12:28:23 -0400  Thibault Saunier <tsaunier@igalia.com>

	* ext/soup/gstsouputils.c:
	* gst/goom/flying_stars_fx.c:
	* gst/goom/goom_tools.h:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpmux.h:
	* sys/v4l2/gstv4l2object.c:
	  doc: Fix some docstrings

2018-10-22 11:39:55 +0200  Thibault Saunier <tsaunier@igalia.com>

	* Makefile.am:
	* configure.ac:
	* docs/Makefile.am:
	* docs/all_index.md:
	* docs/gst_api_version.in:
	* docs/gst_plugins_cache.json:
	* docs/index.md:
	* docs/meson.build:
	* docs/plugins/.gitignore:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/gst-plugins-good-plugins.types:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-directsound.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-osxaudio.xml:
	* docs/plugins/inspect/plugin-osxvideo.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-waveform.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* docs/random/ChangeLog-0.8:
	* docs/random/PORTED_09:
	* docs/sitemap.txt:
	* docs/version.entities.in:
	* ext/aalib/meson.build:
	* ext/cairo/meson.build:
	* ext/flac/meson.build:
	* ext/gdk_pixbuf/meson.build:
	* ext/jack/meson.build:
	* ext/jpeg/meson.build:
	* ext/lame/meson.build:
	* ext/libcaca/meson.build:
	* ext/libpng/meson.build:
	* ext/mpg123/meson.build:
	* ext/pulse/meson.build:
	* ext/raw1394/meson.build:
	* ext/shout2/meson.build:
	* ext/soup/meson.build:
	* ext/speex/meson.build:
	* ext/taglib/meson.build:
	* ext/twolame/meson.build:
	* ext/vpx/meson.build:
	* ext/wavpack/meson.build:
	* gst/alpha/meson.build:
	* gst/apetag/meson.build:
	* gst/audiofx/meson.build:
	* gst/audioparsers/meson.build:
	* gst/auparse/meson.build:
	* gst/autodetect/meson.build:
	* gst/avi/meson.build:
	* gst/cutter/meson.build:
	* gst/debugutils/meson.build:
	* gst/deinterlace/meson.build:
	* gst/dtmf/meson.build:
	* gst/effectv/meson.build:
	* gst/equalizer/meson.build:
	* gst/flv/meson.build:
	* gst/flx/meson.build:
	* gst/goom/filters.c:
	* gst/goom/meson.build:
	* gst/goom2k1/meson.build:
	* gst/icydemux/meson.build:
	* gst/id3demux/meson.build:
	* gst/imagefreeze/meson.build:
	* gst/interleave/meson.build:
	* gst/isomp4/meson.build:
	* gst/law/meson.build:
	* gst/law/mulaw-conversion.c:
	* gst/level/meson.build:
	* gst/matroska/meson.build:
	* gst/monoscope/meson.build:
	* gst/multifile/meson.build:
	* gst/multipart/meson.build:
	* gst/replaygain/meson.build:
	* gst/rtp/meson.build:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/meson.build:
	* gst/rtsp/meson.build:
	* gst/shapewipe/meson.build:
	* gst/smpte/meson.build:
	* gst/spectrum/meson.build:
	* gst/udp/meson.build:
	* gst/videobox/meson.build:
	* gst/videocrop/meson.build:
	* gst/videofilter/meson.build:
	* gst/videomixer/meson.build:
	* gst/wavenc/meson.build:
	* gst/wavparse/meson.build:
	* gst/y4m/meson.build:
	* meson.build:
	* meson_options.txt:
	* sys/directsound/meson.build:
	* sys/oss/meson.build:
	* sys/oss4/meson.build:
	* sys/v4l2/meson.build:
	* sys/ximage/meson.build:
	  doc: Port documentation to hotdoc

2018-11-12 08:05:45 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst/isomp4/gstqtmux.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  Mark some properties as DOC_SHOW_DEFAULT

2018-10-22 11:39:24 +0200  Thibault Saunier <tsaunier@igalia.com>

	* ext/aalib/gstaasink.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/audiofx/gstscaletempo.c:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/equalizer/gstiirequalizernbands.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/flv/gstindex.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/filters.c:
	* gst/goom/goom_config.h:
	* gst/goom/goom_filters.h:
	* gst/goom/goom_plugin_info.h:
	* gst/goom/gstgoom.c:
	* gst/goom/ifs.c:
	* gst/goom/sound_tester.h:
	* gst/goom2k1/filters.h:
	* gst/goom2k1/goom_core.h:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/interleave/deinterleave.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-conversion.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpL24pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvpay.c:
	* gst/rtp/gstrtpstreamdepay.c:
	* gst/rtp/gstrtpstreampay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpptdemux.h:
	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/spectrum/gstspectrum.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videomixer/videomixer2.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* gst/y4m/gsty4mencode.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tunerchannel.c:
	* sys/v4l2/tunernorm.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	  docs: Port all docstring to gtk-doc markdown

2019-05-02 22:14:35 -0700  Thiago Santos <thiagossantos@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: do not try to send EOS with invalid seqnum
	  The second udpsrc (rtcp) might not have seen the segment event if it was
	  not enabled or if rtcp is not available on the server. So if the
	  application tries to send an EOS event it will try to set an invalid
	  seqnum to the event.

2019-04-24 13:54:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Add more information to probation warning

2019-04-24 13:47:54 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Call on-new-ssrc earlier
	  Right now, we may call on-new-ssrc after we have processed the first
	  RTP packet. This prevents properly configuring the source as some
	  property like "probation" are copied internally for use as a
	  decreasing counter. For this specific property, it prevents the
	  application from disabling probation on auxiliary sparse stream.
	  Probation is harmful on sparse streams since the probation algorithm
	  assume frequent and contiguous RTP packets.

2019-02-19 13:34:49 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Write MasteringMetadata and Max{CLL,FALL}
	  Enable muxing with HDR meta data if upstream provided it

2019-02-18 23:28:50 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Add support parsing HDR metadata
	  Set SMPTE ST 2086 mastering-display-metadata and
	  content-light-level to caps, if any

2019-02-19 18:27:23 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	* gst/matroska/matroska-read-common.h:
	  matroska: Remove white space

2019-05-01 10:00:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtprawdepay: Don't get rid of the buffer pool on FLUSH_STOP
	  We expect there to be a pool as long as the caps are known and
	  FLUSH_STOP is not resetting the caps. Getting rid of the pool would
	  cause assertions.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/584

2019-02-08 10:09:17 +0100  Danny Smith <dannys@axis.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Free storage when freeing session

2019-04-25 21:52:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Fix typo in error message

2019-04-25 11:19:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Only set the DISCONT flag on the first buffer after segment start

2019-04-24 02:38:32 +0900  okuoku <mjt@cltn.org>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: Use jack_free(3) to release ports
	  Port objects acquired with jack_get_ports() need to be freed with
	  jack_free(3), not stdlib free().
	  On Windows, Jack may be linked against different libc than GStreamer
	  libraries so free()ing port objects directly might cause crash because
	  of libc mismatch.

2019-04-23 10:10:01 +0100  Philippe Normand <philn@igalia.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Advertise interleaved layout in caps templates
	  Scaletempo doesn't support non-interleaved layout. Not explicitely stating this
	  would trigger critical warnings and a caps negotiation failure when scaletempo
	  is used as playbin audio-filter.
	  Patch suggested by George Kiagiadakis <george.kiagiadakis@collabora.com>.
	  Fixes #591

2019-04-21 20:12:28 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/matroska/meson.build:
	  meson: matroska: Ensure header dependency not only library
	  Library existence does not guarantee header.

2018-11-13 13:48:11 +0100  Robert Rosengren <robertr@axis.com>

	* gst/udp/gstmultiudpsink.c:
	  multidupsink: Use gst_net_utils_set_socket_tos for QoS DSCP
	  Util function in net library exists for setting QoS DSCP on socket, hence
	  use it to simplify code.

2019-04-19 10:27:38 +0100  Tim-Philipp Müller <tim@centricular.com>

	* README:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* meson.build:
	  Back to development

=== release 1.16.0 ===

2019-04-19 00:23:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* README:
	* RELEASE:
	* configure.ac:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.16.0

2019-04-19 00:23:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  Update docs

2019-04-19 00:23:14 +0100  Tim-Philipp Müller <tim@centricular.com>

	* po/el.po:
	* po/zh_CN.po:
	  Update translations

2019-04-18 17:14:18 +0200  Benjamin Sigonneau <b.sigonneau@mediadone.fr>

	* ext/qt/qtplugin.pro:
	  qmlglsink: fix compilation with Qt >= 5.5 on Windows
	  As of Qt >= 5.5, qmake do not link to opengl32 by default anymore. This commit adds opengl32.lib to the .pro
	  file so that the plugin can be build using QtCreator on Windows.

2019-04-17 15:48:26 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  meson: Build qt plugin in C++11 mode explicitly
	  This works implicitly most of the time, but we need to set it
	  explicitly for building with Android.

2019-04-16 14:35:06 +0530  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: fix use after free when handling events
	  The sink_event parent function may consume the event so we shouldn't use
	  it after having calling it.

=== release 1.15.90 ===

2019-04-11 00:26:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.15.90

2019-04-11 00:26:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  Update docs

2019-04-09 23:51:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	  rtpulpfecdec,enc: unbreak plugin gtk-doc build in autotools
	  Fix doc chunks to not use that syntax for links that have the
	  url as description, it will be put verbatim into the xml/*.xml
	  file and then the expat parser will throw a syntax error like:
	  File "../../common/mangle-db.py", line 71, in <module>
	  main()
	  File "../../common/mangle-db.py", line 69, in main
	  patch (details.replace("-details", ""), os.path.basename(details))
	  File "../../common/mangle-db.py", line 20, in patch
	  doc = xml.dom.minidom.parse(related)
	  File "/usr/lib/python2.7/xml/dom/minidom.py", line 1918, in parse
	  return expatbuilder.parse(file)
	  File "/usr/lib/python2.7/xml/dom/expatbuilder.py", line 924, in parse
	  result = builder.parseFile(fp)
	  File "/usr/lib/python2.7/xml/dom/expatbuilder.py", line 207, in parseFile
	  parser.Parse(buffer, 0)
	  xml.parsers.expat.ExpatError: not well-formed (invalid token): line 84, column 7

2019-04-08 11:35:34 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: preserve GST_BUFFER_FLAG_DISCONT on the first outputted buffer
	  If the incoming frame buffer has GST_BUFFER_FLAG_DISCONT set this should
	  be preserved and set for the first output buffer too, like other
	  payloaders do.
	  Spotted with gst-validate-1.0 when adding integration tests for
	  rtpsession, a minimal test to reproduce the issue is:
	  $ gst-validate-1.0 videotestsrc num-buffers=1 ! rtpvrawpay ! identity ! fakesink
	  Starting pipeline
	  Pipeline started
	  warning : Buffer didn't have expected DISCONT flag333 speed: 1.000000 />
	  Detected on <identity0:sink>
	  Detected on <identity0:src>
	  Detected on <fakesink0:sink>
	  Description : Buffers after SEGMENT and FLUSH must have a DISCONT flag
	  Issues found: 1
	  =======> Test PASSED (Return value: 0)

2019-03-22 12:42:14 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	  rtpulpfec*: Replace github URIs with gitlab.fdo ones

2019-03-21 17:01:11 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpredenc.c:
	  rtpred*: Add example pipelines

2019-03-21 16:48:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	  rtpulpfec*: Improve documentation

2019-03-20 18:31:48 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpstorage.c:
	* gst/rtp/gstrtpulpfecdec.c:
	  rtpstorage + rtpulpfecdec: Get the storage using a query as fallback
	  This allows it to be used using gst-launch for easier testing.

2019-03-19 06:22:29 -0700  Dan Kegel <dank@kegel.com>

	* sys/osxvideo/Makefile.am:
	* sys/osxvideo/meson.build:
	  osxvideo: fix mac os 10.14 build
	  lockFocusIfCanDraw is deprecated in mac os 10.14.  Apple suggests a
	  different way to do what that does, but for now, just suppress the deprecation.
	  There's no way to disable just that deprecation, so shut them all down.
	  OpenGL is also deprecated in mac os 10.14.  There is a gentle way to
	  turn off just those deprecations (GL_SILENCE_DEPRECATION), but since
	  this commit turns them all off, that's moot.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/577

2019-04-07 12:00:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtpsession.c:
	  test: rtpsession: Verify on-sending-nacks callback

2019-03-27 16:19:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Allow overriding NACK packet creation
	  This introduce a new signal on RTSession, on-sending-nacks is emited
	  right before the list of seqnums to be nacked are processed and
	  transformed into FB Nack. This allow implementing custom nacks
	  handling through another mechanism with APP feedback.

2018-11-20 02:45:04 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Add disable-sr-timestamp property
	  The Onvif Streaming Spec, in section 6.11, mandates that when
	  Rate-Control is disabled potential RTCP packets shall have
	  their timestamps set to 0.
	  <https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf>

2019-03-05 20:57:44 +0100  Philipp Zabel <philipp.zabel@gmail.com>

	* sys/v4l2/ext/types-compat.h:
	  v4l2: remove __user define from types-compat.h
	  Remove the now unused __user define.

2019-03-05 20:53:47 +0100  Philipp Zabel <philipp.zabel@gmail.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: use opRGB colorspace and xfer func defines
	  AdobeRGB defines have been renamed to opRGB in the kernel headers,
	  use the new names.

2019-01-24 16:12:13 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: support orphaning
	  Recent kernels allow REQBUFS(0) on a queue that still has buffers in
	  use (mmapped or exported via dmabuf), orphaning all buffers on the queue.
	  If this is supported, the v4l2videodec element does not have to send a
	  drain request downstream.

2019-01-24 16:12:13 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: support orphaning
	  Now that the v4l2allocator allows orphaning the V4L2 buffer queue, add
	  support for orphaning in the v4l2bufferpool. gst_v4l2_buffer_pool_orphan
	  can be used as a replacement for gst_v4l2_buffer_pool_stop, without
	  having to wait for buffers to be returned to the pool.

2019-01-24 16:12:13 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	  v4l2allocator: support orphaning
	  Recent kernels allow REQBUFS(0) on a queue that still has buffers in
	  use (mmapped or exported via dmabuf), orphaning all buffers on the queue.
	  Orphaning the allocator causes it to release all buffers with
	  REQBUFS(0), even if they are still in use. An orphaned allocator can
	  only be stopped. It can not be restarted or create new buffers.

2019-01-24 15:36:49 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/ext/v4l2-common.h:
	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	  v4l2: update kernel headers to latest from media tree
	  Update to the latest installed headers (output of make headers_install)
	  from the media tree, keeping the slight modifications to the includes.
	  This includes new HEVC controls, the AdobeRGB -> opRGB rename, a new
	  capabilities field for v4l2_requestbuffers and v4l2_create_buffers, new
	  32-bit YUV formats, and request_fd changes.

2019-04-03 14:13:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/shout2/gstshout2.c:
	  shout2: Fix leak on error in start

2019-03-29 22:48:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtpsession.c:
	  test: rtpsession: Test FB Nack packing
	  We used to split the NACK if a smaller seqnum of a range of seqnum was
	  submited. This test also make sure that the three operations (append,
	  prepend, update) works properly.

2019-03-29 22:34:47 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtpsession.c:
	  test: rtpsession: Test handling of NACK surplus
	  This test verify that NACKs that didn't fit in one packet are properly
	  filtered and inserted into the following pipeline.

2019-03-25 13:42:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: Send as many nack seqnum as possible
	  In order to do that, we now split the nacks registration from the actual
	  FB nack packet construction. We then try and add as many FB Nacks as
	  possible into the active packets and leave the remaining seqnums in the
	  RTPSource. In order to avoid sending outdated NACK later on, we save the
	  seqnum calculated deadline and cleanup the outdated seqnums before the
	  next RTCP send.
	  Fixes #583

2018-04-30 10:54:19 +0200  John Bassett <john.bassett@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Fix race when sending PLI, FIR and NACK packets
	  Calling rtp_session_send_rtcp before marking the source as requiring a
	  pli/fir/nack meant the rtcp_thread could be scheduled and start running
	  before the source was updated. This meant the request would not be sent
	  early but instead was transmitted with the next regular RTCP packet.
	  Add test for nack generation.

2019-03-29 16:49:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Fix early rtcp time comparision
	  If the current time is equal to the early rtcp time deadline, there is
	  no need to schedule a timer. This ensure that immediate feedback is
	  really immediate and simplify implementing unit tests with the test
	  clock, which stops perfectly on the timeout time.
	  This fix has been extracted from Pexip feature patch called
	  "rtpsession: Allow instant transmission of RTCP packets"

2019-01-24 11:54:49 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: preserve features when fixating caps
	  The caps features were lost when sorting caps structures in
	  gst_v4l2src_fixate(). This was breaking alternate as
	  GST_CAPS_FEATURE_FORMAT_INTERLACED was removed from the caps.

2018-11-13 21:23:30 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Set DELTA_UNIT flag when appropriate
	  When used in combination with a rtponviftimestamp element
	  downstream, forwarding this flag ensures it gets correctly
	  serialized in the ONVIF header extension.

2019-04-03 16:42:26 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  docs: fix typo s/abonormally/abnormally/

2019-04-03 16:38:56 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/debugutils/gsttaginject.c:
	* gst/goom2k1/gstgoom.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/rtp/README:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/smpte/gstsmpte.c:
	  docs: fix typo s/incomming/incoming/

2019-04-03 16:34:22 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: fix indentation after G_DEFINE_TYPE
	  A missing colon after G_DEFINE_TYPE declaration was confusing gst-indent
	  and causing problem in the pre-commit hook.
	  Add the missing colon and fix the following function declaration to
	  follow the normal GStreamer style.

2019-03-07 15:34:03 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: fix comment to refer to buffers instead of groups
	  One comments in gst_rtp_session_chain_send_rtp_common() is referring to
	  groups in a buffer list, however this concept of "group" comes from
	  GStreamer 0.10 and does not exist anymore in GStreamer 1.0, so update the
	  comment to refer to buffers instead.

2019-03-06 09:52:45 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: add comment to explain why probation queue is not always cleared

2019-04-02 12:51:04 +0200  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: add test to verify that stats are correct
	  Add a test to verify that stats about sent and received packets are
	  correct even when using buffer lists.
	  NOTE: the newly introduced get_session_source_stats() selects the
	  desired source (sender or receiver) by filtering them by type (using the
	  get_sender parameter) rather than by ssrc because this simplifies the
	  code and it's good enough for testing purposes as there is usually one
	  source per type in the test setup.
	  Filtering by ssrc would have required handling asynchronous signals like
	  "on-new-sender-ssrc", with the relative locking, just to retrieve the
	  actual ssrc of the sender.

2019-03-05 13:43:12 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: fix stats about received packets
	  The update_receiver_stats() function is called also when sending packets
	  in rtp_source_send_rtp(), and sending packets may happen using a buffer
	  list rather than individual buffers.
	  So update the stats using the actual number of packets sent.
	  NOTE: this is fine for the receive path too (rtp_process_send_rtp)
	  because the receive path does not support buffer lists and
	  pinfo->packets would always be equal to 1 in this case.

2019-03-11 10:08:21 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: move buffer list creation next to its validation
	  The tests create a buffer list and then use the chain_list callback to
	  verify that the correct packets have been pushed.
	  Move the creation and validation code next to each other so that the
	  reader can more easily understand what is going on.
	  While at it add some comments to introduce the two related functions.

2019-03-06 19:27:01 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: set the chain_list function directly in the test
	  The helper function set_chain_function does not really do anything useful, remove it.

2019-03-06 19:19:03 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: make check_packet more flexible
	  Make it possible to differentiate between the position in the list and
	  the packet index in the global structures in check_packet, in some
	  future case the list may change, in case some element removes a buffer
	  from the list, and the two indices may not coincide.

2019-03-05 12:47:29 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: factor out a function to create packets buffers

2019-03-04 11:27:33 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: check if the chain_list function has been called
	  Make the test more useful to verify that the chain list function has
	  actually been called.

2019-02-27 12:27:21 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/rtpbin_buffer_list.c:
	  test: rtpbin_buffer_list: port to GStreamer 1.0
	  Port the rtpbin_buffer_list test to GStreamer 1.0 and re-enable it.
	  Some other changes include:
	  - the check on the caps has been moved from the buffer level to the
	  pad level;
	  - remove underscore prefix from static functions names, this is not
	  idiomatic in C and rarely used in the other tests;
	  - the unused header_buffer variable has been removed;
	  - check_group() has been renamed to check_packet() because in
	  GStreamer 1.0 there is no concept of "group" anymore, the comments
	  have also been updated to reflect this.

2019-04-01 18:20:53 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/jpegdec.c:
	  tests: jpegdec: bump discoverer timeout for valgrind
	  Tests might take a bit longer, esp. when run under valgrind
	  and/or they're running on the CI with other things going on,
	  so let's just bump the timeout to something higher and let
	  the test runner time us out if needed.

2019-04-01 18:20:28 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  meson: Only ensure that moc is available on Linux
	  On other OSes, it's not possible to have qmake or the qt5 pkg-config
	  files and not have moc, and `moc` will not be in `PATH`, so this only
	  causes problems.

2019-03-21 18:24:43 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/rtpstoragestream.c:
	  rtpstorage: Limit the queue size
	  Limit to the queue size in case there is no arrival time or in case there is
	  a huge flood of packets.

2019-03-18 15:30:54 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Request the FEC decoder even if ignore-pt is set

2019-03-18 15:27:21 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Factor out the code that exposes the src pad

2019-03-22 02:08:01 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpreddec.c:
	  rtpreddec: Add some more debug prints

2019-03-21 17:32:18 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/rtpstorage.c:
	  rtpstorage: Issue warning if request by size if 0
	  If the size is 0, then nothing will ever be in the storage, if a request is
	  received, it generally implies a misconfigured pipeline.

2019-03-21 17:24:42 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpstorage.c:
	* gst/rtp/rtpstorage.c:
	* gst/rtp/rtpstoragestream.c:
	  rtpstorage: Add more debug messages

2019-03-21 17:12:53 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpstorage.c:
	* gst/rtp/rtpstorage.c:
	* gst/rtp/rtpstoragestream.c:
	* gst/rtp/rtpstoragestream.h:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpstorage.c:
	* tests/check/meson.build:
	  rtpstorage: Make debug category available to sub objects

2019-03-21 17:12:33 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpstorage.c:
	  rtpstorage: Add debug funcptr to chain function

2019-03-22 12:01:01 +0100  Julian Bouzas <julian.bouzas@collabora.com>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	  flac: report latency in flacenc and flacdec
	  The FLAC specification states that the data is processed in blocks, regardless of the number of channels. Thus, The latency can be calculated using the blocksize and rate. For example a 1 second block sampled at 44.1KHz has a blocksize of 44100

2019-03-22 23:36:42 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/rtsp/test-onvif.c:
	  examples: rtsp: fix compiler warning
	  "control reaches end of non-void function"

2019-03-22 15:07:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: Remove set but not use running-time

2019-03-19 09:50:04 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpmanager: Register chain functions to debug

2019-02-27 15:49:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Allow reusing the sender AUX bin
	  This is needed for the case you don't know in advance all the sessions
	  you will be using, but would like to place all the related AUX element
	  in the same GstBin. As per current implementation, each time an sender
	  AUX bin is requested and returned, RTPBin will walk the src pads and
	  create sessions for these pads.
	  In the current implementation, if a src pad already have a sessions, it
	  returns an error and stops. As a side effect, if an AUX bin is reused in
	  a following AUX bin request, it can only work if the pads are created on
	  the last request.
	  This change simply relax the restriction in order to keep walking, and
	  just ensure that all newly created pads have a sessions.

2018-06-25 17:49:07 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: set GstVideoCodecFrame sync point flag
	  The V4L2 elements already set the delta unit buffer flag when dequeueing
	  the buffer, but gst_video_encoder_finish_frame overwrites it from the
	  passed codec frame's sync point flag. Set the flag correctly.

2018-08-23 11:47:14 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: improve stats about rtx requests

2019-03-20 15:45:35 -0400  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: Improve looging of not found RTX packet
	  When an RTX packet is not found, display a message that say if the
	  packet have not arrived yet or if it was already removed from the RTX
	  packet queue.

2018-08-09 16:40:26 +0300  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Remove unused rtp_session_create_source

2019-03-21 11:17:08 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: add -Wno-unused also to C++ args when gst debug system is disabled
	  And check if argument is supported instead of just passing it blindly,
	  and make meson code slightly cleaner, centralising the argument setting
	  in one place.

2019-03-10 19:30:50 +0000  Piotr Drąg <piotrdrag@gmail.com>

	* po/LINGUAS:
	  Update LINGUAS

2019-03-19 12:31:35 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't pass zero to denominator for framerate
	  Need to respect return of gst_video_guess_framerate() to ensure
	  non-zero denominator.
	  This patch is to fix below error with an abnormal (but has valid frame) file.
	  (gst-play-1.0:17940): GStreamer-CRITICAL **: passed '0' as denominator for `GstFraction'

2019-03-05 09:43:47 +0000  Philippe Normand <philn@igalia.com>

	* sys/v4l2/gstv4l2fwhtenc.c:
	* sys/v4l2/gstv4l2h263enc.c:
	* sys/v4l2/gstv4l2h264enc.c:
	* sys/v4l2/gstv4l2h265enc.c:
	* sys/v4l2/gstv4l2jpegenc.c:
	* sys/v4l2/gstv4l2mpeg4enc.c:
	* sys/v4l2/gstv4l2vp8enc.c:
	* sys/v4l2/gstv4l2vp9enc.c:
	  v4l2: Set Hardware classifier on encoders

2019-02-27 11:56:20 +0000  Philippe Normand <philn@igalia.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Set Hardware classifier on video decoders

2019-03-01 14:58:24 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: don't segfault if flushed without pools
	  The v4l2output and v4l2capture v4l2objects can have pool == NULL if they
	  have been stopped before.

2019-02-07 11:58:19 +0000  Charlie Turner <cturner@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Find mp4a esds atoms in protected streams sample description tables.
	  This problem was found in Test. 2 of the YouTube 2018 EME
	  tests[1]. The code was accidentally not finding an mp4a's esds atom in
	  the sample description table when the stream was encrypted. It assumed
	  that if the stream is protected, then only an enca atom will be found
	  here. What happens with YouTube is they often provide protected
	  content with a few seconds of clear content, and then switch to the
	  encrypted stream.
	  The failure case here was an incorrect codec_data field being sent
	  into aacparse. The advertisement of stereo audio @ 44.1kHz for the
	  mp4a (unprotected) stream was incorrect. As usual, the esds contained
	  the real values here which were mono at 22050 Hz.
	  Here's what the MP4 tree looks like for these types of files,
	  demonstrating why the code was making a wrong assumption (or maybe
	  YouTube is being unusual),
	  [ftyp] size=8+16
	  ...
	  [moov] size=8+1571
	  ...
	  [trak] size=8+559
	  ...
	  [stsd] size=12+234
	  entry-count = 2
	  [enca] size=8+147
	  channel_count = 2
	  sample_size = 16
	  sample_rate = 44100
	  [esds] size=12+27
	  ...
	  ...
	  [mp4a] size=8+67
	  channel_count = 2
	  sample_size = 16
	  sample_rate = 44100
	  [esds] size=12+27
	  ...
	  In addition to fixing this, the checks for esds atoms in mp4a and mp4v
	  have been made symmetrical. While I haven't seen a test case for video
	  with the same problem, it seemed better to make the same checks. This
	  also fixes a crash reported from another user[2], they also noted the
	  asymmetry with mp4v and mp4a.
	  [1] https://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2018.html?test_type=encryptedmedia-test
	  [2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/398

2019-03-15 10:41:20 +0100  Andreas Frisch <afrisch@make.tv>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix scale of time values in warning message

2019-03-15 09:18:00 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't remove udpsrc/sink from rtspsrc if they were not added to it
	  This can happen in various error cases that could happen between the
	  creation of the element in question and the adding to the rtspsrc.
	  It causes an ugly critical warning right now but is otherwise harmless.

2019-03-13 14:00:10 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* tests/check/elements/imagefreeze.c:
	  test: imagefreeze: add test for the num-buffers property

2019-03-13 13:03:44 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: add a num-buffers property
	  The imagefreeze element can be handy for benchmarking downstream
	  elements because it re-uses the same buffer memory and introduces less
	  overhead compared to always creating new frames with videotestsrc.
	  However it's not possible to make imagefreeze send EOS when using
	  gst-launch-1.0.
	  Add a num-buffers property to make it look more like a source in the
	  above scenario.

2019-03-12 16:52:45 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: add support for new color primaries

2019-03-07 11:24:38 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: fix pool-less allocation query handling
	  This fixes a critical warning if the last-sample property is enabled:
	  (gst-launch-1.0:391): GStreamer-CRITICAL **: 01:12:57.428: gst_object_unref: assertion 'object != NULL' failed
	  If the allocation query does not contain any allocation pools,
	  gst_query_parse_nth_allocation_pool will leave the local pool,
	  min, and max variables undefined, so check the array length first.
	  If pool is NULL, do not call gst_object_unref.

2019-03-08 11:03:31 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* tests/examples/meson.build:
	  meson: Build v4l2 example only if v4l2 plugin was built
	  Otherwise v4l2 example will be built with MSVC

2019-03-07 12:38:41 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* ext/dv/gstdvdemux.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsrc.c:
	  docs: fix typos s/recieve/receive/

2019-02-28 12:32:51 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: fix documentation of rtp_source_send_rtp parameters
	  In commit 28e5f9098 (rtpbin: use PacketInfo for the sender, 2013-09-13)
	  the rtp_source_send_rtp signature changed but the documentation was not
	  adjusted to match the new one.
	  Update the documentation to match the function signature.

2019-03-06 12:59:52 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: fix typo in a comment, s/SESSION_LOCK/RTP_SESSION_LOCK/
	  Fix a typo in a comment, mainly to avoid confusing autocompletion in
	  text editors.

2019-02-27 16:45:54 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: fix typos and update parameters names in comments
	  Some functions now accept a generic 'gpointer data' parameter because
	  they can work either on a single buffer or a buffer list.
	  However the comments were still referring to the old 'GstBuffer *buffer'
	  parameter, so update the comments to match the actual functions
	  signature.

2019-03-06 16:28:34 +0100  Antonio Ospite <antonio.ospite@collabora.com>

	* gst/rtpmanager/rtpstats.h:
	  rtpstats: fix some fields names in the RTPSourceStats documentation
	  Fix documentation of RTPSourceStats to use the actual fields names.

2019-03-06 17:40:12 +0000  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtpulpfecdec.c:
	  rtpulpfdecdec: only put recovered packet back into storage if not recovered from there

2019-03-06 17:38:03 +0000  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtpulpfecdec.c:
	  rtpulpfecdec: fix buffer leak when packet is recovered from storage
	  Exposed by rtpulpfecdec_recovered_from_storage test.

2019-03-06 17:35:58 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtpulpfec.c:
	  tests: rtpulpfec: fix buffer leak in unit test
	  This freed wrapped memory instead of the GstMemory or buffer.

2019-03-06 17:33:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: fix caps leak
	  Exposed by rtp_h264depay_bytestream() unit test.

2019-03-06 17:28:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: rtpjitterbuffer: fix leaks in new test_push_eos() test

2019-03-06 17:26:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/meson.build:
	  tests: states: blacklist gtk sinks for state change test
	  gtk_init() throws GLib-GIO-WARNING **: unknown schema extension 'd'
	  unrelated to our test environment.

2019-03-06 17:26:03 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/.gitignore:
	* tests/examples/rtp/.gitignore:
	  tests: .gitignore more test and example binaries

2019-03-05 15:26:45 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gtkgstglwidget.c:
	  gtkgl: Also try retrieving an EGL context from Gdk with X11
	  Some embedded platforms will use EGL instead of GLX within the X11
	  ecosystem.

2019-03-04 09:07:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* meson.build:
	  Back to development

2019-02-25 11:23:56 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix AV1 caps when there's no codec_data
	  There is no "byte-stream" format for AV1 in Matroska, this
	  was probably cargo-culted from H.264. codec_data / CodecPrivate
	  is now mandatory for AV1 in Matroska[*], but there are sample
	  files out there which don't have it (e.g. some Elecard ones).
	  [*] https://github.com/Matroska-Org/matroska-specification/blob/master/codec/av1.md#codecprivate-1

2019-02-28 08:52:28 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/meson.build:
	  meson: don't build icles when tests are disabled
	  They are manual tests, so let them be controlled
	  via the tests option.

2019-02-27 15:39:12 +0100  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: small spell correct

=== release 1.15.2 ===

2019-02-26 11:47:29 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.15.2

2019-02-26 11:47:29 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  Update docs

2019-02-26 11:47:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update translations

2019-02-22 12:22:04 -0300  Mauro Carvalho Chehab <mchehab+samsung@kernel.org>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: accept Bayer as possible input/output for V4L2 codecs
	  A V4L2 transform codec may input/output data on Bayer format.
	  Add support for that.

2019-02-22 12:22:44 -0300  Mauro Carvalho Chehab <mchehab+samsung@kernel.org>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: fix a typo on a debug message at v4l2_calls
	  suppored -> supported

2019-02-25 19:08:08 +1100  Matthew Waters <matthew@centricular.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2dec: also remove the colorimetry and chroma-site fields
	  If a different format is chosen, then these values are incorrect.

2019-02-22 16:02:12 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Fix EOS forwarding
	  So far we assumed that if all sources are bye, this meant we needed to
	  send an EOS on the RTCP sink. The problem is that this case may happens
	  if we only had one internal source and it detected a collision.
	  So now we limit the EOS forwarding to when there is a send_rtp_sink pad
	  and that this pad has received EOS. We don'tcheck the recv_rtp_sink
	  since the code does not wait for the bye to be send before sending EOS
	  to the RTCP src pad.

2019-02-25 01:12:56 +1100  Jan Schmidt <jan@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Declare support for RF64
	  RF64 encode support was added to wavenc quite some time
	  ago, but not declared in wavparse. It seems wavparse can
	  decode it though, so add it to the sink pad.
	  The RF64 support was added in
	  https://bugzilla.gnome.org/show_bug.cgi?id=735627

2019-02-12 18:28:40 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtp: Add property to disable RTCP reports per internal rtpsource
	  This is useful when implementing custom retransmission mechanism like
	  RIST to prevent RTCP from being produces for the retransmitted SSRC.
	  This would also be used in general for various purpose when customizing
	  an RTP base pipeline.

2019-02-12 18:26:21 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Emit on-new-sender-ssrc for RTX ssrc also

2019-01-15 18:04:09 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtp jitterbuffer test: Test for queue filling

2019-01-11 17:53:43 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: Limit size to 2^15 packets
	  If it goes over 2^15 packets, it will think it has rolled over
	  and start dropping all packets. So make sure the seqnum distance is not too big.
	  But let's not limit it to a number that is too small to avoid emptying it
	  needlessly if there is a spurious huge sequence number, let's allow at
	  least 10k packets in any case.

2019-02-11 11:33:32 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: There is no automatic reorder threshold

2019-01-30 10:47:49 -0300  Thibault Saunier <tsaunier@igalia.com>

	* ext/pulse/pulsedeviceprovider.c:
	  pulse: Post DEVICE_CHANGED on modification

2018-11-26 13:48:56 -0300  Thibault Saunier <tsaunier@igalia.com>

	* ext/pulse/pulsedeviceprovider.c:
	* ext/pulse/pulsedeviceprovider.h:
	  pulse: Mark default devices as "default"

2019-02-08 16:10:25 +0000  Ilya Smelykh <ilya.smelykh@gmail.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Use 8kHz sample rate for alaw/mulaw audio

2019-02-07 09:54:31 +0000  Ilya Smelykh <ilya.smelykh@gmail.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: set sample rate to 8KHz for G.711 audio

2019-02-08 13:59:19 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Only write timecode trak for video
	  Recent changes in ccextractor were attaching timecode meta to the closed
	  caption track. We shouldn't write timecode information for the closed
	  caption trak.

2019-02-05 22:14:18 +0100  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* configure.ac:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/meson.build:
	  vpx: Fix build against libvpx 1.8
	  The deprecated debug visualizer was removed.

2019-02-05 19:41:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tests/check/elements/rtph264.c:
	* tests/check/elements/rtph265.c:
	* tests/check/elements/rtpulpfec.c:
	  misc: Fix warnings on Cerbero's mingw (gcc 4.7)
	  error: this decimal constant is unsigned only in ISO C90 [-Werror]

2019-02-06 14:43:18 +0530  Arun Raghavan <arunsr@codeaurora.org>

	* ext/pulse/pulsesink.c:
	  pulsesink: Deal with not being able to convert a format to caps
	  It is possible that PulseAudio adds formats that are not yet supported
	  in pulsesink, and in those cases, we want to gracefully skip them rather
	  than cause an assert on a NULL caps.

2019-01-17 09:22:18 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Remove trailing '\n' in debug

2019-02-05 15:27:49 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtgl.h:
	  qmlgl: Fix opengl header guard changes again
	  Reapply 3d708a5bfa8961cc37671bc3226976dfc9ba50ad in the correct place
	  after the iOS additions.

2019-02-02 02:29:10 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: use the correct segment seqnum

2019-02-02 02:26:47 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: use the correct segment seqnum

2019-02-02 02:24:01 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpsession.h:
	  rtpsession: use the correct segment seqnum

2019-01-26 10:35:31 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Do not error out if the first added and chained pad is not linked
	  And let it the oportunity to get its other pad linked
	  Example:
	  ```
	  $ gst-launch-1.0 uridecodebin uri=file:///home/thiblahute/gst-validate.save/gst-integration-testsuites/testsuites/../medias/defaults/flv/819290236.flv caps=audio/x-raw expose-all-streams=FALSE ! fakesink
	  Setting pipeline to PAUSED ...
	  Pipeline is PREROLLING ...
	  ERROR: from element /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: Internal data stream error.
	  Additional debug info:
	  ../subprojects/gst-plugins-good/gst/flv/gstflvdemux.c(2760): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0:
	  streaming stopped, reason not-linked (-1)
	  ERROR: pipeline doesn't want to preroll.
	  Setting pipeline to NULL ...
	  Freeing pipeline ...
	  ```

2019-01-16 23:54:25 -0800  Christopher Snowhill <kode54@gmail.com>

	* gst/matroska/webm-mux.c:
	  webmmux: allow resolutions above 4096
	  Modify the caps string to allow width and height greater than 4096.
	  There is no need to restrict it since the matroska format allows the
	  width and height values to be up to eight bytes long, and this also
	  applies to the webm subset of the format.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/550

2019-02-01 14:27:11 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  meson: qmlgl plugin iOS definitions
	  Tested with cross-ios-arm64 and cross-ios-x86, since those two are the
	  only archs shipped with the official Qt binaries.

2019-02-01 14:27:11 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/gstqtgl.h:
	  qt: Don't define GLsync inside gstglfuncs.h
	  This was originally added for fixing conflicting definitions between
	  Android and Qt, but times have changed and now this breaks the build
	  on iOS:
	  [...]/OpenGLES.framework/Headers/ES3/gl.h:1006:48: error: unknown type name 'gst_qt_GLsync'
	  GL_API void           GL_APIENTRY glGetSynciv (GLsync sync, GLenum pname, GLsizei bufSize, GLsizei* length, GLint* values)   OPENGLES_DEPRECATED(ios(7.0, 12.0), tvos(9.0, 12.0));
	  ^
	  ../ext/qt/gstqtgl.h:49:16: note: expanded from macro 'GLsync'
	  #define GLsync gst_qt_GLsync
	  ^
	  6 errors generated.
	  Instead, we simply avoid defining GLsync ourselves if we're using Qt.

2019-02-01 14:27:11 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  meson: Fix indentation in qt plugin and add a FIXME comment

2019-01-26 21:02:27 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay; Fix handling of marker on aggregated packet
	  When multiple nals are aggrgated, the marker bit should be associated only
	  with the last NAL of the packet. Otherwise we may break rendering in with
	  AU alignment.

2019-01-26 21:01:08 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Fix handling or marker on STAP-A
	  Only forward the marker for the last NAL of the STAP-A. Otherwise each NAL
	  endup being assumed to be a full frame which may break rendering.
	  Fixes 557

2019-01-27 09:19:00 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtph265.c:
	  test: h265depay: Add todo for testing aggregate packets with marker
	  We are missing a sample to test this, but a fix has been made, so add a
	  todo.

2019-01-26 20:42:40 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtph264.c:
	  test: rtph264depay: Check handling of STAP-A marker
	  Related to #557

2019-01-31 15:23:43 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tests/check/meson.build:
	  meson: orc-test is not required
	  This is especially never available on iOS.

2019-01-30 19:44:01 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* meson.build:
	* tests/check/meson.build:
	  meson: Add support orc fallback
	  Allow fallback to orc subproject if any.
	  Additionally 'dependencies' keyword is removed from find_library,
	  because it's invalid keyword for find_library.

2019-01-17 21:06:54 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	  gdkpixbufdec: always output a TIME segment
	  It makes no sense for a decoder to output a BYTES segment, and
	  many elements one would plug downstream of a video decoder assume
	  the segments they receive are in TIME format, for example this fixes:
	  gst-validate-1.0 filesrc location=opacity01.svg ! gdkpixbufdec ! \
	  videobalance ! videoconvert ! fakesink
	  In that case, videobalance was emitting an assertion when trying to
	  call gst_object_sync_values()

2019-01-29 12:12:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  meson: Add macOS definitions for qmlgl plugin
	  Tested with Cerbero.

2019-01-29 12:12:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  meson: Fix building of qmlgl plugin on Android
	  Needs gnustl for C++ STL support, which is the GNU STL on Android API
	  19 and older, and is a wrapper for the llvm-libc++ STL on newer APIs.
	  QtGui C++ templates use GL functions, so GLESv2 is needed at link time

2019-01-24 16:21:12 -0500  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/interleave/deinterleave.c:
	  deinterleave: Allow switching between 1 channel configs
	  regardless of whether they're positioned, since positioning
	  with a 1 channel stream doesn't change anything.

2019-01-22 11:45:49 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* configure.ac:
	  configure.ac: Fix Qt Android integration
	  The Qt Android integration is now signalled with HAVE_QT_ANDROID
	  See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/86

2018-12-18 14:46:25 -0500  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/soup/meson.build:
	  Meson: fallback to libsoup subproject

2019-01-22 12:52:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: detect opengl api from -base .pc files correctly
	  There was a mismatch between the .pc files generated by
	  autotools and by meson that would lead to meson not detecting
	  that opengl api is available even though it is, if -base was
	  built with autotools. The mismatch has now been rectified in
	  -base, so we need to update for that.
	  This is mostly for consistency, this problem didn't seem
	  to affect anything in -good.
	  See https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/871

2019-01-22 09:51:33 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/ximage/meson.build:
	  meson: ximage: check for XShmAttach()
	  Fixes FIXME.

2019-01-22 09:32:31 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson_options.txt:
	* sys/ximage/meson.build:
	  meson: add options for ximagesrc xshm, xfixes, xdamage checks
	  And rename x11 option to ximagesrc.
	  Fixes #553

2019-01-21 11:53:53 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/qt/README.md:
	  qmlgl: add README.md with information on building for non-linux platforms with qmake

2019-01-19 15:46:41 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/qt/meson.build:
	  qmlgl: meson: fix theoretical support for building for android
	  The android code path is slightly different than the EGLFS one,
	  so I added previously a HAVE_QT_ANDROID define for use with qmake.
	  Here I also add it in meson, although I expect nobody will ever use
	  meson to build this, as it's complicated.

2019-01-19 15:37:45 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/qt/qtplugin.pro:
	  qmlgl: qmake: add support for MacOS target

2019-01-19 15:21:43 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/qt/qtplugin.pro:
	  qmlgl: qmake: remove cerbero's include dir from the include path
	  pkg-config should do it's job here, this is unnecessary and implies using cerbero

2019-01-19 15:19:26 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/qt/gstqtgl.h:
	* ext/qt/qtplugin.pro:
	  qmlgl: qualify Qt includes with their module and remove module include dir from the .pro file
	  it is perfectly legal to use the <module/class> style of includes with Qt
	  and it avoids the need for having the module's include dir in the include path

2019-01-19 15:10:09 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/qt/qtplugin.pro:
	  qmlgl: qmake: don't link against QtWidgets, it's not used

2019-01-19 15:07:44 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/qt/gstqtglutility.cc:
	* ext/qt/qtplugin.pro:
	  qmlgl: qmake: fix building for android

2019-01-19 02:39:32 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  meson: Generate pkg-config file for qmlgl plugin

2019-01-17 16:26:56 +0100  Victor Toso <me@victortoso.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading avoid -Wmaybe-uninitialized
	  More false positives as both of them are initialized in the line
	  before they are used, wrapped with fail_unless() check.

2019-01-17 16:19:40 +0100  Victor Toso <me@victortoso.com>

	* tests/check/elements/matroskamux.c:
	  tests: matroskamux avoid -Wmaybe-uninitialized
	  False positive for the three variables but some warnings like:
	  ../tests/check/elements/matroskamux.c:875:10:
	  warning: 'chapters_offset' may be used uninitialized in this function [-Wmaybe-uninitialized]
	  *index = chapters_offset;
	  ~~~~~~~^~~~~~~~~~~~~~~~~
	  The above is false positive as there is a gboolean to check if it was
	  initialized or not (found_chapters_declaration).

2018-05-28 14:39:53 +0530  Arun Raghavan <arunsr@codeaurora.org>

	* ext/pulse/pulseutil.c:
	  pulse: Fix format info to caps conversion for PCM

2019-01-18 12:27:34 +0530  Arun Raghavan <arunsr@codeaurora.org>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  Revert "pulsesrc: Move to extended stream API"
	  This reverts commit 4d67d1bd16bcf25acf89d8acd952badcd5b9a657.
	  Using the extended API for the capture path depends on a fix in
	  PulseAudio
	  (https://gitlab.freedesktop.org/pulseaudio/pulseaudio/merge_requests/49).
	  Until then, let's go back to the standard API.
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/552

2019-01-18 14:41:14 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	  meson: Search for qmake-qt5 before qmake
	  The canonical name for the binary is qmake-qt5, and qmake is the
	  generic name that can also be a qt4 qmake.

2019-01-17 15:30:25 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2.c:
	  v4l2: mark caps from probe as MAY_BE_LEAKED

2019-01-15 18:06:11 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: fix cdata caps leaks
	  The cdata structure was freed but not its caps.
	  It was already done in gst_v4l2_video_dec_subclass_init() and
	  gst_v4l2_video_enc_subclass_init().

=== release 1.15.1 ===

2019-01-17 01:59:28 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.15.1

2019-01-17 01:59:28 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  Update docs

2019-01-17 01:59:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update translations

2019-01-16 14:11:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk/gl: Only unbind buffers/vertex attrib arrays if we can't directly bind the vertex array to 0
	  Binding the vertex array to 0 will unbind everything else already.
	  In the previous order older versions of the Intel GL driver caused
	  errors to be printed for every single call when disabling the vertex
	  attrib arrays after binding the vertex array to 0.

2019-01-16 00:57:46 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/meson.build:
	  meson: enable tests for orc code

2018-11-29 16:07:08 +0100  Patrick Radizi <patrickr@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: send GstRTSPSrcTimeout message on timeout
	  The GstRTSPSrcTimeout message is sent by the rtspsrc when it receives
	  the on-timeout signal from rtpsession. This can be used by an
	  application for error handling.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/499

2019-01-09 17:52:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Handle the encoder metadata the same as metadatacreator
	  And store it in our ENCODER tag.

2019-01-09 17:48:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Add encoder metadata to the header
	  And also add a property for setting this. By default it has the same
	  value as the metadatacreator metadata.
	  Various software is using encoder instead of metadatacreator, others are
	  using them both for different purposes. As such it's useful to have
	  support for setting both here.

2018-05-28 14:41:05 +0530  Arun Raghavan <arunsr@codeaurora.org>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulsesrc: Move to extended stream API
	  This is needed as a precursor to allowing capture of IEC61937
	  formats. We now also need to include the channel map while converting
	  format info to caps so that a correct channel mask is generated for
	  pulsesrc's caps.

2019-01-09 16:27:16 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* tests/check/elements/rtph265.c:
	  test: rtph265pay: Verify we only mark the last fragment

2019-01-09 16:24:54 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* tests/check/elements/rtph265.c:
	  test: rtph265pay: Use a bigger test frame
	  The existing frame's last slice is too small to be used for
	  fragmentation tests.

2019-01-09 15:59:16 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* tests/check/elements/rtph264.c:
	  test: rtph264pay: Verify we only mark the last fragment

2019-01-09 16:25:36 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Only mark the last fragment of an AU
	  Commit e721071dcac9f231e5e10b4bb31323658a6cdd1a removed the check for
	  the end of fragmentation. As a result, all fragments of an AU's last
	  NALU were marked.

2019-01-09 15:56:51 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Only mark the last fragment of an AU
	  Commit 4add820cce278213ede3d5fce427ea92e0619b6f removed the check for
	  the end of fragmentation. As a result, all fragments of an AU's last
	  NALU were marked.
	  Potential fix for https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/540

2019-01-09 11:48:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Refactor part preparation code and remove "prepared" signal from reader helper object
	  We don't need a special signal anymore but can directly work with
	  async-done

2019-01-09 11:42:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multifile/gstsplitmuxsrc.h:
	  splitmuxsrc: Implement state change asynchronously instead of blocking
	  Blocking in change_state() is a recipe for disaster, even more so if
	  we wait for another thread that also calls into various element API and
	  could then lead to deadlocks on e.g. the state lock.

2019-01-05 23:10:46 +0400  Marc-André Lureau <marcandre.lureau@redhat.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: fix checking for invalid stream index
	  PA_INVALID_INDEX, the default value, is unfortunately !0.
	  Setting the volume before the stream is created will put the ring
	  buffer in error state. Unfortunately, that's what spice-gtk does.

2018-12-20 12:14:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Split CEA608 buffers correctly so that each output buffer represents a single frame

2018-12-20 11:45:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Refactor buffer pushing into its own function

2018-12-20 11:31:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Extract CEA608 framerate from the (first) video stream
	  EA608 closed caption tracks are a bit special in that each sample
	  can contain CCs for multiple frames, and CCs can be omitted and have to
	  be inferred from the duration of the sample then.
	  As such we take the framerate from the (first) video track here for
	  CEA608 as there must be one CC byte pair for every video frame
	  according to the spec.
	  For CEA708 all is fine and there is one sample per frame.

2018-12-30 21:55:28 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* tests/check/meson.build:
	  tests: Enable more unit tests on Windows

2018-12-30 21:54:44 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* tests/check/elements/audioamplify.c:
	* tests/check/elements/audiodynamic.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/audiopanorama.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/avisubtitle.c:
	* tests/check/elements/capssetter.c:
	* tests/check/elements/level.c:
	* tests/check/elements/matroskamux.c:
	* tests/check/elements/multifile.c:
	* tests/check/elements/qtdemux.h:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/shapewipe.c:
	* tests/check/elements/spectrum.c:
	* tests/check/elements/splitmux.c:
	* tests/check/elements/udpsrc.c:
	* tests/check/elements/videobox.c:
	* tests/check/elements/videocrop.c:
	* tests/check/elements/videofilter.c:
	* tests/check/elements/videomixer.c:
	* tests/check/elements/wavpackparse.c:
	* tests/check/elements/y4menc.c:
	* tests/check/generic/states.c:
	  tests: Remove pointless unistd.h include

2018-12-26 20:27:58 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't leak allocated index memory
	  Don't forget to free returned memory from _search_pos()

2018-12-25 15:31:44 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/files/Makefile.am:
	  tests: dist new rtph265.rtp file
	  Fixes make distcheck.

2018-12-25 14:51:38 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofx.c:
	* gst/audiofx/gststereo.c:
	* gst/audiofx/meson.build:
	  audiofx: add stereo element which was moved from -bad to build
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/457

2018-12-25 13:07:23 +0100  Tim-Philipp Müller <tim@centricular.com>

	  Move stereo plugin from -bad
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/457

2018-12-22 17:55:51 +0100  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Offset correction for track language code parsing
	  The duration field being a uint64, is stored in 8 bytes, not 4. So the offset of
	  the following field, language code, needs to be updated accordingly so that the
	  parsed language code is not garbage.

2018-12-21 10:59:22 +0100  Juan Navarro <juan.navarro@gmx.es>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Accept NULL for "port-range" property
	  The documentation of "port-range" implies that passing NULL should be
	  valid, but currently it is not. Without this check, the sscanf() call
	  will crash.

2018-12-19 14:28:54 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.signals:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/rtpbundle.c:
	* tests/check/meson.build:
	* tests/examples/rtp/.gitignore:
	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-rtpbundle.c:
	* tests/examples/rtp/meson.build:
	* tests/examples/rtp/server-rtpbundle.c:
	  Revert "rtpbin: receive bundle support"
	  This reverts commit dcd3ce9751cdef0b5ab1fa118355f92bdfe82cb3.
	  This functionality was implemented for gstopenwebrtc, but it
	  turned out this was not actually needed for webrtc bundling
	  support, as shown in webrtcbin. It also doesn't correspond
	  to any standards.
	  This is an API break, but nothing should actually depend on
	  this, at least not for its initial purpose.
	  Changes in rtpbin.c were reverted manually, to preserve some
	  refactoring that had occurred in the original commit.
	  Fixes #537

2018-12-19 11:36:37 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	  rtph264pay/rtph265pay: Fix use after free
	  We can't assume a buffer that has been pushed in the adapter is still
	  valid. This fixes a use after free detect when running test on jenkins.

2018-12-19 22:51:11 +0900  KimTaeSoo <myrandy1@gmail.com>

	* tests/check/pipelines/tagschecking.c:
	  tagschecking: Use gst_message_parse_warning in case of GST_MESSAGE_WARNING
	  Bus message handler of tags checking unit test uses gst_message_parse_error()
	  in case of GST_MESSAGE_ERROR and GST_MESAGE_WARNING.
	  If gst_message_parse_error() is called in case of GST_MESSAGE_WARNING, assert occurs.
	  So modified to use gst_message_parse_warning() in case of GST_MESSAGE_WARNING.

2018-12-19 09:51:10 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/Makefile.am:
	  test: rtph264/265: Add libgstrtp in auto-tool makefile

2018-12-18 12:43:30 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/Makefile.am:
	* tests/check/elements/rtph265.c:
	* tests/check/meson.build:
	* tests/files/h265.rtp:
	  test: rtph265: Copy and port tests from rtph264
	  This copy and port all the relevant tests from rtph264.

2018-12-14 17:54:36 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtph264.c:
	  test: rtph264depay: Check the marker is converted to flag

2018-12-14 17:53:17 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtph264.c:
	  test: rtph264depay: Check that EOS drains the depayloaded
	  In AU mode, the depayloader may have accumulated NALs, test that
	  these NALs are drained and not dropped.

2018-12-14 15:30:21 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtph264.c:
	  test: rtph264pay: Add tests for marker bit
	  Test that marker bit is transferred when input buffer has the
	  marker flag set but also that it's set whenever the payloader
	  receives complete AU.

2018-12-13 15:57:24 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtph264.c:
	  test: rtph264pay: Verify slices timestamp
	  This test make sure that timestamps are properly transfered
	  to each NALU.

2018-12-04 16:06:15 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/rtph264.c:
	* tests/check/meson.build:
	  test: rtph264pay: Add reserved nals test

2018-12-18 13:16:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Don't wait for next nal when input is aligned
	  This is the same as what was done on rtph264pay in the patch
	  d5d28055c1e816e90e8c2d1151816b0c3e760ff3

2018-12-18 12:53:15 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: Drain on EOS event

2018-12-18 12:50:40 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: Factor out the code that push
	  This will be needed to implement draining on EOS.

2018-12-17 16:48:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Drain on EOS event

2018-12-14 18:19:42 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Factor out the code that push
	  This will be needed to implement draining on EOS.

2018-12-14 15:51:51 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	  rtph26xpay: Remove unused IS_ACCESS_UNIT macro
	  This macro is not longer used. It was secretly checking if that nal was
	  a slice, and confusingly name to that one may think it was checking if
	  the nal is an AUD.

2018-10-03 14:14:17 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Fix reading timestamps from adapter
	  The code was reading the timestamp from the adapter before pushing the
	  new buffer into it. As a side effect, if the adapter was empty, we'd end
	  up using an older timestamp. In alignment=au, it means that all
	  timestamp was likely one frame in the past, while in alignment=nal, with
	  multiple slices per frame, the first slice would have the timestamp of
	  the previous one.

2018-10-03 13:46:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265pay: Forward the marker bit as buffer flag
	  We have a buffer flag to represent the marker bit (when present).
	  Forward this bit by setting the buffer flag accordingly.

2018-10-03 13:44:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Properly set the marker bit
	  The marker bit is used for efficient decoding. The assumption that
	  it should be set on the AUD is wrong, since the AUD is conceptually
	  starts the frame, while the marker is to indicate the end.
	  So properly set the marker bit as soon as we know we are ending an
	  AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
	  flag.

2018-09-25 11:49:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Fix reading timestamps from adapter
	  The code was reading the timestamp from the adapter before pushing the
	  new buffer into it. As a side effect, if the adapter was empty, we'd end
	  up using an older timestamp. In alignment=au, it means that all
	  timestamp was likely one frame in the past, while in alignment=nal, with
	  multiple slices per frame, the first slice would have the timestamp of
	  the previous one.

2018-09-24 15:31:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Properly set the marker bit
	  The marker bit is used for efficient decoding. The assumption that
	  it should be set on the AUD is wrong, since the AUD is conceptually
	  starts the frame, while the marker is to indicate the end.
	  So properly set the marker bit as soon as we know we are ending an
	  AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
	  flag.

2018-09-24 15:27:41 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Forward the marker bit as buffer flag
	  We have a buffer flag to represent the marker bit (when present).
	  Forward this bit by setting the buffer flag accordingly.

2018-09-21 20:22:43 +0000  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Protect against use of reserved NAL types
	  Don't allow external encoder to use one of the reserved NAL type
	  implicated in NAL aggreation. These out-of-spec NAL types, if passed
	  from the outside world will lead to an invalid RTP payload being
	  created.

2018-12-07 21:46:12 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* meson.build:
	* tests/check/meson.build:
	* tests/meson.build:
	  tests: Enable unit test on Windows
	  Allow run some unit tests on Windows.
	  * Remove hardcoded path separator in whitelist env for Meson to choose
	  OS-specific separator automatically (i.e., ';' for windows and ':' for *nix)
	  * Add dependency explicitly for some test cases, otherwise plugins couldn't be
	  loaded on uninstalled environment of Windows.

2018-12-18 20:39:40 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* meson.build:
	* tests/check/meson.build:
	  meson: Prefer to use join_paths() over '/'
	  ... to avoid mixing '/' and '\' in a path string on Windows.

2018-12-17 18:04:37 +0000  Jonny Lamb <jonnylamb@jonnylamb.com>

	* tests/check/elements/rtpulpfec.c:
	  rtpulpfec: stop and start the harness when setting error-after
	  gstreamer!55 makes some changes to how the `error-after` counter works
	  which breaks this test. This change makes the test not rely on the
	  ability to alter `error-after` at runtime and explicitly stops and
	  starts the harness before pushing data.
	  An alternative would be to add another argument to
	  `harness_rtpulpfecdec` to set `error-after` on construction but that's
	  slightly more long-winded. so I went for this approach instead.
	  Fixes #532, even though that's already closed.

2018-12-17 18:59:34 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/rtpaux.c:
	* tests/check/meson.build:
	  tests: remove rtpaux test
	  The initial mission statement for this test was:
	  * demonstrate usage of the request-aux-* signals in rtpbin
	  * test the rtx elements
	  We have examples that serve the first use case, and better
	  (harnessed) tests for the second use case.
	  This test is slow and racy, it served its purpose but can now
	  be removed.
	  Fixes #533

2018-12-17 19:18:43 +0100  Nicola Murino <nicola.murino@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: check difference in time from the last socket read before changing blocksize
	  If the pipeline consumes the data slower than the available network speed,
	  for example because sync=true, is useless to increase the blocksize and
	  reading in too big blocksizes can cause the connection to time out
	  Closes #463

2018-08-08 09:27:09 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Avoid code duplication
	  The function gst_v4l2_object_add_interlace_mode() has repeating code so
	  it's best use a loop instead. That will make it easy and simple to add
	  additional interlace modes in a following patch.

2018-06-27 23:20:33 +0200  Zeeshan Ali <zeenix@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Make use of gst_video_interlace_mode_to_string()
	  Instead of a custom map to translate the interlace modes to strings, let's
	  make use of the base API provided.

2018-12-17 13:45:36 +0100  Nicola Murino <nicola.murino@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxcoreaudio: fix typo
	  kAudioFormatFlagIsSignedInteger is a format flags
	  Closes #394

2018-12-17 09:33:39 +0100  Edward Hervey <edward@centricular.com>

	* ext/qt/gstqtgl.h:
	  qtgl: Handle OPENGL header guard changes
	  In 2018 khronos changed the gl header guards. If we don't detect
	  this properly we would end up with plenty of symbol redifinition
	  (since we would be importing twice the "same" header).
	  Instead detect if the "newer" header was already included and if
	  so define the "old" define to avoid this situation
	  Fixes #523

2018-12-10 17:34:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/qtdemux.c:
	  isomp4: Replace GST_VIDEO_CAPTION_TYPE_CEA608_IN_CEA708_RAW with CEA608_S334_1A
	  For the demuxer we have to select line offset 0 for the time being as
	  this information is not passed over MOV.

2018-12-13 20:45:23 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer tests: Validate the number of buffers

2018-12-13 19:17:43 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Run all timers immediately on EOS
	  When the EOS event is received, run all timers immediately and avoid
	  pushing the EOS downstream before this has been run. This ensures that
	  the lost packet statistics are accurate.

2018-12-13 19:16:11 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer test: Stop jitterbuffer before pads to avoid race
	  The teardown of the pads checks the refcount, but there are timers
	  inside the jitterbuffer that can push things, so if we're not lucky,
	  things could be pushed while the pads are being shut down. Putting the
	  jitterbuffer to NULL first avoids this.

2018-11-22 10:41:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Stop waiting after EOS
	  After EOS is received, it is pointless to wait for further events,
	  specially waiting on timers. This patches fixes two cases where we could
	  wait instead of returning GST_FLOW_EOS and trigger a spin of the loop
	  function when EOS is queued, regardless if this EOS is the queue head or
	  not.

2018-10-27 13:41:46 +0200  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Use new channel count for audio info

2018-10-27 13:36:16 +0200  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Caps may have changed on FLAC metadata change
	  If the decoder signals metadata change we need to update the output
	  format and negotiate with downstream elements.

2018-10-27 13:28:56 +0200  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Reset decoder on set_format()
	  Any call to set_format() could mean that the stream type changed so we
	  reset the decoder and mark got_headers FALSE.

2018-12-05 18:42:55 +0100  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: On sink caps change restart parser
	  Draining the parser is not enough here, on caps change we need to
	  reset it so it is ready to accept new caps.

2018-12-04 18:50:51 +0100  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: Update pad caps if inline caps change
	  If the inlined caps change while using the same CV we need to update the
	  source pad caps.

2018-12-14 12:21:58 +0900  Justin Kim <jeongseok.kim@sk.com>

	* sys/osxvideo/meson.build:
	  osxvideo: meson: Add dependencies by using appleframeworks
	  Otherwise, it fails to link.
	  gst-build#13

2018-12-07 19:09:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/cairo/gstcairooverlay.c:
	  cairooverlay: Optimize premultiplication/unpremultiplication loops
	  Pull in video frame fields into local variables. Without this the
	  compiler must assume that they could've changed on every use and read
	  them from memory again.
	  This reduces the inner loop from 6 memory reads per pixels to 4, and the
	  number of writes stays at 3.

2018-12-05 19:37:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Put framerate into the closedcaption caps if it can be calculated from the stream
	  Using the same calculation used for video streams.

2018-12-05 19:31:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Set timescale of closedcaption tracks to the one of the main video track

2018-12-05 17:24:13 -0300  Thibault Saunier <tsaunier@igalia.com>

	* common:
	  Automatic update of common submodule
	  From ed78bee to 59cb678

2018-11-19 18:20:52 +0000  Maciej Wolny <maciej.wolny@codethink.co.uk>

	* gst/flv/gstflvmux.h:
	* sys/v4l2/gstv4l2allocator.h:
	  Remove duplicate declarations
	  This causes 'redefinition of typedef ...' errors on GCC 4.5.3

2018-11-30 23:56:12 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	  tests: rtpssrcdemux: fix uninstalled autotools build and distcheck

2018-11-30 19:29:30 +0100  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: set need_segment after a second moov
	  stream.segment should be updated with the values of the current edit
	  list, also when a new `moov` is received. Unfortunately this was not
	  being the case because of an early return.
	  As a consequence of this bugs, no end of movie clipping was being
	  performed on the new moov and no segment event was being emitted.
	  When performing stream switching (e.g. in MSE) the new moov may have a
	  different edit list. This is often the case when switching between
	  baseline H.264 (which lacks B-frames) and more demanding profiles. For
	  this reason it's important to emit a new segment in order to be able
	  to get matching stream times.

2018-11-29 22:42:34 +0100  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Initialize QtDemuxStream.segment in its constructor
	  This patch moves the initialization of QtDemuxStream.segment from
	  gst_qtdemux_add_stream() to _create_stream(). This ensures the segment
	  is always initialized when the stream is created.
	  Otherwise the segment format is left as GST_FORMAT_UNDEFINED in the case
	  were a track is reparsed and qtdemux_reuse_and_configure_stream() is
	  called instead of gst_qtdemux_add_stream(). (See
	  qtdemux_expose_streams() in the non streams-aware case.)

2018-11-29 13:48:33 +0100  Miguel Paris <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: properly handle rtcp_feedback_retention_window
	  - Consider GST_CLOCK_TIME_NONE as not to be used.
	  - Complete "rtcp-feedback-retention-window" property getter/setter
	  implementation.

2018-11-29 13:02:53 +0100  Miguel Paris <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsource: properly prune RTCP packets out of feedback_retention_window
	  Closes #522

2018-11-29 13:01:44 +0100  Miguel Paris <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: properly compare buffer PTSs

2018-11-29 12:58:18 +0100  Miguel Paris <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: retain_rtcp_packet: warning if invalid running_time

2018-11-29 12:55:38 +0100  Miguel Paris <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: properly set the running_time for rtcp packet info

2018-11-29 14:54:06 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Rename confusingly name lock macros
	  This is an extra internal recurisve lock use to avoid having to take
	  both sink pad streams lock all the time. This patch renamed it
	  INTERLNAL_STREAM_LOCK/UNLOCK() to avoid confusion with possible upstream
	  GST_PAD API.

2018-11-28 17:14:11 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Hold on internal stream lock while pushing sticky
	  This reverts "6f3734c305 rtpssrcdemux: Only forward stick events while
	  holding the sinkpad stream lock" and actually hold on the internal
	  stream lock. This prevents in some needed case having a second
	  streaming thread poping in and messing up event ordering.

2018-11-27 17:10:57 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/Makefile.am:
	* tests/check/elements/rtpssrcdemux.c:
	* tests/check/meson.build:
	  test: rtpssrcdemux: Test event forwarding
	  This the first unit test of this element. It adds a test that verify
	  that events are forwarded correctly.

2015-11-04 12:52:17 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix handling of MS ACM audio
	  Pass riff codec-data as strf, not strd, which is where
	  gst_riff_create_audio_caps() expects the WAVEFORMATEXTENSIBLE
	  data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757583
	  Fixes #234

2018-11-28 05:52:16 +0200  Jordan Petridis <jordan@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/udp/gstudpsrc.c:
	* sys/v4l2/gstv4l2jpegenc.c:
	* tests/check/elements/rtpmux.c:
	* tests/check/elements/rtpsession.c:
	  Run gst-indent through the files
	  This is required before we enabled an indent test in the CI.
	  https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33

2018-11-26 08:10:24 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst/videocrop/gstaspectratiocrop.c:
	  aspectcropration: Fix potential unref of NULL pointer

2018-11-25 11:31:11 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstaspectratiocrop.h:
	  aspectcropratio: Set caps from the streaming thread on property changes
	  Otherwise it might lead to deadlocks
	  See https://gitlab.gnome.org/GNOME/pitivi/issues/2259
	  Closes #518

2018-11-23 14:01:35 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Forward serialized events to all pads
	  While forwarding serialized event, we use gst_pad_forward() function.
	  In the forward callback (GstPadForwardFunction) we always return
	  TRUE. Returning true there will stop the dispatching procedure. As a
	  side effect, only one events is receiving the events. This breaks
	  when sending EOS from the applicaiton, it also breaks the latency
	  tracer.

2018-11-24 19:13:28 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* meson.build:
	  meson: Specify encoding to UTF-8 when building with MSVC
	  Use build arguments consistent with core and -base. This can also
	  remove noisy "C4819" warning of non-us locale MSVC.

2018-11-21 15:11:00 -0500  Xavier Claessens <xavier.claessens@collabora.com>

	* meson.build:
	  Check for zlib header

2018-11-21 18:53:39 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	  v4l2: Properly fix Android build
	  The previous patch did not even compile on any possible platform or C
	  standard. That commit also didn't have a proper commit message.
	  Android ships Linux with a different signature for ioctl. They first
	  released an ioctl with int as request type, and later "fixed" it by
	  adding an override with unsign, which is still not matching Linux and
	  BSD implementation which uses unsigned long int.

2018-11-21 16:11:02 -0500  Xavier Claessens <xavier.claessens@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	  Fix ioctl() signature on Android

2018-10-09 16:43:08 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* meson.build:
	  Fix zlib detection when there is no pkg-config file

2018-11-19 20:05:39 +0530  Arun Raghavan <arunsr@codeaurora.org>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulse: Expose the correct max rate that we support
	  PulseAudio defines PA_RATE_MAX as the maximum sampling rate that it
	  supports. We were previously exposing a maximum rate of INT_MAX, which
	  is incorrect, but worked because nothing was really using a rate greater
	  than 384000 kHz.
	  While playing DSD data, we hit a case where there might be very high
	  sample rates (>1MHz), and pulsesink fails during stream creation with
	  such streams because it erroneously advertises that it supports such
	  rates.
	  Since PA_RATE_MAX is #define'd to (8*48000U), we can't just use it in
	  the caps string. Instead, we fix up the rate to what we actually support
	  whenever we use our macro caps.

2018-11-14 08:57:55 +0100  Alicia Boya García <aboya@igalia.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: Defer seeks received before GST_MATROSKA_READ_STATE_DATA
	  This patch enables matroskademux to receive seeks before it reaches
	  GST_MATROSKA_READ_STATE_DATA.
	  Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/514
	  This also enables receiving seeks in the element READY state.
	  When such a seek is received, it is stored to be later handled when
	  GST_MATROSKA_READ_STATE_DATA is reached.

2018-10-16 12:38:46 +0200  Linus Svensson <linussn@axis.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	  rtpsession: Implement reset
	  Reset RTPSession when rtpsession changes state from PAUSED to READY.
	  Without this change, a stored last_rtptime in RTPSource could interfere
	  with RTP timestamp generation in RTCP Sender Report.
	  Fixes #510

2018-11-06 15:05:54 +0100  Linus Svensson <linussn@axis.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: test: Plug memory leak

2018-11-13 00:37:11 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	* gst/rtpmanager/gstrtpfunnel.h:
	  rtpfunnel: Stop using G_DECLARE_FINAL_TYPE
	  Fixes #516

2018-11-12 13:42:29 +0200  Jordan Petridis <jordan@centricular.com>

	* .gitlab-ci.yml:
	  Add Gitlab CI configuration
	  This commit adds a .gitlab-ci.yml file, which uses a feature
	  to fetch the config from a centralized repository. The intent is
	  to have all the gstreamer modules use the same configuration.
	  The configuration is currently hosted at the gst-ci repository
	  under the gitlab/ci_template.yml path.
	  Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29

2018-10-18 22:23:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Read driver selected interlace mode
	  If there was no interlace-mode field in the caps. Read back the value
	  selected by the driver. This way, if the driver does not support
	  progressive, then it will automatically negotiate the returned mode
	  unless this mode is not supported by GStreamer.
	  This method was already used for colorimetry. Just like colorimetry, the
	  interlace mode is not longer probed by v4l2src dues to performance
	  issues.
	  Fixes #511

2018-05-17 21:58:25 +1000  Matthew Waters <matthew@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska: implement preliminary support for the bitrate query
	  Return the size / total duration as a ballpark estimate.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/60

2018-05-17 21:53:56 +1000  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: add preliminary support for the bitrate query
	  Return the upstream size over the duration as a first estimate.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/60

2018-11-06 23:02:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Sink jitterbuffer/storage before passing as parameters to signals
	  Otherwise signal handlers from bindings will take ownership of them as
	  they are still floating, and we won't own a reference inside rtpbin
	  anymore.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/515

2018-10-27 18:00:52 +0100  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/flvmux.c:
	  flvmux: Test that timestamps are always increasing
	  Decreasing timestamps break rtmpsink.
	  With contributions from Olivier Crête.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796382

2018-10-27 19:27:12 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Force timestamps to always be increasing
	  https://bugzilla.gnome.org/show_bug.cgi?id=796382

2018-11-05 05:36:26 +0000  Matthew Waters <matthew@centricular.com>

	* .gitmodules:
	  Update common submodule location
	  Remove the git directory

2018-11-05 12:16:46 +0800  Haihao Xiang <haihao.xiang@intel.com>

	* .gitmodules:
	* gst-plugins-good.doap:
	  Clone the code from gitlab
	  This fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/513

2018-11-01 20:37:12 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Ignore corrupted CTTS box
	  If ctts (CompositionOffsetBox) has larger sample_offset
	  (offset between PTS and DTS) than (2 * duration) of the stream,
	  assume the ctts box to be corrupted and ignore the box.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797262

2018-10-23 09:45:36 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Implement SEGMENT query
	  https://bugzilla.gnome.org/show_bug.cgi?id=797313

2018-10-23 09:42:21 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Implement SEGMENT query
	  https://bugzilla.gnome.org/show_bug.cgi?id=797313

2018-10-28 17:12:59 +0000  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfsrc.c:
	  dtmfsrc: Declare output as interleaved
	  This element doesn't support planar audio yet.

2018-10-28 14:09:21 +0000  Nirbheek Chauhan <nirbheek@centricular.com>

	* tests/icles/meson.build:
	  meson: Add some missing test dependencies
	  Without these dependencies, the enumtype may not be generated when the
	  test is built, which will cause a compile failure.

2018-10-28 14:07:54 +0000  Nirbheek Chauhan <nirbheek@centricular.com>

	* tests/check/meson.build:
	  meson: Cleanup old FIXMEs that relied on meson changes

2018-10-16 17:28:00 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Allow changing the SDES at runtime
	  Make it possible to modify the SDES in a packet at runtime.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763502

2018-03-01 17:25:07 +0100  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/gstqtmux.c:
	  qtmux: round to nearest when computing mehd and tkhd duration
	  This fixes a bug where in some files mehd.fragment_duration is one unit
	  less than the actual duration of the fragmented movie, as explained below:
	  mehd.fragment_duration is computed by scaling the end timestamp of
	  the last frame of the movie in (in nanoseconds) by the movie timescale.
	  In some situations, the end timestamp is innacurate due to lossy conversion to
	  fixed point required by GstBuffer upstream.
	  Take for instance a movie with 3 frames at exactly 3 fps.
	  $ gst-launch-1.0 -v videotestsrc num-buffers=3 \
	  ! video/x-raw, framerate="(fraction)3/1" \
	  ! x264enc \
	  ! fakesink silent=false
	  dts: 999:59:59.333333334,  pts: 1000:00:00.000000000, duration: 0:00:00.333333333
	  dts: 999:59:59.666666667,  pts: 1000:00:00.666666666, duration: 0:00:00.333333334
	  dts: 1000:00:00.000000000, pts: 1000:00:00.333333333, duration: 0:00:00.333333333
	  The end timestamp is calculated by qtmux in this way:
	  end timestamp = last frame DTS + last frame DUR - first frame DTS =
	  = 1000:00:00.000000000 + 0:00:00.333333333 - 999:59:59.333333334 =
	  = 0:00:00.999999999
	  qtmux needs to round this timestamp to the declared movie timescale, which can
	  ameliorate this distortion, but it's important that round-neareast is used;
	  otherwise it would backfire badly.
	  Take for example a movie with a timescale of 30 units/s.
	  0.999999999 s * 30 units/s = 29.999999970 units
	  A round-floor (as it was done before this patch) would set fragment_duration to
	  29 units, amplifying the original distorsion from 1 nanosecond up to 33
	  milliseconds less than the correct value. The greatest distortion would occur
	  in the case where timescale = framerate, where an entire frame duration would
	  be subtracted.
	  Also, rounding is added to tkhd duration computation too, which
	  potentially has the same problem.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793959

2018-05-16 14:15:13 +0200  Marc Leeman <marc.leeman@gmail.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: print information about bind_error socket error
	  In some cases, a bind error occurs during operation. Printing
	  the information about the problem is critical for finding the
	  conflict
	  https://bugzilla.gnome.org/show_bug.cgi?id=797340

2018-10-17 12:58:08 +0200  Johan Bjäreholt <johanbj@axis.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-read-common.c:
	  matroska-demux: Fix caps memleak
	  https://bugzilla.gnome.org/show_bug.cgi?id=797326

2018-10-11 09:24:53 +0900  Wonchul Lee <w.lee@lge.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: fix typo resurect to resurrect
	  https://bugzilla.gnome.org/show_bug.cgi?id=797273

2018-10-18 12:29:00 +0530  Amit Pandya <apandya@nvidia.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2h265enc.c:
	* sys/v4l2/gstv4l2h265enc.h:
	* sys/v4l2/meson.build:
	  v4l2videoenc: Add HEVC support
	  Add HEVC encoder support.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797141

2018-10-19 17:37:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  vl42allocator: Don't dup exported dmabufs
	  We can now use the new GstFAllocator to ask the allocator not to close
	  the wrapped FD. This way the dup is no longer needed.

2018-10-19 17:14:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Don't dup imported DMABuf FD
	  There is no specific needs to duplicate the FD. Unlike the exportation,
	  we don't depend on code that will call close. This will make debugging
	  easyer since the traced FD will match the exporter.

2018-10-23 13:04:34 +0200  Johan Bjäreholt <johanbj@axis.com>

	* gst/matroska/matroska-ids.c:
	  matroska-ids: Fix uninitialized memory in contexts
	  https://bugzilla.gnome.org/show_bug.cgi?id=797327

2018-10-19 17:02:11 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Add property for providing a threshold after which we create an edit list for gaps at the start
	  https://bugzilla.gnome.org/show_bug.cgi?id=797290

2018-10-22 12:21:54 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: Correctly set tkhd width/height to the display size
	  It was previously set to the display aspect ratio, e.g. 4x3, 16x9, etc.
	  but should be set to the display size.
	  This is a regression from e655d47dfce1652630fe8ff5fb6be56370087004
	  (1.5.1) and was correct before that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797318

2018-10-21 11:15:15 +0900  Yeongjin Jeong <gingerbk247@gmail.com>

	* tests/check/elements/flvmux.c:
	  tests: flvmux: Fix pushing invalid audio caps in tests
	  Previous commit created caps with incorrect aac codec data
	  that did not match the audio channel.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797256

2018-10-20 00:10:04 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix build with GLib versions < 2.54
	  g_ptr_array_find_with_equal_func was introduced in glib 2.54
	  which is a higher version than our minimum required one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797239

2018-10-17 13:52:20 +0200  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: fix up GHashTable-behavior dependent tests
	  GHashTable iteration order changed in recent GLib,
	  and tests were relying on that.
	  https://mail.gnome.org/archives/desktop-devel-list/2018-October/msg00016.html

2018-10-07 20:07:39 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't switch active streams and old streams ...
	  ... before the old streams is not exposed yet for MSS stream.
	  In case of DASH, newly configured streams will be exposed
	  whenever demux got moov without delay.
	  Meanwhile, since there is no moov box in MSS stream,
	  the caps will act like moov. Then, there is delay for exposing new pads
	  until demux got the first moof.
	  So, following scenario is possible only for MSS but not for DASH,
	  STREAM-START -> CAPS -> (configure stream but NOT EXPOSED YET)
	  -> STREAM-START-> CAPS (configure stream again).
	  In above scenario, we can reuse old stream without any stream reconfigure.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797239

2018-10-07 16:43:34 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Use GPtrArray to store QtDemuxStream structure
	  GPtrArray has less overhead than linked list and the length also
	  can be auto updated by using it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797239

2018-10-07 16:50:45 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Make QtDemuxStream refcounted structure
	  This a prework for porting GPtrArray.
	  Refcounting will help the use of g_ptr_array_new_with_free_func()
	  with QtDemuxStream structure
	  https://bugzilla.gnome.org/show_bug.cgi?id=797239

2018-10-06 20:19:40 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Make function foreach method friendly
	  https://bugzilla.gnome.org/show_bug.cgi?id=797239

2018-07-26 15:25:06 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Only set width/height in caps if they're non-0
	  If they are not valid, then let a downstream parser complete them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796878

2018-08-16 12:07:30 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix misleading debug line

2018-06-22 16:00:11 +0100  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Avoid warning when reporting about decryptors
	  https://bugzilla.gnome.org/show_bug.cgi?id=796652

2018-10-17 14:15:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audiofx/meson.build:
	* gst/deinterlace/meson.build:
	* gst/videobox/meson.build:
	* gst/videomixer/meson.build:
	  meson: Replace empty configuration_data() with copy keyword
	  Use 'copy' keyword to avoid meson warning message.
	  Note that 'copy' keyword in configure_file() is available
	  since meson 0.47.0
	  https://bugzilla.gnome.org/show_bug.cgi?id=797298

2018-10-16 15:42:12 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Do not hardcode frames_of_daily_jam
	  Apart from the obvious drawbacks of hardcoding, the drawback here was
	  that, if we subtracted 2 frames (instead of 2.6) from the target running
	  time, we'd request the next keyframe a bit too far into the future,
	  which would make our files split at the wrong position.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797293

2018-10-02 19:32:47 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Allow up to 1% of frame rate for lateness
	  https://bugzilla.gnome.org/show_bug.cgi?id=797290

2018-09-18 13:15:06 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	  rtpfunnel: fix shutdown
	  By disposing of the ssrc_to_pad map in finalize instead of
	  dispose.

2017-10-18 11:14:36 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpfunnel.c:
	* gst/rtpmanager/gstrtpfunnel.h:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/meson.build:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpfunnel.c:
	* tests/check/meson.build:
	  Initial commit of GstRtpFunnel
	  For funneling together rtp-streams into a single session.
	  Use-cases include multiplexing and bundle.

2018-10-12 22:33:15 +0900  Yeongjin Jeong <gingerbk247@gmail.com>

	* tests/check/elements/flvdemux.c:
	  tests: flvdemux: Add new test for channel detect using aac codec-data
	  https://bugzilla.gnome.org/show_bug.cgi?id=797275

2018-10-11 16:36:17 +0900  Yeongjin Jeong <gingerbk247@gmail.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Use aac codec-data to adjust channels if needed
	  Flv does not support various channels in AAC stream format, for example
	  flvdemux detect an audio channels of 2(stereo) when the AAC really is 1(mono).
	  https://bugzilla.gnome.org/show_bug.cgi?id=797275

2018-10-11 14:31:20 +0900  Yeongjin Jeong <gingerbk247@gmail.com>

	* tests/check/elements/flvmux.c:
	  tests: flvmux: Add new test for caps change after starting to write headers
	  https://bugzilla.gnome.org/show_bug.cgi?id=797256

2018-10-05 17:16:26 +0900  Yeongjin Jeong <gingerbk247@gmail.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Don't refuse caps changes after starting to write headers in streamable mode.
	  Flv does support changing the stream type and stream properties
	  after the headers were started to be written, and for example H264
	  codec_data changes can be supported.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797256

2018-10-11 13:55:01 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix if condition in drop-frame timecode wrap-around
	  Was previously: if ( x | y && a == b). Changed it into if ((x & y) && (a
	  == b)).

2018-10-09 16:39:11 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Subtract daily jam offset when day wraps around
	  For drop-frame framerates, when the expected next max timecode wraps
	  around at the end of the day, we have to subtract the offset of the
	  daily jam, otherwise we end up with a duration that's a few frames too
	  long.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797270

2017-09-25 14:30:13 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	* tests/check/elements/rtpmux.c:
	  rtpmux: respect downstream "timestamp-offset" in caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795162

2016-06-07 14:38:19 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* tests/check/elements/rtpmux.c:
	  rtpmux: cleanup ssrc-handling code a bit
	  And add some better logging.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795162

2016-05-04 11:48:04 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: protect against NULL caps
	  Due to state-changes deactivating the pad from another thread,
	  this can happen.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795162

2015-07-22 09:47:22 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	* tests/check/elements/rtpmux.c:
	  rtpmux: property should overrule both upstream and downstream
	  https://bugzilla.gnome.org/show_bug.cgi?id=762213
	  https://bugzilla.gnome.org/show_bug.cgi?id=795162

2018-10-08 20:45:08 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: use new 'python' module instead of deprecated
	  https://github.com/mesonbuild/meson/pull/4169

2018-10-08 20:35:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/gtk/meson.build:
	  meson: only build gtk gl examples if gst-gl was found
	  And fix typo in glliveshader example binary name.

2018-10-03 16:17:22 +0200  Peter Körner <git@mazdermind.de>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: accept pads named 'sink' on the muxer, handle static pads as well
	  https://bugzilla.gnome.org/show_bug.cgi?id=797241

2018-09-25 17:44:15 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairooverlay.h:
	  cairooverlay: Don't map input buffers if we just attach the overlay as meta
	  https://bugzilla.gnome.org/show_bug.cgi?id=797091

2018-09-25 17:02:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairooverlay.h:
	  cairooverlay: Add overlay as meta to the buffers if we can
	  This requires that downstream supports it and
	  draw-on-transparent-surface is enabled.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797091

2018-09-25 15:34:40 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/cairo/gstcairooverlay.c:
	  cairooverlay: Pre-multiply and un-premultiply alpha in case of ARGB32
	  Cairo expects pre-multiplied alpha, we work on un-premultiplied alpha.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797091

2018-09-25 15:31:20 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairooverlay.h:
	  cairooverlay: Add property for drawing on a transparent surface and then blending
	  This allows us to use the GstVideoOverlayComposition API and correctly
	  handle pre-multiplied alpha, while also only doing the alpha conversion
	  once instead of twice for the whole frame.
	  At a later point we can attach the meta to the buffer instead of
	  blending ourselves if downstream supports that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797091

2018-10-03 17:34:49 +0200  Thibault Saunier <tsaunier@igalia.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-read-common.c:
	  matroskdemux: do not use MapInfo.data after unmapping
	  And minor gst-indenting

2018-09-30 19:28:07 +0200  Yacine Bandou <yacine.bandou@softathome.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: Add the WebM encrypted content support in matroskademux
	  This commit:
	  1. Reads the WebM and Matroska ContentEncryption subelements.
	  2. Creates a GST_PROTECTION event for each ContentEncryption, which
	  will be sent before pushing the first source buffer.
	  The DRM system id field in this event is set to GST_PROTECTION_UNSPECIFIED_SYSTEM_ID,
	  because it isn't specified neither by Matroska nor by the WebM spec.
	  3. Reads the protection information of encrypted Block/SimpleBlock and
	  extracts the IV and the partitioning format (subsamples).
	  4. Creates the metadata protection for each encrypted Block/SimpleBlock,
	  with those informations: KeyID (extracted from ContentEncryption element),
	  IV and partitioning format.
	  5. Adds a new caps for WebM encrypted content named "application/x-webm-enc",
	  with the following new fields:
	  "encryption-algorithm": The encryption algorithm used.
	  values: "None", "DES", "3DES", "Twofish", "Blowfish", "AES".
	  "encoding-scope": The field that describes which Elements have been modified.
	  Values: "frame", "codec-data", "next-content".
	  "cipher-mode": The cipher mode used in the encryption.
	  Values: "None", "CTR".
	  https://bugzilla.gnome.org/show_bug.cgi?id=765275

2018-09-26 17:43:05 +0300  John Nikolaides <jnikolaides@toolsonair.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Added a split-at-running-time action signal
	  The video file can now be split at an arbitrary time, given by the user
	  as an argument to the action signal.
	  https://bugzilla.gnome.org/show_bug.cgi?id=787922

2018-09-21 19:47:44 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gdepay.h:
	  rtpmp4gdepay: detect broken senders who send AAC with ADTS frames
	  Strip ADTS headers if we detect any, apparently some Sony cameras
	  send AAC with ADTS headers. We could also change the stream-format
	  in the output caps, but that would be unexpected to pipeline builders
	  and would not exactly be backwards compatible.

2018-09-21 18:17:25 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: factor out pushing of output buffer

2018-09-26 13:29:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Allow ANY capsfeatures

2018-09-26 00:06:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	  docs: update for git master

2018-06-22 12:05:17 +0100  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: PIFF track encryption box support
	  The PIFF track encryption box is a UUID box containing the default encryption
	  values that should be used for PIFF sample encryption.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796647

2018-09-24 11:45:46 +0200  Nicola Murino <nicola.murino@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: add support for parsing more channel layouts ...
	  ... and fallback to gst_audio_info_set_format for not yet supported layouts.
	  Fix audio playback on iOS 12.
	  Based on patch from Byron Schiel <byron@canary.is>
	  https://bugzilla.gnome.org/show_bug.cgi?id=796919

2018-09-22 17:22:46 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: turn impossible condition into an assert
	  qtdemux_update_streams() is only ever called after checking
	  `qtdemux->streams_aware` is TRUE. There is no need to check for that
	  condition again.
	  `qtdemux->streams_aware` is only modified when the demuxer is
	  hard-resetted, which is mutually exclusive with demuxing, so it cannot
	  be modified during the call.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797191

2018-09-21 22:24:02 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Emit no-more-pads after parsing Tracks
	  Currently matroskademux does not emit no-more-pads until the first
	  Cluster is parsed, even though the Tracks have already been parsed and
	  from that point on there can be no more tracks.
	  This is important in MSE because the browser needs to know when the MSE
	  initialization segment has been completely parsed so that it can expose
	  the tracks to the user. Some applications depend on this been done
	  before they feed frames to the demuxer.
	  As a consequence, historically WebKit has relied on hacks such as
	  listening to the `pad-added` event, which made impossible to support
	  multiple tracks in the same file. Let's fix that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797187

2018-09-21 20:38:02 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Parse successive Tracks elements
	  This patch allows matroskademux to parse a second Tracks element,
	  erroring out if the tracks are not compatible (different number, type or
	  codec) and emitting new caps and tag events should they have changed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793333

2018-09-21 16:23:57 +0200  Alicia Boya García <aboya@igalia.com>

	  matroskademux: Refactor track parsing out from adding tracks
	  This splits gst_matroska_demux_add_stream() into:
	  * gst_matroska_demux_parse_stream(): will read the Matroska bytestream
	  and fill a GstMatroskaTrackContext.
	  * gst_matroska_demux_parse_tracks(): will check there are no repeated
	  tracks.
	  * gst_matroska_demux_add_stream(): creates and sets up the pad for the
	  track.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793333

2017-11-30 20:44:23 +0100  Alicia Boya García <ntrrgc@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Allow Matroska headers to be read more than once
	  This is necessary for MSE, where a new MSE initialization segment may be
	  appended at any point. These MSE initialization segments consist of an
	  entire WebM file until the first Cluster element (not included). [1]
	  Note that track definitions are ignored on successive headers, they must
	  match, but this is not checked by matroskademux (look for
	  `(!demux->tracks_parsed)` in the code).
	  Source pads are not altered when the new headers are read.
	  This patch has been splitted from the original patch from eocanha in [2].
	  [1] https://www.w3.org/TR/mse-byte-stream-format-webm/
	  [2] https://bug334082.bugzilla-attachments.gnome.org/attachment.cgi?id=362212
	  https://bugzilla.gnome.org/show_bug.cgi?id=793333

2018-08-16 21:42:37 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Implement split-after
	  The behaviour of split-now is to output the current GOP after
	  starting a new file.
	  The newly-added split-after signal will output the current GOP
	  to the old file if possible once a new GOP is opened.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796982

2018-09-20 12:12:55 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Don't leak codec_data buffer
	  Use gst_buffer_replace() to prevent buffer leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=797179

2018-09-18 18:13:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Set Closed Caption track width/height to that of the first video track
	  Otherwise software like Premiere or Final Cut Pro won't like our files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797111

2018-09-19 11:45:59 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* meson_options.txt:
	  meson: add glib-checks option to disable API guards and such
	  We want this enabled by default, also in releases, but people
	  may want to disable this for performance-critical workloads or
	  on embedded devices.

2018-09-19 11:45:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson_options.txt:
	  meson: fix missing closing bracket in option descriptions

2018-09-06 20:10:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Initialize caption track language code to 0 instead of "und"
	  Without this, Final Cut considers it "non-standard" and 0 (english) is a
	  good default for closed captions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=797111

2018-09-13 03:16:32 +0000  Song Bing <bing.song@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Add HEVC decoder support
	  https://bugzilla.gnome.org/show_bug.cgi?id=771686

2018-09-13 02:35:39 +0000  Nicolas Dufresne <nicolas@ndufresne.ca>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Move capture probe after input format is set
	  This is to support Amlogic CODEC driver which does not provide a full
	  list of formats when the driver is initially opened. GStreamer does
	  not strictly need this full list initially, but only later, in order
	  to negotiate with downstream if multiple format can be selected.
	  With this change, we will no longer probe twice the device, since the
	  probed list can be directly used for negotation.

2018-09-11 16:46:34 -0300  Ezequiel Garcia <ezequiel@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: Add a debug message beforing waiting for codec stop
	  Add a debug message right before waiting for the driver.
	  This is useful in order to debug drivers without a properly
	  implemented decoder or encoder stop command.

2018-09-10 13:18:45 -0300  Ezequiel Garcia <ezequiel@collabora.com>

	* sys/v4l2/gstv4l2.c:
	  v4l2: Add a debug message indicating probe operation
	  It's useful to see the v4l2 element running the probe
	  operation, to confirm it's turned on and working.

2018-09-10 13:18:30 -0300  Ezequiel Garcia <ezequiel@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2jpegenc.c:
	* sys/v4l2/gstv4l2jpegenc.h:
	* sys/v4l2/meson.build:
	  v4l2: Add JPEG encoding support
	  This commit adds the support for V4L JPEG stateful encoders.

2018-09-10 16:20:52 -0300  Ezequiel Garcia <ezequiel@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2fwhtenc.c:
	* sys/v4l2/gstv4l2fwhtenc.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/meson.build:
	  v4l2: Add FWHT codec support
	  The recently added vicodec (virtual codec) V4L driver
	  uses the Fast Walsh-Hadamard Transform for encoding
	  and decoding.
	  Add support for it.

2018-09-12 21:28:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/v4l2-common.h:
	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	  v4l2: Sync kernel header with linuxtv tree
	  This notably add HEVC and FWHT support, and VP8/9 profiles are now an
	  enumeration and their control exposed as a menu.

2018-09-12 17:24:00 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Allow up to 1 trak timescale unit of lateness in prefill mode
	  For 59.94 FPS, it's common to set 60000 as timescale. For that
	  timescale, if the audio is late by as little as 0:00:00.000016666
	  (definitely less than one audio sample), lateness gets rounded to 1.
	  Added a safeguard that allows lateness up to 1 sample with the specific
	  trak's timescale, to make sure that values less than e.g. one audio
	  sample won't break the prefill mode. What will happen in this case is
	  that the audio will get squeezed back to the video's timestamp, which in
	  practice means that the audio will be 0.000016666 seconds early (with
	  the patch).
	  https://bugzilla.gnome.org/show_bug.cgi?id=797133

2018-09-10 20:20:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix indentation

2018-09-11 00:18:32 +0000  Nicolas Dufresne <nicolas@ndufresne.ca>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Protect against zero PAR num/demu
	  This fixes an assertion when the driver implement CROPCAP but does
	  not set the PAR.

2018-09-12 00:52:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	  wavpackparse: fix handling of correction streams
	  Accept wavpack correction streams (.wvc) on sink pad, so
	  that wavpackparse can also be used to packetise correction
	  streams.
	  Fix parsing of subblock ID tags - the higher bits are
	  flags and are not part of the ID. This resulted in
	  correction blocks not being recognised properly and
	  the output not having the right (correction) caps.

2018-09-07 18:47:22 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/speex/meson.build:
	  meson: Explicitly pass -DWIN32 while building speex
	  The speex headers assume that WIN32 will always be defined when
	  building on Windows, but this is only true by default on MinGW.
	  Always set it explicitly.

2018-09-06 13:13:19 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Don't omit streamheader from caps on downstream reconfigure
	  The reconfigured downstream elements (e.g., dynamically added sink element)
	  most likely require the flv streamheader
	  https://bugzilla.gnome.org/show_bug.cgi?id=797089

2018-09-05 16:11:00 -0700  Martin Kelly <mkelly@xevo.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: don't store used UIDs
	  Currently, whenever we generate a 128-bit UID, we store it in a list and
	  return 0 if we ever encounter a collision. This is so mathematically
	  improbable that it's not worth checking for, so we can save memory and
	  time by not tracking the UID. Even if a collision happened, a list of
	  only 10 UIDs would be unlikely to detect it.
	  This article has a good description of how improbable a collision is:
	  https://en.wikipedia.org/wiki/Universally_unique_identifier#Collisions
	  https://bugzilla.gnome.org/show_bug.cgi?id=797086

2018-09-06 20:06:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Use existing helper function to create "und" language code

2018-09-05 20:15:57 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/meson.build:
	  meson: Don't skip plugins that don't build with MSVC
	  We now have options for all plugins, so we will just disable these in
	  the cerbero recipe instead. These require external deps, so they won't
	  affect gst-build either.

2018-09-03 16:04:33 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/mpg123/gstmpg123audiodec.h:
	  mpg123: Remove ssize_t fallback, not needed anymore
	  The mpg123 headers now contain a definition for ssize_t and building
	  with MSVC fails because of a redefinition for ssize_t

2018-07-31 12:52:36 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Keep sample data from the current fragment only (push mode)
	  This patch clears the sample table whenever the demuxing of a new
	  fragment begins. This avoids increasing memory usage for long videos.
	  This behavior was already present when upstream_format_is_time; this
	  patch extends it to all push mode operation (e.g. Media Source
	  Extensions).
	  https://bugzilla.gnome.org/show_bug.cgi?id=796899

2018-09-01 09:30:23 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	* sys/osxaudio/meson.build:
	  meson: Fix osxaudio build on iOS
	  Must define HAVE_IOS, and use appleframeworks dependency to ensure the
	  right frameworks are picked up.

2018-08-22 19:23:53 +0000  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Only offer MMAP/DMABUF pool
	  The propose allocation was offering a pool even in DMABUF_IMPORT or
	  USERPTR mode. These pool are internal only.

2018-08-22 17:51:52 +0000  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2transform.h:
	  v4l2transform: Add "disable-passthrough" property
	  This allow forcing going through the transform driver even if there
	  isn't an conversion happening. This is usedful when the m2m driver can
	  be used to adapt the type of memory between two drivers.

2018-08-31 14:25:09 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Reset frame/tc/caption pointer to NULL after moving it in the history

2018-08-29 09:51:42 +0200  Edward Hervey <edward@centricular.com>

	* gst/rtp/gstrtpmp4vpay.c:
	  rtpmp4vpay: Increase ranking
	  Both rtpmp4vpay and rtpmp4gpay support MPEG4 elementary streams. But
	  the most supported variant is the video-specific one (rtpmp4vpay),
	  therefore increase the rank of that one so that auto-plugging of
	  payloaders for MPEG4 elementary streams ends up picking that one
	  and not the generic one.

2018-08-15 12:53:34 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: implement keyframe search also without cluster prev size
	  If we have cluster prev size (GStreamer muxer will write it by default),
	  we can go back to the previous cluster efficiently, but if we don't then
	  just search backwards until we find a cluster ebml identifier, like we
	  do when searching for clusters in the bisection loop.

2018-08-15 12:14:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: make max backtrack distance for keyframe search configurable
	  Add property instead of hardcoding it in the code.
	  In some scenarios such as CCTV variable fps and extra long GOPs are
	  used to minimise storage space, for example. In those cases there might
	  not be any keyframes for many minutes, so provide a property to override
	  the max allowed distance.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790696

2018-08-15 11:49:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: set limit how much to backtrack to find a keyframe
	  If we seek without an index and land on a cluster that starts
	  with a delta frame.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790696

2018-08-15 11:25:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: no need to search for keyframes for intra-only streams
	  If the video streams are all I-frame only then we don't need to look
	  for a cluster with a keyframe, we can just assume there will be one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790696

2018-08-15 01:10:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: figure out if we have prev_size when starting up
	  This is useful to know in case someone initiates a seek or
	  direction change before we reach the second cluster.

2018-08-08 12:37:54 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: try to ensure keyframe when seeking without index
	  When seeking in pull mode without an index (because there is no index
	  or the file is still being written to) we bisect to find the right
	  cluster to jump to. However, it's possible the cluster we found doesn't
	  start with a keyframe, which leads to decoding errors, so if we know
	  that the found cluster starts with a delta frame try to scan back to
	  previous clusters until we find one that starts with a keyframe or
	  we are back at the beginning. Theoretically it's possible that all
	  clusters but the first one do not start with a keyframe and the
	  keyframes are in the middle of clusters, but this is extremely
	  unusual, so we will cover this case with a basic sanity check.
	  This problem is especially problematic with content recorded with
	  dynamic GOP and FPS, where long GOP lengths and low FPS may cause a
	  large set of clusters to lack key frames. Playback would then be
	  started on a non-keyframe cluster, and the large number of such frames
	  would make the content impossible to decode fo a long stretch of time.
	  Based on patch by: Mats Lindestam <matslm@axis.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=790696

2017-01-18 10:27:38 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: extract cluster prevsize if available
	  This is useful for reverse playback/trickmodes
	  without an index, and will also be useful in the
	  seek handler if we need to scan back to find a cluster
	  that starts with a keyframe.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790696

2018-07-25 19:27:01 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Validate stride/offset when importing
	  This will prevent situation where buffer size allow importing but rendering
	  goes wrong due to a miss-match in expected stride and offset.
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-08-01 13:07:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Add a method to try and import buffers
	  This method will check if a buffer, base on it's video meta,
	  can be imported. It will also try and adapt the request stride
	  in case this is the only that miss-match.
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-08-01 12:07:20 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Trace the buffer index we import to
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-07-25 22:16:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Fix typo in error message
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-07-24 12:07:22 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Only queue buffer if preparation worked
	  The preparation code imports the buffer, doing bunch of
	  validation. Only queue the buffer in the driver if the
	  importation worked. This way we don't rely on the driver
	  to validate.
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-07-24 12:05:45 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Only allow DMABuf export for STREAMING device
	  DMABuf exportation requires mmap, which requires STREAMING
	  capabilities.
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-07-13 14:42:21 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Activate the other pool first
	  This change has no effect. We will need to acquire a buffer from the
	  pool later in order to validate / adapt with the video alignment for
	  the downstream buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-07-09 15:33:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Simplify format handling
	  Always initially use try_format(), delaying set_format() to when the
	  allocation is being negotiated. This avoid having two code paths, and
	  will be help adding support for properly importing buffers of specific
	  strides and offsets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-08-23 22:57:35 +0200  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	  matroska: fix handling of FlagInterlaced
	  This is an enum not a boolean, and a value of 2 signals
	  that the video is progressive, but we would mistakenly set
	  interlace-mode=mixed on the output caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=787206

2018-08-09 15:14:05 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: complete colorspace info in debug log
	  The desired colorimetry is logged with all parameters (colorpsace,
	  range, matrix, and transfer function), but of the values actually
	  set by the driver, only colorspace is logged. Complete the debug
	  log message to display all colorimetry parameters:
	  Desired colorspace is 8:1:1:1
	  Got format of 640x480, format YU12, nb planes 1, colorspace 8
	  ->
	  Desired colorspace is 8:1:1:1
	  Got format of 640x480, format YU12, nb planes 1, colorspace 8:0:0:0
	  https://bugzilla.gnome.org/show_bug.cgi?id=796940

2018-08-09 15:12:57 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: fix typo in comment
	  https://bugzilla.gnome.org/show_bug.cgi?id=796940

2018-08-09 15:08:59 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: improve colorspace handling for JPEG sources
	  gstjpegdec sets 1:4:0:0 colorimetry (full range BT.601 YCbCr encoding
	  with unknown primaries and unknown transfer function). This currently
	  gets translated to bt601 or bt709 depending on resolution.
	  Both cases result in a negotiation failure:
	  ERROR: from element /GstPipeline:pipeline0/v4l2video0convert:v4l2video0convert0: Device '/dev/video0' does not support 1:4:0:0 colorimetry
	  Improve the guessing game by selecting JPEG colorimetry (JPEG colorspace
	  with sRGB transfer function) under these specific conditions, and loosen
	  the matching so that 1:4:0:0 input gets accepted if the device is
	  actually configured to 1:4:7:1 (V4L2_PIX_FMT_JPEG default).
	  https://bugzilla.gnome.org/show_bug.cgi?id=796940

2018-08-09 17:24:35 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: stop V4L2 from zeroing extended colorimetry for non-mplane
	  Setting the priv field to a magic value stops V4L2 core from zeroing
	  the extended colorimetry fields quantization, ycbcr_enc, and xfer_func
	  for non-mplane queues.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796940

2018-08-19 15:39:16 +0200  Zeeshan Ali <zeenix@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Remove a trailing whitespace
	  Otherwise, the latest gst-indent check doesn't pass.

2018-08-18 21:08:55 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* meson_options.txt:
	  meson: add options to disable gobject cast checks and glib asserts
	  ... and define G_DISABLE_DEPRECATED for development versions,
	  like we do in autotools.

2018-08-18 21:01:52 +0100  Tim-Philipp Müller <tim@centricular.com>

	* REQUIREMENTS:
	* ext/jpeg/meson.build:
	  meson: find libjpeg via pkg-config
	  This effectively (but optionally) requires libjpeg-turbo which
	  ships with a .pc file and is what pretty much everyone these days
	  uses anyway for libjpeg, so shouldn't be a problem hopefully.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796947

2018-08-17 17:35:43 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix build when SO_RCVBUFFORCE is not defined
	  This shoudl fix the mingw build.

2018-08-17 14:17:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Balance Linux value of get/set_rcvbuf
	  On Linux, the kernel returns twice the size as it will allocate extra
	  space for accouting. We devides this value by two in order to ensure
	  that get/set value now match. This fixes the set buffer size validation
	  and allow having a nice warning when the size if surpassed and the
	  process does not have CAP_NET_ADMIN capabilities.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727067

2018-08-17 14:05:04 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  updsrc: set udp buffer size forcibly
	  The udp buffer size is limited to a maximum of around 100K.
	  Some apps need to set the force bufsize for their own operation.
	  Use the SO_RCVBUFFORCE option in order to override the rmem_max limit
	  of linux kernel. Require user to have the CAP_NET_ADMIN privilege to
	  work.
	  Original patch from Kyungnam Bae <kyungnam.bae@lge.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=727067

2018-08-17 13:59:00 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: factor out gst_udpsrc_get_rcvbuf()
	  No semantic change.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727067

2018-08-17 19:11:21 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/libcaca/meson.build:
	* meson_options.txt:
	  meson: Rename caca option to libcaca
	  All options must match the plugin directory name.

2018-08-17 18:56:54 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson_options.txt:
	* sys/directsound/meson.build:
	* sys/meson.build:
	* sys/osxaudio/meson.build:
	* sys/osxvideo/meson.build:
	* sys/waveform/meson.build:
	  meson: Add build files for osxaudio, osxvideo, waveform
	  osxaudio is for macOS and iOS
	  osxvideo is for macOS
	  waveform is for Windows

2018-08-17 14:44:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	  docs: update for changes in master

2018-08-17 11:45:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/Makefile.am:
	  examples: dist qt examples
	  https://bugzilla.gnome.org/show_bug.cgi?id=796968

2018-08-17 00:27:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	* meson_options.txt:
	* tests/meson.build:
	  meson: Add an option for tests
	  This is needed because we don't always have gstreamer-check available,
	  for instance inside Cerbero on iOS.

2018-08-16 18:55:29 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/multifile/gstsplitmuxsink.c:
	  mp4 robust muxing: improve documentation and logging

2018-07-03 23:11:56 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	  rtph26*pay: Update param set timestamp even if parameters unchanged
	  rtph264pay and rtph265pay skip updating the parameter set timestamp if
	  the units they see contain no new configuration. This can result in
	  them injecting duplicate parameters.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796748

2018-08-15 13:43:53 +0200  Ulf Olsson <ulfo@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add support for SET_PARAMETER and GET_PARAMETER using signals
	  https://bugzilla.gnome.org/show_bug.cgi?id=792131

2018-08-15 02:28:20 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't leak old muxer/sink in async mode
	  Make sure to clear the reference taken earlier in the function
	  when switching muxer/sink asynchronously so they don't leak

2018-08-15 02:10:25 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Fix reference counting loop
	  The stream context was holding a reference to the
	  internal queue and pads, with pad probes that were
	  in turn holding references to the stream context.
	  This lead to a leak if the request pads weren't explicitly
	  released.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796893

2018-08-11 16:45:25 +0800  Roland Jon <rlandjon@gmail.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix codec_data buffer leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=740101

2018-08-02 16:12:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Handle closed captions as subtitle streams

2018-08-02 08:40:17 +0200  Iñigo Huguet <inigohuguet@hotmail.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: fix first input used is always used next times
	  The input from an v4l2 device that was used the first time was
	  remembered for next times, and set again always the pipeline is
	  set to READY state. This was making that users wasn't able to
	  select a different input without having to create a new pipeline.
	  This patch makes that v4l2src element forget previous used input
	  when going to NULL state, so it will check again for the current
	  selected input when going again to READY state. Users can change
	  to NULL state, select a new input with a VIDIOC_S_INPUT ioctl
	  and change to PLAYING again.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796908

2018-08-02 13:40:09 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: The sample size we have to reserve is 256+8 bytes for the header for CDP packets

2018-08-02 12:27:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Properly allocate 256 bytes per CDP packet in prefill mode
	  Instead of allowing 256 but only pre-allocating 100.

2018-08-02 12:27:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  Revert "qtmux: Allow for CDP packets up to 320 bytes"
	  This reverts commit 5eed1d49bdb7e7a632c7135656c482ed38a6ac2a.
	  255 is actually the maximum, there's a bug if more is arriving.

2018-08-01 16:50:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Allow for CDP packets up to 320 bytes
	  Apparently they can be bigger than 256 bytes sometimes.

2018-07-13 22:31:04 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp9pay.c:
	  rtppayload: Fix VP8/VP9/OPUS dual encoding name handling
	  All these were copy pasted and would lead to assertion when chained with
	  rtpmux. This commit rewrite the negotiation with downstream. This also
	  drop the fallback to ancient names if the pad is unlinked. This was
	  completly arbitrary decision that made no sense.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796809

2018-08-01 12:06:23 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtgl.h:
	  qt: Ensure GL headers are included
	  Otherwise there may be no valid typedef of GLsync.
	  ...
	  /usr/include/gstreamer-1.0/gst/gl/gstglfuncs.h:93:24: note: in definition of macro 'GST_GL_EXT_FUNCTION'
	  ret (GSTGLAPI *name) args;
	  ^~~~
	  /usr/include/gstreamer-1.0/gst/gl/glprototypes/sync.h:33:23: error: 'GLsync' has not been declared
	  (GLsync sync))
	  ^~~~~~
	  ...
	  https://bugzilla.gnome.org/show_bug.cgi?id=796879

2018-08-01 03:18:58 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/oss4/meson.build:
	  meson: Fix oss4 header checks
	  Otherwise, oss4 ends up getting built when force-disabled.

2018-08-01 01:10:49 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/v4l2/meson.build:
	  meson: Fix missing variable in v4l2 build

2018-07-31 12:47:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtdemux: Don't assert in prefill mode if a track has no samples at all
	  Just write it with a duration of 0, no samples, etc.

2018-07-31 12:33:54 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't assert if a file does not have any active streams
	  ** (gst-play-1.0:9113): CRITICAL **: 12:31:54.360: qtdemux_is_streams_update: assertion 'qtdemux->active_streams != NULL' failed

2018-07-30 13:33:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Reserve 256 bytes for CDP packets in pre-fill mode
	  92 is sometimes too small and compared to the wasted space for other
	  codecs 256 bytes is small (and should be the maximum CDP packet size)

2018-07-25 07:35:28 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/aalib/meson.build:
	* ext/cairo/meson.build:
	* ext/dv/meson.build:
	* ext/flac/meson.build:
	* ext/gdk_pixbuf/meson.build:
	* ext/gtk/meson.build:
	* ext/jack/meson.build:
	* ext/jpeg/meson.build:
	* ext/lame/meson.build:
	* ext/libcaca/meson.build:
	* ext/libpng/meson.build:
	* ext/mpg123/meson.build:
	* ext/pulse/meson.build:
	* ext/qt/meson.build:
	* ext/raw1394/meson.build:
	* ext/shout2/meson.build:
	* ext/soup/meson.build:
	* ext/speex/meson.build:
	* ext/taglib/meson.build:
	* ext/twolame/meson.build:
	* ext/vpx/meson.build:
	* ext/wavpack/meson.build:
	* gst/matroska/meson.build:
	* gst/meson.build:
	* meson.build:
	* meson_options.txt:
	* sys/directsound/meson.build:
	* sys/meson.build:
	* sys/oss/meson.build:
	* sys/oss4/meson.build:
	* sys/v4l2/meson.build:
	* sys/ximage/meson.build:
	* tests/examples/qt/qmlsink/meson.build:
	* tests/examples/qt/qmlsrc/meson.build:
	* tests/icles/meson.build:
	* tests/meson.build:
	  meson: Add feature options for all plugins
	  Checks for GL, Qt5, and C++ are still automagic. FIXMEs have been
	  added for these so they can be fixed later.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795107

2018-07-25 17:15:53 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	  deinterlace: Closed caption pass-through
	  Pass through closed caption data when deinterlacing. When two
	  deinterlaced frames are created for the same interlaced frame (e.g.
	  fields=all), the second of the two frames will have no closed caption
	  data.
	  Also fixed memory leaks related to timecode meta pass-through.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796876

2018-07-25 18:37:48 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Implement muxing of AV1 into MP4 files
	  According to
	  https://aomediacodec.github.io/av1-isobmff/

2018-07-25 17:09:06 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Put codec_data as CodecPrivate for AV1

2018-07-25 17:08:53 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Accept muxing AV1

2018-07-25 16:51:38 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Recognize more AV1 atoms

2018-07-25 16:39:18 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Extract codec_data for AV1
	  According to
	  https://github.com/Matroska-Org/matroska-specification/blob/av1-mappin/codec/av1.md

2018-07-25 14:31:39 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Extract AV1 codec_data and put it in the caps
	  Also extract the presentation-delay and put it in the caps.

2018-07-25 10:43:11 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add initial support for AV1 demuxing
	  Following the spec at
	  https://aomediacodec.github.io/av1-isobmff/

2018-07-27 00:41:57 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add a small configurable teardown delay
	  This causes rtspsrc to send a teardown and wait on
	  PAUSED->READY transition, with a configurable delay.
	  Otherwise, typically teardown never gets sent in
	  playbin / uridecodebin where the transition back to NULL
	  happens too quickly.
	  The timeout is set to 100ms default.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751994

2018-07-26 16:43:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  rtpgstpay: Add support for force-keyunit events
	  This triggers immediate re-sending of the configuration data in-band.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796877

2018-07-13 19:45:19 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp: Use running_time instead of PTS for config-interval calculations
	  PTS can start again from a different offset while the running time is
	  increasing. The only thing that matters here is the running time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796807

2018-07-19 22:48:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/examples/gtk/meson.build:
	* tests/examples/meson.build:
	  example: Build GTK  and GTK GL example code

2018-07-19 17:31:03 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/rtp/gstrtpL8pay.c:
	  rtpL8pay: don't try to modify a read-only structure
	  Just remove the code. It's not doing anything useful anyways. The modified
	  caps are the result of a caps query, so either not used afterwards of a
	  reference to some internal caps of another element that should not be
	  modified.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796837

2018-07-17 08:23:54 +0200  Iñigo Huguet <inigohuguet@fanamoel.com>

	* ext/qt/gstqtgl.h:
	  qmlgl: Fix conflicting declaration of type GLsync for non-android
	  https://bugzilla.gnome.org/show_bug.cgi?id=796821

2018-07-16 19:03:39 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/gstdeinterlacemethod.h:
	  deinterlace: Timecode pass-through
	  When it is trivial to pass-through a timecode, by only removing the
	  "interlaced" flag, do pass-through. Otherwise, double the fps_n and
	  adjust the "frames" field.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796818

2018-07-17 00:03:19 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: Improve handling of repeated timestamps
	  When handling input with timestamps that repeat, sometimes
	  splitmuxsink would get confused and ignore a keyframe.
	  The logic in question is a holdover from before the cmd queue
	  moved the file cutting to the multiqueue output side and made
	  it deterministic, so it's no longer needed on the input
	  here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796773

2018-07-17 01:33:55 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  Revert "splitmuxsrc: Make sure events are writable"
	  This reverts commit 3ac5430311b20f30814cdabf5724fb687748bb5b.
	  There's no need to make a freshly created event writable,
	  and the other half of this patch was already fixed
	  and pushed in f2f15a1

2018-07-16 23:43:29 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Make sure events are writable
	  Before setting the seqnum on events sent downstream,
	  make sure they are writable.

2018-07-13 16:51:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Validate that capture buffers were queued
	  When the pool is started, we allocate and release buffer, expecting
	  the pool release-buffer handler to queue them. Though, as we rely
	  on release function, there is no direct way to detect that this
	  process didn't work.
	  To check this, validate that the number of queued buffer is the same
	  as the number of allocated buffers. This allow returning an error
	  when buffer importation was refused by the driver.
	  https://bugzilla.gnome.org/show_bug.cgi?id=583890

2018-07-13 16:02:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Only return eos for M2M devices
	  This will avoid sending EOS on v4l2src when a driver sends an empty
	  buffers. This case would be a bug in the driver, but yet the camera
	  should keep running.
	  This also removes the check for corrupted buffers, as this check is
	  already done later.
	  https://bugzilla.gnome.org/show_bug.cgi?id=794842

2018-07-13 15:58:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/v4l2-utils.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: Add a macro to check for M2M
	  https://bugzilla.gnome.org/show_bug.cgi?id=794842

2018-07-13 14:41:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix userptr importation
	  The length passed to the driver was always 0 instead of the size of
	  the memory. This would fail validation in videobuf2.

2018-07-12 15:11:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Remove duplicate check
	  We were calling gst_v4l2_is_buffer_valid() before and inside
	  gst_v4l2_buffer_pool_qbuf() as we needed to access the group. The second
	  check failed since the writability of the buffer get inherited from the
	  GstMemory, which lead to pipeline failure. As we cannot avoid the extra
	  ref, it would be racy otherwise, just pass the group to _dbuf() so it
	  does not have to call gst_v4l2_is_buffer_valid() again.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796692

2017-08-25 11:58:12 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Don't start the RTCP thread until it's needed
	  Always wait with starting the RTCP thread until either a RTP or RTCP
	  packet is sent or received. Special handling is needed to make sure the
	  RTCP thread is started when requesting an early RTCP packet.
	  We want to wait with starting the RTCP thread until it's needed in order
	  to not send RTCP packets for an inactive source.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2018-07-11 12:21:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Try to avoid TRY_FMT when camera is streaming
	  Some camera firmware crash is TRY_FMT is called during streaming. As a
	  side effect. This try and detect that the same format as currently
	  running is about to be tried, and skip renegotiation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796789

2018-07-09 13:59:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Protect double calls to set_format()
	  In some cases, set_format() may get called twice before the output
	  format is set. Running an allocation query in this case is both not
	  needed and will cause assertion due tot he NULL caps.

2018-07-08 20:08:18 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Mark as Audio in classification

2018-07-06 15:21:33 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Store and propagate SEGMENT sequence numbers
	  * When receiving a segment in TIME, use that seqnum
	  * Only reset the stored sequence number when doing HARD reset
	  (and not when we get a FLUSH event from upstream)

2018-07-01 15:27:32 -0400  Michael Tretter <m.tretter@pengutronix.de>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Implement stable element name
	  The first converter to be found will now gain the name v4l2convert.
	  Other converters will be named after the m2m dev node end point they are
	  attached to.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784958

2018-06-13 17:39:57 +0100  Philippe Normand <philn@igalia.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Set subtitle tag title from TrackName field
	  GUI applications can then use the title tag to set menu items or labels
	  representing the track.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796567

2018-06-28 19:08:35 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Only renegotiate with upstream
	  When the decoder get linked further, it will receive a renegotiation
	  event from downstream. This case is not supported and should be ignored.
	  This fixes issues when this encoder is used inside an GstRtspServer
	  pipeline.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796525

2018-06-09 23:58:01 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: rework segment event pushing, again
	  This patch aims at fixing the recent regressions in the adaptive test
	  suite.
	  All segment pushing in push mode is now done with
	  gst_qtdemux_check_send_pending_segment(), which is idempotent and
	  handles both edit lists cases and cases where the upstream TIME segments
	  have to be sent directly.
	  Fragmented files that start with a non-zero tfdt are also taken into
	  account, but their handling has been vastly simplified: now they are
	  handled as implicit default seeks so there is no need to extend the
	  GstSegment formulas as was being done before.
	  qtdemux->segment.duration is no longer modified when
	  upstream_format_is_time, respecting in this way the durations provided
	  by dashdemux and fixing bugs in reverse playback tests where mangled
	  durations appeared in the emitted segments.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2018-06-17 02:01:59 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't send EOS during upstream reverse playback
	  Upstream driving elements such as dashdemux often do reverse playback by
	  feeding qtdemux with the fragments containing the requested playback
	  range in reverse order.
	  But the requested playback range stop may be somewhere in the
	  middle of a fragment. In that case, a naive pts >= segment.stop
	  condition may declare end of segment prematurely when demuxing this
	  first fragment.
	  This used not to happen because there were places in moov parsing where
	  segment.stop was overwritten to GST_CLOCK_TIME_NONE even if
	  upstream_format_is_time -- resulting in this case in a segment with rate
	  < 0 and stop == -1 and hence not triggering the EOS check, but that was
	  likely an accident.
	  This patch modifies the EOS check to take this case into account, not
	  sending EOS when upstream_format_is_time if rate < 0.
	  This fixes adaptive.dash.playback.seek_end_live.DASHIF_livestream_testpic_2s
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2018-02-06 13:51:14 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: fold property set/get PROP_OUTPUT_IO_MODE case into default
	  https://bugzilla.gnome.org/show_bug.cgi?id=796714

2018-06-22 14:56:31 +0000  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Don't set colorimetry on capture
	  The colorimetry will be set along with the raw format and those fields
	  will then be copied from sink to src caps by the gst encoder.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791471

2018-06-27 16:57:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Really always set colorimetry
	  This fixes patch dd1c5aed656e07e3dad01f83410f3af16cfb14cf which
	  pretended to always set colorimetry but the patch was incomplete.
	  This is again best effort considering the spec says that for CAPTURE
	  you may only read this value.

2018-06-26 15:04:39 +0200  Michael Tretter <m.tretter@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: do not call streamon while pool is flushing
	  gst_v4l2_buffer_pool_flush() executes streamoff for the output, but
	  streamoff->streamon for the capture of the decoder.
	  gst_v4l2_buffer_pool_streamon() on capture assumes that is able to
	  resurrect the buffers from the pool, but acquiring buffers fails if the
	  buffer pool is still flushing.
	  The decoder needs to stop flushing the pools before calling
	  gst_v4l2_buffer_pool_flush() to restart the v4l2 device. Otherwise
	  starting the decoding thread might fail, because there are no buffers in
	  the capture pool.
	  This fixes a regression that was introduced in 97985a335c78
	  ("v4l2videodec: Add dynamic resolution change support").
	  https://bugzilla.gnome.org/show_bug.cgi?id=796681

2018-06-25 16:03:17 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: use S_SELECTION instead of S_CROP in gst_v4l2_object_set_crop
	  The S_CROP call doesn't work on mem2mem output queues. Use the
	  S_SELECTION call to set the crop rectangle and only fall back to
	  S_CROP for ancient kernels.
	  This will allow v4l2videoenc to set the coded size on the output
	  queue via S_FMT and then set the visible size via the crop rectangle,
	  as required by the V4L2 codec API.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796672

2018-06-27 13:46:00 +0000  Marian Mihailescu <mihailescu2m@gmail.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: activate capture pool after output pool
	  Some drivers need output buffers set before capture buffers.
	  CODA cannot set output format if capture is streaming.
	  Exynos MFC fails on output STREAMON if capture is already streaming.
	  This patch delays capture activation until output is configured and
	  streaming
	  https://bugzilla.gnome.org/show_bug.cgi?id=796693

2018-06-23 23:44:19 +0200  Tim-Philipp Müller <tim@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	  Update for g_type_class_add_private() deprecation in recent GLib
	  https://gitlab.gnome.org/GNOME/glib/merge_requests/7

2018-06-20 10:03:59 +0200  Edward Hervey <edward@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Protect input stream with lock
	  This was the last remaining place where modifying/unreffing the
	  input stream was not protected by the lock
	  https://bugzilla.gnome.org/show_bug.cgi?id=796639

2018-06-18 12:13:48 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Make sure events are writable before setting their seqnum

2018-05-28 15:19:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Drop truncated frames
	  Drop truncated frames regardless if they have the ERROR flag or not.
	  Truncated frame causes video frame map failure in many elements
	  including cluttersink, glupload etc.

2018-04-02 12:59:33 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Try return input buffer soon
	  In this patch we use a non-blocking poll in order to return all input
	  buffers (buffers from v4l2-output queue). This prevent holding too long
	  on upstreaming buffer in importing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=794904

2018-06-07 13:56:03 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/meson.build:
	* tests/examples/qt/qmlsink/meson.build:
	* tests/examples/qt/qmlsrc/meson.build:
	  qt: also check for un-suffixed moc
	  e.g. Qt windows installer doesn't have suffixes

2018-06-06 11:44:33 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Do not set INVALID seqnum on events

2018-06-01 22:47:10 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* tests/check/elements/qtdemux.c:
	  tests: qtdemux: Add checking exposed segment event
	  https://bugzilla.gnome.org/show_bug.cgi?id=796480

2018-06-01 21:08:10 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Forward upstream time-format segment without mapping
	  Sample table based segment event (genereted by qtdemux) could break
	  presentation timeline. For example, qtdemux should not modify upstream
	  time format segment (e.g., adaptivedemux use case)
	  https://bugzilla.gnome.org/show_bug.cgi?id=796480

2018-04-19 08:14:47 +0200  Edward Hervey <edward@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Seek handling is always done with a valid event
	  Remove the checks

2018-06-06 07:46:54 +0200  Edward Hervey <edward@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Don't set invalid seqnum on events
	  Some codepath will call gst_wavparse_perform_seek without an event
	  and therefore without a valid seqnum

2018-05-25 12:28:04 +0200  Thibault Saunier <tsaunier@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Clarify field name about stream-encryption-system
	  This field is actually only informatory and the user can potentially
	  choose something else. EME tests in WebKit testsuite actually doesn't
	  take it into and force another encryption system to be used, and expects
	  to be given the occasion to do so.
	  This basically also reverts 3e063703b3a51b8aaa7f75f36c4660c583a60e93.

2018-05-28 11:01:42 -0700  Thiago Santos <thiagossantos@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: mark segment as sent after pushing when moov is received
	  Otherwise we would try to send it a second time if the same moov is
	  received or in any other situation that might trigger segment sending.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2018-05-28 10:59:14 -0700  Thiago Santos <thiagossantos@gmail.com>

	* tests/check/elements/qtdemux.c:
	  tests: qtdemux: Avoid using data beyond array and improve error msg
	  Makes it easier to debug the failures as well as prevents problems
	  reading out of bounds data.

2018-05-16 20:16:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't open the device in get property
	  This is both racy and inefficient. This function is still missing some
	  locking which will be address in later patch.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796185

2018-05-27 20:29:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtpstorage.c:
	* tests/check/elements/rtpulpfec.c:
	  tests: rtpstorage: fix potential crashes / test failures on 32-bit
	  Pass 64 bits to g_object_set() for 64-bit integer properties like
	  rtpstorage's "size-time" property.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796429

2018-05-13 21:59:49 -0700  Thiago Santos <thiagossantos@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: do not update segment.stop is it is not a valid time
	  Otherwise it overflows and starts having a meaningful and wrong value.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2016-04-26 16:54:30 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: offset edts segments by the min timestamp of the stream
	  Otherwise if the stream is starting at timestamp=X it would wait
	  'X' to start playing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2016-04-26 14:34:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: rework segment event pushing
	  Instead of always keeping a safe segment (start=0) event from the beginning,
	  delay the creation of this event to when we really know the timestamp of the
	  first sample. This is important to properly start fragmented streams that
	  we might join in the middle or to play isolated fragment files that might
	  have an advanced tfdt.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2018-05-25 10:49:21 +0200  Thibault Saunier <tsaunier@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Do not unref a NULL stream_tags
	  stream->stream_tags is reset to NULL once we expose the stream and
	  these have been consumed, we need to check that when cleaning up
	  the stream.

2018-05-25 10:17:29 +0200  Thibault Saunier <tsaunier@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Do not run the preferred decryptor context query if no decryptor avalaible
	  Ultimately this avoids a segfault as the code expect a non NULL array
	  here.

2018-03-30 17:03:13 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Allow edit lists on fragmented files on push mode
	  Fragmented files often use elst.duration=0 which before
	  ee78825eaef2c5fffac7d6c5526fe18cec6b3eef was wrongly interpreted as
	  having no frames.
	  Since that issue has now been fixed, there is no reason to disable edit
	  lists in fragmented files. This commit enables them, therefore producing
	  correct stream time for files containing edit lists.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793058

2018-05-24 12:58:00 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix computation of first_duration for fragmented files in push mode
	  Since ca068865c391e87932b1268d0c675be233dd2ffe the duration of the first
	  frame is not used for estimating the frame rate.
	  For this purpose, stream->first_duration was initialized with the
	  duration of the first frame. In fragmented files, this was previously
	  done by peeking the first moof, but that can only be done in pull mode.
	  Fortunately, we don't really need to do that, at least with the current
	  design: When we are estimating the frame rate we already have the
	  sample table, regardless of the scheduling mode and whether the file is
	  fragmented or not, so we can obtain first_duration there much more
	  reliably.
	  This fixes frame rate estimation for fragmented files in push mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796384

2017-06-13 17:42:55 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* tests/check/elements/splitmux.c:
	  splitmuxsink: Added new async-finalize mode
	  This mode is useful for muxers that can take a long time to finalize a
	  file. Instead of blocking the whole upstream pipeline while the muxer is
	  doing its stuff, we can unlink it and spawn a new muxer+sink combination
	  to continue running normally.
	  This requires us to receive the muxer and sink (if needed) as factories,
	  optionally accompanied by their respective properties structures. Also
	  added the muxer-added and sink-added signals, in case custom code has to
	  be called for them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783754

2018-05-23 19:00:48 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't send gaps bigger than 1 second (now in push mode too)
	  This applies the same workaround to gaps that is being used in pull
	  mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778426

2018-05-23 20:08:56 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Properly handle edit list in push mode
	  If there are empty segments in edit list, demux should
	  adjust "accumulated_base" to apply it into running time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778426

2018-05-22 22:14:03 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: write colorimetry
	  This is a straightforward translation of 5dd39d8, can be trivially
	  checked by running:
	  gst-launch-1.0 -v videotestsrc ! video/x-raw, colorimetry=2:4:7:1 ! \
	  matroskamux ! matroskademux ! fakesink
	  and verifying that the colorimetry is correctly preserved.
	  https://bugzilla.gnome.org/show_bug.cgi?id=796344

2018-03-31 17:19:03 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix buggy duration in edits with duration=0 in fragmented files without a mehd
	  https://bugzilla.gnome.org/show_bug.cgi?id=794858

2018-05-23 13:14:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstmultiudpsink.c:
	  docs: fix typos

2018-03-31 18:42:47 +0900  Seungha Yang <pudding8757@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Clarify variable name
	  As defined by spec, use "empty edit". It's more straightforward.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778426

2017-06-21 17:59:21 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: add context for a preferred protection
	  qtdemux selected the first system corresponding to a working GStreamer
	  decryptor. With this change, before selecting that decryptor, qtdemux
	  will check if it has context (a preferred decryptor id) and if not, it
	  will request it.
	  The request includes track-id, available key system ids for the
	  available decryptors and even the events so that the init data is
	  accessible.
	  [eocanha@igalia.com: select the preferred protection system even if not available]
	  Test "4. ClearKeyVideo" in YouTube leanback EME conformance tests 2016 for
	  H.264[1] uses a media file[2] with cenc encryption which embeds 'pssh' boxes
	  with the init data for the Playready and Widevine encryption systems, but not
	  for the ClearKey encryption system (as defined by the EMEv0.1b spec[3] and with
	  the encryption system id defined in [4]).
	  Instead, the ClearKey encryption system is manually selected by the web page
	  code (even if not originally detected by qtdemux) and the proper decryption key
	  is dispatched to the decryptor, which can then decrypt the video successfully.
	  [1] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2016.html?test_type=encryptedmedia-test&webm=false
	  [2] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/media/car_cenc-20120827-86.mp4
	  [3] https://dvcs.w3.org/hg/html-media/raw-file/eme-v0.1b/encrypted-media/encrypted-media.html#simple-decryption-clear-key
	  [4] https://www.w3.org/Bugs/Public/show_bug.cgi?id=24027#c2
	  https://bugzilla.gnome.org/show_bug.cgi?id=770107

2017-05-20 16:55:40 +0000  Enrique Ocaña González <eocanha@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: also push buffers without encryption info instead of dropping them
	  Test "17. PlayReadyH264Video" in YouTube leanback EME conformance tests 2016
	  for H.264[1] uses a media file[2] with cenc encryption whose first two 'moof'
	  boxes have no encryption information (no 'saiz' and 'saio' boxes).
	  Those boxes are actually not encrypted and the current qtdemux implementation
	  was just dropping them, breaking the test use case.
	  This patch detects those kind of situations and just lets the unencrypted
	  buffers pass. Of course, this needs some collaboration by the decryptors,
	  which should also do the same and not to try to decrypt those clear buffers.
	  [1] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2016.html?test_type=encryptedmedia-test&webm=false
	  [2] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/media/oops_cenc-20121114-142.mp4
	  https://bugzilla.gnome.org/show_bug.cgi?id=770107

2018-05-21 11:49:08 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: use cdata.set_quoted() in more places

2018-05-21 11:46:59 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* meson_options.txt:
	  meson: add 'nls' option to disable translations
	  And enable by default. Was implicitly disabled because
	  ENABLE_NLS was not defined.

2016-02-09 14:00:00 -0800  Andre McCurdy <armccurdy@gmail.com>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: ensure valid sentinal for gst_structure_get()
	  gst_structure_get() is declared with G_GNUC_NULL_TERMINATED, ie
	  __attribute__((__sentinel__)), which means gcc will generate a
	  warning if the last parameter passed to the function is not NULL
	  (where a valid NULL in this context is defined as zero with any
	  pointer type).
	  The C code callers to gst_structure_get() within gst-plugins-good
	  use the C NULL definition (ie ((void*)0)), which is a valid sentinel.
	  However gstid3v2mux.cc uses the C++ NULL definition (ie 0L), which
	  is not a valid sentinel without an explicit cast to a pointer type.
	  Upstream-Status: Pending
	  Signed-off-by: Andre McCurdy <armccurdy@gmail.com>

2016-02-03 18:12:38 -0800  Andre McCurdy <armccurdy@gmail.com>

	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	  raw1394: avoid including <sys/poll.h> directly
	  Note from Edward Hervey: Patch from git.yoctoproject.org
	  musl libc generates warnings if <sys/poll.h> is included directly.
	  Upstream-Status: Pending
	  Signed-off-by: Andre McCurdy <armccurdy@gmail.com>

2018-02-23 13:38:32 +0100  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux_parse_segments: remove superfluous variable
	  https://bugzilla.gnome.org/show_bug.cgi?id=793751

2018-04-23 13:29:30 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Remove custom get_next_time implementation
	  GstAggregator now does the same thing in the simple implementation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795486

2018-05-15 11:50:30 +0100  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: Add tests for PLI and FIR
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2018-04-30 08:41:19 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: make "clear-pt-map" action signal actually work
	  Needed for PLI + FIR unit tests in follow-up commit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2016-10-06 16:08:38 +0200  Mikhail Fludkov <misha@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Avoid unnecessary copy of stats structure
	  The code before copied GstStructure twice. The first time inside
	  gst_value_set_structure and the second time in g_value_array_append.
	  Optimized version does no copies, just transfers ownership to
	  GValueArray. It takes advantage of the fact that array has already
	  enough elements preallocated and the memory is zero initialized.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2018-05-15 10:35:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/replaygain/gstrgvolume.c:
	  Revert "BugFix : Change peak value to normalize audio file with fallback gain"
	  This reverts commit 36e49fd6f872f0b3f33083107a55fb7f671a47d0.
	  Breaks unit test, someone needs to investigate if it's the
	  patch's fault or if the test needs adjusting/updating.
	  https://bugzilla.gnome.org/show_bug.cgi?id=673970

2016-12-13 10:13:52 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Drop packet if trying to send from non-internal source
	  If obtain_internal_source() returns a source that is not internal it
	  means there exists a non-internal source with the same ssrc. Such an
	  ssrc collision should be handled by sending a GstRTPCollision event
	  upstream and choose a new ssrc, but for now we simply drop the packet.
	  Trying to process the packet further will cause it to be pushed
	  usptream (!) since the source is not internal (see source_push_rtp()).
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2018-05-14 00:29:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: tag disabled streams with FLAG_UNSELECT
	  So they're never picked as default, only by explicit
	  user action.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690911

2018-05-14 21:06:55 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Print expected/actual values in debug log on mismatch in prefill mode
	  This helps debugging a lot.

2018-04-10 18:05:47 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Try media_ssrc if no src can be found for PLI sender_ssrc
	  Some RTP stacks out there does not set the sender_ssrc. In order to be
	  more robust, try to lookup the media_ssrc before dropping the PLI.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2017-08-25 11:59:00 +0200  Mikhail Fludkov <misha@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Fix on-feedback-rtcp race
	  If there is an external source which is about to timeout and be removed
	  from the source hashtable and we receive feedback RTCP packet with the
	  media ssrc of the source, we unlock the session in
	  rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
	  allowing rtcp timer to kick in and grab the lock. It will get rid of
	  the source and rtp_session_process_feedback will be left with RTPSource
	  with ref count 0.
	  The fix is to grab the ref to the RTPSource object in
	  rtp_session_process_feedback.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2017-11-27 10:56:47 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Add missing lock around sess->ssrcs iteration
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2017-08-25 11:22:47 +0200  John-Mark Bell <jmb@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: do not emit RBs for internal senders.
	  These are the sources we send from, so there is no reason to
	  report receive statistics for them (as we do not receive on them,
	  and the remote side has no knowledge of them).
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2018-04-10 18:22:57 +0200  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpsession.c:
	  tests: rtpsession: fix indentation
	  https://bugzilla.gnome.org/show_bug.cgi?id=795139

2018-05-12 08:03:28 +0200  Edward Hervey <edward@centricular.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Fix typo in debug messages
	  It's a decoder, not an encoder :)
	  https://bugzilla.gnome.org/show_bug.cgi?id=795941

2018-03-22 18:00:37 +0100  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Added caption_%u pad template
	  For closed-caption-enabled muxers (e.g. qtmux)

2018-05-10 13:57:30 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Initialize riff library
	  Avoids debugging message issues. Also just use the main riff header

2018-05-08 20:31:41 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* tests/check/elements/qtdemux.c:
	* tests/check/elements/qtdemux.h:
	  tests: qtdemux: Add test for stream change
	  Add test case to verify track-id change and stream change
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-05-08 20:30:18 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Protect _expose_streams() from flush event
	  Flush during stream change can break autoplugging or the
	  flush event could be dropped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-05-08 20:26:41 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Try to expose whenever got new moov or new stream-start
	  Whenever got new moov or new stream-start,
	  demux will try to expose new pad by following rule.
	  Comparing stream-id in the current moov with previous one, then
	  * If matched stream-id is found from previous one,
	  reuse existing pad (most common case)
	  * Otherwise, expose new pad with new stream-start
	  * No more used stream will be freed
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-05-08 20:10:39 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Remove duplication of initializing member variables
	  Most initialization of variables in gst_qtdemux_init() are duplicated in
	  gst_qtdemux_reset() function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-05-08 20:09:10 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Create stream whenever got new moov
	  Whenever demux got moov, demux will create new stream. Only exception is
	  duplicated track-id in a moov box. In that case the first stream
	  will be accepted. This patch is pre-work for rework of moov handling.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-05-08 19:57:11 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Store stream-id to manage streams
	  In order to figure out stream change such as
	  track-id change or stream-id change, demux will store
	  stream-id per QtDemuxStream structure.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-05-08 19:39:02 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Use GList to manage QtDemuxStream
	  * Move to GList from static array
	  * Logging track-id instead of array index. It's more meaningful.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-05-08 18:44:15 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Adjust the number of args of some functions
	  To be used with g_list_free_full in the next patch
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-05-08 18:22:58 +0900  Seungha Yang <seungha.yang@navercorp.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add parentheses in macro
	  https://bugzilla.gnome.org/show_bug.cgi?id=684790

2018-03-19 23:36:13 +0100  Marinus Schraal <mschraal@gnome.org>

	* gst/isomp4/qtdemux.c:
	  isomp4: Use full date time if available
	  The ©day tag contains a full date time, use it for the DATE_TIME tag
	  instead of just the DATE tag. This overrules the unreliable qt creation
	  time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731029

2018-03-15 22:59:39 +1100  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix doc comment markers

2016-02-02 18:58:24 +0200  Kyrylo Polezhaiev <kirushyk@gmail.com>

	* gst/icydemux/gsticydemux.c:
	  icydemux: avoid timestamp field initialisation for tag event
	  This field is not used and will be removed in 2.0 API.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761462

2014-10-05 15:51:18 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: do not accept header with invalid channel count
	  https://bugzilla.gnome.org/show_bug.cgi?id=737928

2018-05-05 19:27:24 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	* meson_options.txt:
	* sys/v4l2/meson.build:
	  meson: Update option names to omit disable_ and with- prefixes
	  Also yield common options to the outer project (gst-build in our case)
	  so that they don't have to be set manually.

2012-04-12 09:53:24 +0200  Anthony Violo <anthony.violo@ubicast.eu>

	* gst/replaygain/gstrgvolume.c:
	  BugFix : Change peak value to normalize audio file with fallback gain
	  https://bugzilla.gnome.org/show_bug.cgi?id=673970

2018-05-05 16:32:59 +0200  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: don't use buffer lists if everything fits into one buffer
	  People might use very large mtu sizes where every payload
	  fits into a single output packet.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795758

2018-04-04 15:50:55 +0200  Kirill Marinushkin <kmarinushkin@de.adit-jv.com>

	* configure.ac:
	  configure: Fix hard-coded enabled v4l2 probe on Linux/ARM
	  Currently, enable_v4l2_probe is hard-coded to "yes" on linux, platforms
	  arm and aarch64. This even overrides the --disable-v4l2-probe argument.
	  As a result, it is impossible to disable v4l2_probe. It becomes a problem
	  for use-cases, when startup time is critical, because the v4l2_probe
	  feature increases the initialization time.
	  This commit makes the v4l2_probe feature configurable.
	  On linux, platforms arm and aarch64, the default value is still "yes".
	  But now it can be disabled by the --disable-v4l2-probe argument.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795200

2018-04-23 11:26:12 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Don't wake up the muxer unless there is data
	  https://bugzilla.gnome.org/show_bug.cgi?id=795332

2018-04-23 11:19:18 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Save the current position in the output segment
	  https://bugzilla.gnome.org/show_bug.cgi?id=795332

2018-04-19 17:53:51 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/flv/gstflvmux.c:
	* tests/check/elements/flvmux.c:
	  flvmux: Wait for caps from both srcs before writing header
	  Wait for caps on all pads to start writing data even when source is live.
	  Includes unit test by Havard Graff that simulates it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=794722

2018-04-13 13:29:06 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: rely on gst_v4l2_dup() to set no_initial_format and keep_aspect
	  gst_v4l2_dup() will now take care of setting
	  v4l2capture->no_initial_format and keep_aspect instead of doing it
	  manually.
	  Fix a typo as keep_aspect was set twice on v4l2output but never on
	  v4l2capture.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795028

2018-04-24 14:06:10 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/cairo/meson.build:
	* ext/dv/meson.build:
	* ext/flac/meson.build:
	* ext/gdk_pixbuf/meson.build:
	* ext/gtk/meson.build:
	* ext/jack/meson.build:
	* ext/jpeg/meson.build:
	* ext/lame/meson.build:
	* ext/libpng/meson.build:
	* ext/mpg123/meson.build:
	* ext/pulse/meson.build:
	* ext/shout2/meson.build:
	* ext/soup/meson.build:
	* ext/speex/meson.build:
	* ext/taglib/meson.build:
	* ext/twolame/meson.build:
	* ext/vpx/meson.build:
	* ext/wavpack/meson.build:
	* gst/alpha/meson.build:
	* gst/apetag/meson.build:
	* gst/audiofx/meson.build:
	* gst/audioparsers/meson.build:
	* gst/auparse/meson.build:
	* gst/autodetect/meson.build:
	* gst/avi/meson.build:
	* gst/cutter/meson.build:
	* gst/debugutils/meson.build:
	* gst/deinterlace/meson.build:
	* gst/dtmf/meson.build:
	* gst/effectv/meson.build:
	* gst/equalizer/meson.build:
	* gst/flv/meson.build:
	* gst/flx/meson.build:
	* gst/goom/meson.build:
	* gst/goom2k1/meson.build:
	* gst/icydemux/meson.build:
	* gst/id3demux/meson.build:
	* gst/imagefreeze/meson.build:
	* gst/interleave/meson.build:
	* gst/isomp4/meson.build:
	* gst/law/meson.build:
	* gst/level/meson.build:
	* gst/matroska/meson.build:
	* gst/monoscope/meson.build:
	* gst/multifile/meson.build:
	* gst/multipart/meson.build:
	* gst/replaygain/meson.build:
	* gst/rtp/meson.build:
	* gst/rtpmanager/meson.build:
	* gst/rtsp/meson.build:
	* gst/shapewipe/meson.build:
	* gst/smpte/meson.build:
	* gst/spectrum/meson.build:
	* gst/udp/meson.build:
	* gst/videobox/meson.build:
	* gst/videocrop/meson.build:
	* gst/videofilter/meson.build:
	* gst/videomixer/meson.build:
	* gst/wavenc/meson.build:
	* gst/wavparse/meson.build:
	* gst/y4m/meson.build:
	* meson.build:
	* sys/directsound/meson.build:
	* sys/v4l2/meson.build:
	* sys/ximage/meson.build:
	  Meson: Generate pc file for all plugins in good
	  https://bugzilla.gnome.org/show_bug.cgi?id=794568

2018-04-25 10:58:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: use -Wl,-Bsymbolic-functions where supported
	  Just like the autotools build.

2018-04-25 10:37:40 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Read caption from input buffer
	  And not from unallocated output buffer GstMapInfo
	  CID #1435131

2018-02-07 11:00:18 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/gstqtmuxmap.h:
	  isomp4: qtmux: Add Closed Caption support
	  Supports CEA 608 and CEA 708 CC streams
	  Also supports usage in "Robust Prefill" mode if the incoming caption
	  stream is constant (i.e. there is one incoming CC buffer for each
	  video frame).
	  https://bugzilla.gnome.org/show_bug.cgi?id=606643

2018-02-06 15:38:00 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	  isomp4: Make 'gmhd' atom usage more generic
	  Only the 'gmin' atom is required. Any other entry within it are
	  optional.
	  https://bugzilla.gnome.org/show_bug.cgi?id=606643

2018-04-22 10:40:19 -0300  Thibault Saunier <tsaunier@igalia.com>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Accept sof-marker=4
	  sof-marker is 4 when input is in the RGB colorspace.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795463

2018-04-02 16:06:35 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecdec.h:
	* tests/check/elements/rtpulpfec.c:
	  ulpfecdec: output perfect seqnums
	  ULP FEC, as defined in RFC 5109, has the protected and protection
	  packets sharing the same ssrc, and a different payload type, and
	  implies rewriting the seqnums of the protected stream when encoding
	  the protection packets. This has the unfortunate drawback of not
	  being able to tell whether a lost packet was a protection packet.
	  rtpbasedepayload relies on gaps in the seqnums to set the DISCONT
	  flag on buffers it outputs. Before that commit, this created two
	  problems:
	  * The protection packets don't make it as far as the depayloader,
	  which means it will mark buffers as DISCONT every time the previous
	  packets were protected
	  * While we could work around the previous issue by looking at
	  the protection packets ignored and dropped in rtpptdemux, we
	  would still mark buffers as DISCONT when a FEC packet was lost,
	  as we cannot know that it was indeed a FEC packet, even though
	  this should have no impact on the decoding of the stream
	  With this commit, we consider that when using ULPFEC, gaps in
	  the seqnums are not a reliable indicator of whether buffers should
	  be marked as DISCONT or not, and thus rewrite the seqnums on
	  the decoding side as well to form a perfect sequence, this
	  obviously doesn't prevent the jitterbuffer from doing its job
	  as the ulpfec decoder is downstream from it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=794909

2018-04-17 17:57:16 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* tests/examples/rtsp/test-onvif.c:
	  Revert "rtspsrc: Fix up sendonly/recvonly attribute handling"
	  This reverts commit af273b4de9eb292c0b6af63665e10ca015895902.
	  While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
	  the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
	  the opposite, just like the ONVIF standard.
	  Let's follow those RFCs as we're doing RTSP here, and add a property at
	  a later time if needed to switch to the SDP RFC behaviour.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793964

2018-04-16 21:27:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Drain the parser when a CAPS event is received
	  After a CAPS event, in theory a new stream can start and it might start
	  with the FLAC headers again. We can't detect FLAC headers in the middle
	  of the stream, so we drain the parser to be able to detect either FLAC
	  headers after the CAPS event or the continuation of the previous stream.
	  This fixes for example
	  gst-launch-1.0 audiotestsrc num-buffers=200 ! flacenc ! c. \
	  audiotestsrc num-buffers=200 freq=880 ! flacenc ! c. \
	  concat name=c ! rtpgstpay ! udpsink host=127.0.0.1 port=5000
	  gst-launch-1.0 udpsrc multicast-group=127.0.0.1 port=5000 \
	  caps=application/x-rtp,media=application,clock-rate=90000,encoding-name=X-GST ! \
	  rtpgstdepay ! flacparse ! flacdec ! audioconvert ! pulsesin

2018-04-16 10:52:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* README:
	* common:
	  Automatic update of common submodule
	  From 3fa2c9e to ed78bee

2018-04-05 16:05:12 +1000  Matthew Waters <matthew@centricular.com>

	* ext/meson.build:
	* ext/qt/gstqtglutility.cc:
	* ext/qt/meson.build:
	* tests/examples/meson.build:
	* tests/examples/qt/meson.build:
	* tests/examples/qt/qmlsink/CMakeLists.txt:
	* tests/examples/qt/qmlsink/meson.build:
	* tests/examples/qt/qmlsink/play.pro:
	* tests/examples/qt/qmlsink/qmlsink.qrc:
	* tests/examples/qt/qmlsrc/grabqml.pro:
	* tests/examples/qt/qmlsrc/meson.build:
	* tests/examples/qt/qmlsrc/qmlsrc.qrc:
	  meson: add build files for the qml plugin
	  Tested on linux with X11/wayland and semi-tested on Windows.
	  Windows crashes on item destruction however this is better than nothing.
	  Fix up some win32 build issues on the way with mismatched {} and
	  G_STMT_{START,END}

2018-04-13 23:02:26 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/elements/flvmux.c:
	  flvmux test: refactor looped test.
	  Looping the test 500 times to only execute the test once every
	  33 times means we inited and deinited gstreamer 467 times
	  for no reason at all, which was annoying when running the test
	  with valgrind.

2018-04-13 23:01:20 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: unref return of aggregator_pad_peek_buffer
	  We ended up leaking every single buffer going through the
	  muxer, which is far from ideal

2018-04-13 22:49:43 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix leak
	  gst_qt_mux_can_renegotiate () gets called everywhere following
	  that pattern:
	  return gst_qt_mux_can_renegotiate (ref(self));
	  This means the reference must be released both in the success
	  and failure cases, it was only done in the success case.

2018-04-13 22:44:14 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: aggregate should not push EOS itself
	  Instead it is expected to return GST_FLOW_EOS, and let the
	  base class handle that.

2018-04-13 21:19:02 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/gst-plugins-good.supp:
	  valgrind supps: ignore gnutls leaking a certificate
	  After investigating, we do dispose of the TLS connections
	  appropriately in the souphttpsrc test, which in turn
	  calls gnutls_deinit, but certificates get leaked anyway.

2018-04-13 20:35:24 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/elements/souphttpsrc.c:
	  souphttpsrc test: free g_get_current_dir return

2018-04-13 20:31:07 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/gst-plugins-good.supp:
	  valgrind supps: bring getaddrinfo suppression from -base

2018-04-13 20:28:35 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/gst-plugins-good.supp:
	  valgrind supps: ignore more twolame conditional moves

2018-04-13 17:37:47 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/elements/rtpulpfec.c:
	  rtpulpfec tests: Fix leaks

2018-02-16 23:40:50 +0100  Alicia Boya García <aboya@igalia.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Add comment about Opus clipping
	  https://bugzilla.gnome.org/show_bug.cgi?id=793523

2018-04-11 20:28:00 +0000  Whoopie <whoopie79@gmx.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Disable DMABuf for emulated formats
	  libv4l2 does not prevent exporting DMABuf even when emulated formats are
	  in use. As a side effect, userspace ends up with buffers of the original
	  formats which will cause issues.
	  https://bugzilla.gnome.org/show_bug.cgi?id=795097

2018-04-08 20:42:16 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Only use BT2020_12 for BT2020 v4l2 colorspace
	  BT2020_12 is not represented in V4L2, so drivers providing full colority
	  for BT2020 will set V4L2_XFER_FUNC_709 transfer function. To fix the
	  issue, we bump this to BT2020_12 if the resoltion is 4K, but we should
	  only do that if the colorspace is BT2020 to start with, otherwise it's
	  not possible to use normal BT709 for 4K 8bit formats.

2018-04-08 13:43:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Always set the colorimetry in S_FMT
	  So far we were only setting colorimetry for OUTPUT devices (v4l2sink or
	  m2m sink pad). This prevented selecting through caps negotiation the
	  colorimetry for CAPTURE devices (v4l2src or m2m src pad). This is rarely
	  selectable, but trying is harmless.

2018-04-11 21:41:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/monoscope/gstmonoscope.c:
	  monoscope: Only fixate pixel-aspect-ratio if the field exists

2018-04-11 17:54:38 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't send fragment-opened-closed message if the reference ctx is NULL
	  It can happen during teardown that the reference context becomes NULL.
	  In that case, trying to send the fragment-opened-closed message would
	  lead to a crash.

2018-04-11 09:12:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Run gst_iterator_foreach() as long as it returns GST_ITERATOR_RESYNC
	  CID 1434160

2018-04-11 08:51:32 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix comparision for extra caption atom
	  We want to make sure we have *enough* data for the potential 2nd
	  caption atom.
	  CID #1434161

2018-04-11 08:42:54 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Handle bogus caption samples
	  Corrupted files could potentially have multiple cdat/cdt2 atoms in
	  a sample entry, which is unclear how to handle.
	  Ignore repeated ones.
	  CID #1434162
	  CID #1434159

2018-04-10 21:15:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/monoscope/gstmonoscope.c:
	  monoscope: Fixate pixel-aspect-ratio too and make sure the final caps are completely fixated
	  Otherwise e.g. this fails with assertions:
	  gst-launch-1.0 audiotestsrc ! audioconvert ! monoscope ! videoconvert ! \
	  videoscale ! video/x-raw,width=800,height=600 ! ximagesink

2018-03-08 10:10:01 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Add comments and doc about prefill mode

2018-02-06 14:36:50 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Refactor pad re-negotiation code
	  It was similar for all pads
	  https://bugzilla.gnome.org/show_bug.cgi?id=606643

2018-01-31 15:10:03 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Detect and expose CEA 608/708 Closed Caption tracks
	  https://bugzilla.gnome.org/show_bug.cgi?id=606643

2018-04-04 01:48:44 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtxsend: fix wrong memory layout assumption
	  The code responsible for creating retransmitted buffers
	  assumed the stored buffer had been created with
	  rtp_buffer_new_allocate when copying the extension data,
	  which isn't necessarily the case, for example when
	  the rtp buffers come from a udpsrc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=794958

2018-04-02 23:04:06 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: new signal "get-storage"
	  Similar to the get-session and get-internal-session signals,
	  we expose a get-storage signal in addition to the
	  get-internal-storage signal to give access to the actual
	  element for applications that need to set properties on the
	  element, in particular "size-time"
	  https://bugzilla.gnome.org/show_bug.cgi?id=794910

2018-03-29 19:19:21 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Add new reset-muxer property
	  With this the muxer is not set to NULL after each segment but instead
	  only flush events are sent to it to reset the EOS state.
	  As a result, the muxer will keep stream state and e.g. mpegtsmux will
	  keep the packet continuity counter continuous between segments as needed
	  by hlssink2.
	  https://bugzilla.gnome.org/show_bug.cgi?id=794816

2018-04-02 12:48:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/Makefile.am:
	* tests/icles/meson.build:
	* tests/icles/v4l2src-test.c:
	  tests: remove broken and now pointless v4l2src-test
	  This tests APIs that don't exist any longer and also doesn't
	  work at all, and was last touched in a meaningful way in 2006.

2018-03-21 00:19:37 +0900  Seungha Yang <pudding8757@gmail.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Fix unknown type name ‘off_t’ error
	  Fix following build error
	  gstv4l2object.h:197:17: error: unknown type name ‘off_t’
	  gint fd,  off_t offset);
	  ^
	  https://bugzilla.gnome.org/show_bug.cgi?id=794533

2017-05-25 03:44:39 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: reject segment seeks
	  https://bugzilla.gnome.org/show_bug.cgi?id=784681

2018-02-13 11:50:05 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Handle variant of vorbis in mp4
	  Comes from gpac apparently. The codec_data uses the same packing
	  mechanism as matroska.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738244

2018-03-22 15:20:47 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check sample count is valid in PIFF parsing
	  The value stored in cenc_aux_sample_count wasn't in sync with the
	  parsing code that followed which checks whether all entries are
	  valid and present.
	  Only write the actual sample count when we know for sure.
	  CID #1427087

2018-03-04 15:14:08 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>

	* configure.ac:
	* ext/qt/gstqtglutility.cc:
	  qt: Get EGL native display from QPA if platform header is available
	  https://bugzilla.gnome.org/show_bug.cgi?id=792378

2018-03-06 02:14:34 +0100  Petr Kulhavy <brain@jikos.cz>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: switch to using a buffer pool
	  This exposes a new property, mtu, which is used to determine the
	  initial size of buffers from the buffer pool. If received data
	  exceeds this, the element gracefully handles that in a manner similar
	  to what we had previously: a large memory gets filled and reallocated
	  at the next call to "fill".
	  The default size is set to 1500, which should cover most use cases.
	  With contributions from Mathieu Duponchelle <mathieu@centricular.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=772841

2016-11-15 09:39:31 +0100  Petr Kulhavy <brain@jikos.cz>

	* gst/udp/gstudpsrc.h:
	  udpsrc: optimize GstUdpSrc object for cache performance
	  Optimize GstUdpSrc for cache performance.
	  Move the hot properties, which are used by the read function, to the top:
	  @used_socket, @addr, @cancellable, @skip_first_bytes, @timeout,
	  @retrieve_sender_address.
	  Remove the unused property @ttl.
	  Where needed reorder so that holes are avoided (the 64-bit @timeout)
	  https://bugzilla.gnome.org/show_bug.cgi?id=772841

2018-03-05 12:48:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix seeking on streams with frame reordering
	  The samples table is sorted by DTS, not PTS. As such we can only get the
	  correct result when using a binary search on it, if we search for the
	  DTS.
	  Also if we only ever search for the frame, where the following frame is
	  the first one with a PTS after the search position, we will generally
	  stop searching too early if frames are reordered.
	  In forwards playback this is not really a problem (after the decoder
	  reordered the frames, clipping is happening), in reverse playback
	  it means that we can output one or more frames too few as we stop too
	  early and the decoder would never receive it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782118

2018-03-20 11:36:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpredenc.c:
	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/rtp/rtpstoragestream.c:
	* tests/check/elements/rtpred.c:
	* tests/check/elements/rtpulpfec.c:
	  rtp: Fix compilation with non-C99 compilers
	  By moving variable declarations out of loop headers.

2018-03-20 09:24:19 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* meson.build:
	  Back to development

=== release 1.14.0 ===

2018-03-19 20:18:22 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.14.0

2018-03-19 20:18:22 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  Update docs

2018-03-19 18:39:08 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpulpfecdec.c:
	  rtpulpfecdec: fix build with older gcc
	  As on Ubuntu Trusty.
	  https://bugzilla.gnome.org/show_bug.cgi?id=794493

2018-03-19 10:58:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Allow splitting at exactly the time/bytes threshold
	  76e458a119926424e9dd5acf3210a592a314d713 changed the conditions from
	  "queued > threshold" to "queued >= threshold", which broke hlssink2 and
	  resulting in too small fragments being created although keyframes would
	  be at *exactly* the configured threshold.
	  https://bugzilla.gnome.org/show_bug.cgi?id=794440

2018-03-17 20:29:35 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/rtpulpfeccommon.h:
	  rtpulpfec: fix unconditional use of __attribute__ ((packed))
	  Fix compilation with MSVC. We still assume that attribute
	  is supported by all other relevant compilers, which seems
	  to be the case since we haven't had any complaints about
	  similar code in rtpsbcpay.

2018-03-17 13:04:47 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/rtp/rtpulpfeccommon.c:
	  rtpulpfec: don't use non-portable notation for 64-bit int constants
	  Use GLib macro instead, even if it's a bit unwieldy.

2018-03-17 12:55:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpulpfecdec.c:
	  rtpulpfecdec: don't use __builtin_ctzll unconditionally
	  Fixes build with MSVC, and possibly other compilers too.

=== release 1.13.91 ===

2018-03-13 19:16:42 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.13.91

2018-03-13 19:16:42 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  Update docs

2018-03-12 13:21:08 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  docs: rtpbin: add some Since markers for new properties

2018-03-10 18:57:38 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/directsound/meson.build:
	  meson: Add deviceprovider changes to directsoundsink
	  These were missed when they were added to Makefile.am

2018-03-08 10:12:16 +0100  Michael Tretter <m.tretter@pengutronix.de>

	* configure.ac:
	  configure.ac: enable largefile support if possible
	  https://bugzilla.gnome.org/show_bug.cgi?id=793103

2018-03-07 14:16:02 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Fix support for 32bit mmap
	  https://bugzilla.gnome.org/show_bug.cgi?id=793103

=== release 1.13.90 ===

2018-03-03 22:19:36 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.13.90

2018-03-03 22:19:36 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  Update docs

2018-03-01 18:24:33 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/flv/gstflvmux.c:
	* tests/check/elements/flvmux.c:
	  flvmux: Duration & unit tests
	  The muxed buffers will not carry the duration of the
	  incoming buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793457

2018-03-01 17:15:02 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Set PTS based on running time
	  https://bugzilla.gnome.org/show_bug.cgi?id=793457

2018-03-01 18:13:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Ignore sendonly/recvonly attributes unless a backchannel is configured
	  This works around a bug in various ONVIF cameras that implement the
	  attributes the wrong way around. They still won't work with a
	  backchannel but at least normal playback will work for the time being.
	  It restores pre-1.14 behaviour where we would fail to preroll on any SDP
	  that lists a recvonly stream. For 1.16 a better solution should be
	  found.
	  The problem here is that the ONVIF spec has the meaning of the two
	  attributes the wrong way around in the examples, compared to RFC4566.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793715

2018-03-01 18:16:24 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* meson.build:
	  meson: enable more warnings
	  https://bugzilla.gnome.org/show_bug.cgi?id=793961

2018-03-01 00:34:20 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/flv/gstflvmux.c:
	  Port to latest GstAggregator segment API
	  The aggregator segment is now exposed on the src pad
	  https://bugzilla.gnome.org/show_bug.cgi?id=793945

2018-03-01 15:34:13 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/directsound/gstdirectsoundplugin.c:
	  directsoundsink: Downgrade rank to match directsoundsrc in -bad
	  As stated in commit c2956036b8da4b8f22a63a4f5a254be03e870aa6 in -bad,
	  the wasapi elements are now better than directsound, and should be
	  preferred if they are available.
	  For a later release, once the elements have more testing, we can
	  consider moving them to -good.

2018-02-28 19:21:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Only mark new clusters as keyframe if they start on a keyframe or we're muxing only audio
	  Based on a patch by Nicola Murino <nicola.murino@gmail.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=792775

2018-02-28 19:19:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Clip maximum cluster duration to the maximum possible value
	  Only up to timescale * G_MAXINT16 is possible as cluster duration, which
	  is already higher than our default value. Using higher values would
	  cause overflows and broken files.
	  Based on the investigation by Nicola Murino <nicola.murino@gmail.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=792775

2018-02-26 13:03:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Refuse caps changes after starting to write headers
	  Matroska does not support changing the stream type and stream properties
	  after the headers were started to be written, and for example H264
	  codec_data changes can't be supported.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782949

2018-02-27 16:33:53 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/elements/rtpred.c:
	  tests: fix redenc tests
	  The default of the allow-no-red-blocks property was changed in a
	  previous commit, thus breaking the test assumptions

2018-02-27 13:13:49 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/rtpulpfeccommon.c:
	  rtp: fix another debug log printf format warning on 32-bit systems
	  rtpulpfeccommon.c:432:27: error: format ‘%lx’ expects argument of type
	  ‘long unsigned int’, but argument 10 has type ‘guint64 {aka long long unsigned int}’
	  https://bugzilla.gnome.org/show_bug.cgi?id=793732

2018-02-26 17:02:52 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: provide example usage for ignored-payload-types

2018-02-26 16:53:08 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpbin, rtpptdemux: Add missing Since markers

2018-02-26 15:57:28 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpredenc.c:
	* gst/rtp/gstrtpstorage.c:
	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/rtp/gstrtpulpfecenc.h:
	  FEC elements: document, remove irrelevant properties
	  The ulpfecenc "mux-seq" and "ssrc" properties were initially added
	  because the element did more than implement ULPFEC. As it was
	  decided that FLEXFEC would be implemented in a separate element,
	  both properties are now unneeded and confusing.
	  Change the default for the ulpfecenc multi-packet property,
	  as it is expected that most users of this element will be protecting video
	  streams.
	  Change the default property for the rtpredenc allow-no-red-blocks
	  property, as it should also be its default mode of operation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793843

2018-02-24 20:05:05 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: do not warn when caps were not yet received
	  It is expected that when connecting to a stream that has
	  already started, the caps will only arrive at the interval
	  specified on rtpgstpay, we shouldn't be warning as this is
	  a normal mode of operation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793798

2018-02-22 21:53:40 +0100  Arnaud Bonatti <arnaud.bonatti@gmail.com>

	* gst/rtp/gstrtpulpfecdec.c:
	  rtpulpfec: fix debug log printf format warning on 32-bit platforms
	  https://bugzilla.gnome.org/show_bug.cgi?id=793732

2018-02-22 14:58:12 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-rtp.xml:
	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpredenc.c:
	* gst/rtp/gstrtpstorage.c:
	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	  docs: hook up new RTP FEC elements
	  https://bugzilla.gnome.org/show_bug.cgi?id=792696

2018-02-22 14:57:58 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update for git master

2018-02-22 10:54:02 +0000  Tim-Philipp Müller <tim@centricular.com>

	* .gitignore:
	* tests/check/elements/.gitignore:
	  .gitignore more test binaries

2018-02-21 20:46:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	  tests: also dist new fec test header file

2018-02-21 20:44:26 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/Makefile.am:
	  rtp: dist new header files
	  Fixes make distcheck

2018-02-21 18:52:44 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpstorage.c:
	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/rtp/rtpulpfeccommon.c:
	* gst/rtp/rtpulpfeccommon.h:
	  rtp: fec: fix build with gstreamer debug log system disabled

2018-02-21 19:59:04 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: do no assume sink caps are non NULL

2018-02-21 18:51:17 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* tests/check/Makefile.am:
	  check: Fix ulpfec test build
	  The test name was updated but not the build definition

2017-11-28 06:02:05 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: Expose FEC support signals
	  Also slightly refactor complete_session_src
	  https://bugzilla.gnome.org/show_bug.cgi?id=792696

2017-11-17 03:52:03 +0100  Mikhail Fludkov <misha@pexip.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpreddec.c:
	* gst/rtp/gstrtpreddec.h:
	* gst/rtp/gstrtpredenc.c:
	* gst/rtp/gstrtpredenc.h:
	* gst/rtp/gstrtpstorage.c:
	* gst/rtp/gstrtpstorage.h:
	* gst/rtp/gstrtpulpfecdec.c:
	* gst/rtp/gstrtpulpfecdec.h:
	* gst/rtp/gstrtpulpfecenc.c:
	* gst/rtp/gstrtpulpfecenc.h:
	* gst/rtp/meson.build:
	* gst/rtp/rtpredcommon.c:
	* gst/rtp/rtpredcommon.h:
	* gst/rtp/rtpstorage.c:
	* gst/rtp/rtpstorage.h:
	* gst/rtp/rtpstoragestream.c:
	* gst/rtp/rtpstoragestream.h:
	* gst/rtp/rtpulpfeccommon.c:
	* gst/rtp/rtpulpfeccommon.h:
	* tests/check/Makefile.am:
	* tests/check/elements/packets.h:
	* tests/check/elements/rtpred.c:
	* tests/check/elements/rtpstorage.c:
	* tests/check/elements/rtpulpfec.c:
	* tests/check/meson.build:
	  rtp: Implement ULPFEC (RFC 5109)
	  We expose a set of new elements:
	  * ULPFEC encoder / decoder
	  * A storage element, which should be placed before jitterbuffers,
	  and is used to store packets in order to attempt reconstruction
	  after the jitterbuffer has sent PacketLost events
	  * RED encoder / decoder (RFC 2198), these are necessary to
	  use FEC in webrtc, as browsers will propose and expect ulpfec
	  packets to be wrapped in red packets
	  With contributions from:
	  Mathieu Duponchelle <mathieu@centricular.com>
	  Sebastian Dröge <sebastian@centricular.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=792696

2017-11-28 01:11:54 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpptdemux.h:
	  rtpptdemux: Add ignored-payload-types property
	  Packets with these payload types will be dropped. A use case
	  for this is FEC, where we want FEC packets to go through the
	  jitterbuffer, but not be output by rtpbin.
	  https://bugzilla.gnome.org/show_bug.cgi?id=792696

2017-11-20 18:08:38 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: Add ssrc to output caps
	  It may be useful downstream
	  https://bugzilla.gnome.org/show_bug.cgi?id=792696

2018-02-21 11:12:10 +0100  Arnaud Bonatti <arnaud.bonatti@gmail.com>

	* ext/gtk/gstgtkbasesink.c:
	  gtk: fix compiler warning with recent glib
	  https://bugzilla.gnome.org/show_bug.cgi?id=793688

2018-02-21 11:35:33 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qt: don't use libEGL functions when we don't link to libEGL
	  Use the provided wrapper available from libgstgl.
	  https://bugzilla.gnome.org/show_bug.cgi?id=793547

2018-02-18 21:38:13 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/monoscope/gstmonoscope.c:
	* gst/monoscope/gstmonoscope.h:
	  monoscope: Forward the SEGMENT event from the chain function
	  Otherwise we'll break the event order and forward the SEGMENT event
	  before sending a CAPS event.

2018-02-16 12:25:29 +0000  James Stevenson <james@stev.org>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix missing read property of backchannel
	  Add missing read property code for backchannel
	  https://bugzilla.gnome.org/show_bug.cgi?id=793507

2018-02-16 09:42:59 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/rtsp/meson.build:
	  examples: rtsp: fix meson build take 2

2018-02-16 11:30:01 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/rtsp/meson.build:
	  rtsp: Fix meson.build of the example

2018-01-26 16:33:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Implement ONVIF backchannel support via TCP

2017-10-13 18:05:54 +0300  Nirbheek Chauhan <nirbheek@centricular.com>

	* configure.ac:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	* tests/examples/Makefile.am:
	* tests/examples/meson.build:
	* tests/examples/rtsp/Makefile.am:
	* tests/examples/rtsp/meson.build:
	* tests/examples/rtsp/test-onvif.c:
	  rtspsrc: Implement ONVIF backchannel support
	  Set backchannel=onvif to enable, and use the 'push-backchannel-sample'
	  action signal with the correct stream id.

2018-02-16 01:49:57 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multifile/gstsplitmuxsrc.h:
	  splitmuxsrc: Improve not-linked handling.
	  Don't report not-linked unless all pads have
	  returned not-linked.

2018-02-15 19:44:19 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* meson.build:
	  Back to development

=== release 1.13.1 ===

2018-02-15 17:06:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* configure.ac:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.13.1

2018-02-15 17:05:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gtk.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-lame.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mpg123.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-twolame.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update plugin docs

2018-02-15 13:32:20 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/bg.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/fr.po:
	* po/hr.po:
	* po/hu.po:
	* po/nb.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: update translations

2018-02-14 16:38:07 +0100  Patrick Radizi <patrickr@axis.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: allow timestamps to move backwards
	  The original solution for #784002 incorrectly assumed that timestamps
	  may not move backwards and changed timestamps that did so.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784002

2018-02-15 00:58:38 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/lame/gstlamemp3enc.c:
	* gst/flv/gstindex.c:
	* sys/v4l2/gstv4l2src.c:
	  docs: remove pointless Since: 0.10.x markers

2017-09-27 16:01:35 +0200  Alban Bedel <alban.bedel@avionic-design.de>

	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbisdepay: fix unbounded memory usage
	  All received configurations are parsed and added to a list, this lead
	  to an unbounded memory usage. As the configuration is resent every
	  second this quickly lead to a large memory usage.
	  Add a check to only add the config if it is not already available in
	  the list. This fix only handle the typical case of a well behaved
	  stream, a malicious server could still send many useless
	  configurations to raise the client memory usage.

2018-02-12 18:41:41 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-qmlgl.xml:
	  docs: add qt plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=754094

2018-02-12 18:34:16 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* ext/Makefile.am:
	* ext/meson.build:
	* tests/examples/meson.build:
	  qt: hook up to build
	  https://bugzilla.gnome.org/show_bug.cgi?id=754094

2018-02-12 18:13:17 +0000  Tim-Philipp Müller <tim@centricular.com>

	  Move qt plugin from -bad
	  https://bugzilla.gnome.org/show_bug.cgi?id=754094

2018-02-12 15:44:35 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	  configure: fix build with --disable-external

2018-02-10 20:31:49 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-gtk.xml:
	  docs: add moved gtk plugin to docs

2018-02-10 20:28:46 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	  docs: update for git master

2018-02-12 11:02:12 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/gtk/meson.build:
	* ext/meson.build:
	* meson.build:
	* tests/examples/meson.build:
	  gtk: hook up to meson build

2018-02-10 13:20:43 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* ext/Makefile.am:
	* ext/gtk/Makefile.am:
	* tests/examples/Makefile.am:
	* tests/examples/gtk/.gitignore:
	* tests/examples/gtk/Makefile.am:
	  gtk: hook up to autotools build

2018-02-10 12:49:36 +0000  Tim-Philipp Müller <tim@centricular.com>

	  Move gtk plugin from -bad
	  https://bugzilla.gnome.org/show_bug.cgi?id=754094

2018-02-09 11:26:56 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix comment typo in previous commit

2018-02-09 11:20:38 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: More 'meta' atom parsing fixes
	  Turns out everybody is doing it their own way, so peek into the
	  meta atom itself to figure out which spec it is following

2018-02-02 13:51:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Add support for muxing svmi atom for stereoscopic video information
	  https://bugzilla.gnome.org/show_bug.cgi?id=793120

2018-02-09 08:59:56 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Minor cleanup
	  Just move variables to the blocks where they are used.
	  That function is massive, could do with some splitting up for
	  readability :(

2018-02-09 08:54:05 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Cope with difference between QTFF and ISO BMFF specs
	  The 'meta' atom is defined differently in QTFF and BMFF, so try
	  to guess which spec the current stream applies to by looking
	  at the major file type.

2018-02-09 08:35:52 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux_dump.c:
	  isomp4: Make 'hdlr' atom dump more flexible
	  The smallest possible is 24 (and not 25) bytes.
	  The last "name" field can according to QTFF specifications not be present
	  at all. The parser will handle this fine and so will the rest of
	  the qtdemux code.

2018-02-09 08:35:25 +0100  Edward Hervey <edward@centricular.com>

	* gst/audiofx/audiopanoramaorc-dist.c:
	* gst/deinterlace/tvtime-dist.c:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videomixer/videomixerorc-dist.c:
	  Update ORC files

2018-02-08 19:09:45 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: make version numbers ints and fix int/string comparison
	  WARNING: Trying to compare values of different types (str, int).
	  The result of this is undefined and will become a hard error
	  in a future Meson release.

2017-10-01 18:21:26 +0200  Jérôme Laheurte <jerome@jeromelaheurte.net>

	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix build on macOS versions < 12.0
	  Use value instead of version macro when testing for mac OS version,
	  since the define for the newer version may not be defined when
	  compiling against older versions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=788402

2018-02-07 20:15:00 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qt: don't #include platform specific gstglcontext_*.h headers
	  They aren't public headers

2018-02-04 11:47:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* tests/check/Makefile.am:
	  autotools: use -fno-strict-aliasing where supported
	  https://bugzilla.gnome.org/show_bug.cgi?id=769183

2017-12-04 20:12:40 +0900  Justin Kim <justin.kim@collabora.com>

	* gst/isomp4/gstqtmux.c:
	* gst/multifile/gstsplitmuxsink.c:
	  qtmux: send stream warning when refusing video caps
	  If codec_data is changed, the stream is no longer valid.
	  Rather than keeping running when refusing new caps,
	  this patch send a warning  to the bus.
	  Also fix up splitmuxsink to ignore this warning while changing caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790000

2017-11-29 21:30:11 +0900  Justin Kim <justin.kim@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: update output caps regardless format
	  `codec_data` should be transfered if any information of
	  SPS/PPS is changed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790000

2018-01-31 19:11:16 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_types.c:
	  isomp4: Add gmhd/gmin debugging
	  * gmhd is a container, mark it as such so we can see/dump
	  what is contained within
	  * Add dumping for the Base Media Information atom (gmin)

2015-09-23 10:01:32 +0200  Matthieu Crapet <mcrapet@gmail.com>

	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpegenc: add snapshot property
	  Like pngenc, automatically send an EOS message.
	  Example of bin:
	  appsrc ! jpegenc snapshot=true ! filesink location=out.jpg
	  This is especially useful for limited/slow hardware.
	  Otherwise calling gst_video_convert_sample() is a better option
	  (internally uses videoconvert and videoscale).
	  https://bugzilla.gnome.org/show_bug.cgi?id=755453

2018-01-31 15:02:50 +0000  Philippe Normand <philn@igalia.com>

	* gst/interleave/interleave.c:
	  interleave: fix memory leak of GAP buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=793067

2018-01-31 11:38:35 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux_dump: Demote verbose logging to TRACE level

2018-01-31 11:22:23 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux: Re-enable full debug logging of stsz entries
	  No idea why it was disabled (was the case since 2007)

2018-01-30 20:34:32 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/taglib/meson.build:
	* meson.build:
	  meson: use -fno-strict-aliasing where supported
	  https://bugzilla.gnome.org/show_bug.cgi?id=769183

2017-12-12 00:14:02 +0900  Seungha Yang <pudding8757@gmail.com>

	* gst/isomp4/qtdemux.h:
	  qtdemux: Remove white space at end of line
	  https://bugzilla.gnome.org/show_bug.cgi?id=791483

2017-12-12 00:11:24 +0900  Seungha Yang <pudding8757@gmail.com>

	* gst/isomp4/Makefile.am:
	* gst/isomp4/gstisoff.c:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux_debug.h:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_lang.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Apply qtdemux debug category to gstisoff
	  .. instead of the use of default debug category.
	  And, make new header to declare the debug category
	  https://bugzilla.gnome.org/show_bug.cgi?id=791483

2018-01-25 00:46:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: properly set total duration on outgoing segment
	  We would accidentally pass through the duration value from the
	  demuxer from a single fragment, which causes problems when
	  feeding the stream from splitmuxsrc to rtsp-server. Streaming
	  would stop after one fragment due to that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=792861

2018-01-25 00:42:52 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: don't respond to duration query with CLOCK_TIME_NONE
	  total_duration is initialised to CLOCK_TIME_NONE, not 0, so check
	  for that as well in order not to return an invalid duration to
	  a duration query. Doesn't fix anything particular observed in
	  practice, just seemed inconsistent.

2018-01-25 20:48:42 +0100  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add more prose to the comment of gst_qtdemux_find_sample()
	  https://bugzilla.gnome.org/show_bug.cgi?id=792910

2011-02-09 12:48:00 +0000  Oleksij Rempel <linux@rempel-privat.de>

	* ext/vpx/gstvpxdec.c:
	  vpx: add VP8_DEBUG_TXT_* flags for postprocessing
	  https://bugzilla.gnome.org/show_bug.cgi?id=641399

2018-01-25 21:22:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: Add missing \ in multi-line #define

2018-01-22 15:07:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* sys/directsound/Makefile.am:
	* sys/directsound/gstdirectsounddevice.c:
	* sys/directsound/gstdirectsounddevice.h:
	* sys/directsound/gstdirectsoundplugin.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: Add support for a DeviceProvider
	  https://bugzilla.gnome.org/show_bug.cgi?id=792782

2018-01-23 18:37:09 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: fix up uri handler a little
	  Fix path escaping when creating URI from location in get_uri().
	  Return FALSE with an error when URI can't be parsed in set_uri().
	  https://bugzilla.gnome.org/show_bug.cgi?id=783581

2017-06-15 13:37:28 +0200  Dimitrios Katsaros <patcherwork@gmail.com>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: implement uri handler
	  With this patch we can now provide a set of files
	  created by multifilesink as a source for uri elements.
	  e.g. gst-launch-1.0 playbin uri=multifile://img%25d.ppm
	  Note that for the %d pattern you need to replace % with %25.
	  This is to be compliant with URL naming standards.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783581

2018-01-19 15:05:26 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Make sure timecode uses the same timescale as video
	  Don't blindly derive it from the frame rate, but try to get the per-pad
	  configured timescale first (if it exists)
	  https://bugzilla.gnome.org/show_bug.cgi?id=792680

2018-01-18 18:36:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Allow configuring trak timescale per pad/trak
	  It generally makes not much sense to configure it for all pads/traks at
	  once as this value is usually different for each of them. As such, add a
	  new property on the pads in addition to the existing property on the
	  whole muxer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=792649

2018-01-23 09:46:32 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvmux.c:
	  Update for renamed aggregator pad API
	  https://bugzilla.gnome.org/show_bug.cgi?id=791204

2018-01-22 12:24:18 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix up sendonly/recvonly attribute handling
	  We can't handle recvonly streams, sendonly streams are perfectly fine.
	  The direction is the one from the point of view of the SDP offerer
	  (i.e. the RTSP server), and a recvonly stream would be one where the
	  server expects us to send media.
	  RFC 3264, section 5.1:
	  If the offerer wishes to only send media on a stream to its peer, it
	  MUST mark the stream as sendonly with the "a=sendonly" attribute.
	  This is mixed up in the ONVIF streaming specification examples, but
	  actual implementations and conformance tools seem to not care at all
	  about the attributes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=792376

2017-11-11 13:49:22 +0900  paul.kim <paul.hyunil@lge.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Reset retry_count to 0 when GST_FLOW_FLUSHING
	  If a lot of seek method is called very quickly, sometimes data reading
	  and do_request occurs while seek flush event is occurring and error
	  occurs because retry_count
	  reaches to the max. Thus, reset retry_count if flush occurs after
	  do_request and read_buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790199

2018-01-18 15:09:04 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* tests/check/elements/aacparse.c:
	  tests: aacparser: Test that short raw frames don't get concatenated
	  https://bugzilla.gnome.org/show_bug.cgi?id=792644

2018-01-18 14:23:07 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: When parsing raw input, accept frames of any size
	  Raw AAC streams might have very small frames, e.g. 6 byte frames
	  when encoding silence. These frames are then smaller than aacparse's
	  default min_frame_size of 10 bytes (ADTS_MAX_SIZE).
	  When passthrough is disabled or aacparse has to output ADTS, GstBaseParse
	  will concatenate these short frames to the following frame before
	  handling them to aacparse, which processes each input buffer as a single
	  frame, producing bad output.
	  To avoid this problem, set the min_frame_size to 1 when receiving a raw
	  stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=792644

2017-05-02 21:24:06 -0300  Adrián Pardini <github@tangopardo.com.ar>

	* ext/shout2/gstshout2.c:
	  shout2send: print actual username in debug log out
	  https://bugzilla.gnome.org/show_bug.cgi?id=782093

2018-01-15 18:13:37 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* tests/check/elements/rtpbin.c:
	  rtpbin: fix leak of elements requested by signals
	  When the signal returns a floating reference, as its return type
	  is transfer full, we need to sink it ourselves before passing
	  it to gst_bin_add (which is transfer floating).
	  This allows us to unref it in bin_remove_element later on, and
	  thus to also release the reference we now own if the signal
	  returns a non-floating reference as well.
	  As we now still hold a reference to the element when removing it,
	  we also need to lock its state and setting it to NULL before
	  unreffing it
	  Also update the request_aux_sender test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=792543

2018-01-17 11:10:37 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix division by 0 for complex video formats
	  So complex video formats have 0 as pstride. Don't try to divide the
	  stride in such cases.
	  https://bugzilla.gnome.org/show_bug.cgi?id=792596

2018-01-17 11:08:25 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: display stride and width values if stride is too small
	  https://bugzilla.gnome.org/show_bug.cgi?id=792596

2018-01-16 13:19:29 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: document unit of "max-file-duration" property

2018-01-12 12:21:37 +0100  Florent Thiéry <florent.thiery@ubicast.eu>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix typo in documentation
	  https://bugzilla.gnome.org/show_bug.cgi?id=792458

2018-01-12 09:53:37 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: add property set/get PROP_CAPTURE_IO_MODE error handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=791841

2018-01-12 09:46:30 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: fold property set/get PROP_OUTPUT_IO_MODE case into default
	  https://bugzilla.gnome.org/show_bug.cgi?id=791841

2018-01-12 09:49:14 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: add property set/get PROP_CAPTURE_IO_MODE error handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=791841

2018-01-12 09:44:03 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: fold property set/get PROP_OUTPUT_IO_MODE case into default
	  https://bugzilla.gnome.org/show_bug.cgi?id=791841

2018-01-11 10:44:18 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: fix capture-io-mode property get
	  https://bugzilla.gnome.org/show_bug.cgi?id=791841

2018-01-11 17:47:39 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Maintain downstream caps order
	  The g_list_insert_sorted() will behave like prepend when the compare
	  function returns 0. In our case, we want to maintain the order hence
	  append. This fixes this issue and improve the sorting algorithm to make
	  a 10x10 prefered over 10x200 with a preference of 10x8 (and similar
	  cases which was badly handled). This fixes generally fixes issue were a
	  sub-optimal format / size is picked.
	  https://bugzilla.gnome.org/show_bug.cgi?id=792435

2017-12-21 23:02:30 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Also re-enabled paused task
	  When we only run _finish(), the task is never stopped externally,
	  instead it's only paused from the inside. We still want to restart
	  it in this case.

2018-01-08 15:23:24 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: flush flac decoder on lost sync.
	  This to allow the decoder to start searching for a new
	  frame again.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791473

2017-12-21 22:56:51 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Call stop on object before renegotiation
	  Otherwise renegotiation fails as we are still streaming.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791338

2017-12-21 22:55:49 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Remove dead code
	  gst_v4l2_object_stop() will free and nullify the pool, so the
	  following if will never be true.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791338

2017-12-21 22:29:06 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Delay capture pool activation
	  This is support CODA driver which prevents setting the output format if
	  the capture is streaming.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791338

2017-12-13 20:23:46 +0000  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Add dynamic resolution change support
	  This implements a "big hammer" reallocation method. We effectively
	  drain and stop both side of the decoder and restart. This though is
	  the most generic method. This change should enable on most drivers
	  adaptive streaming.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752962

2017-12-30 01:52:13 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: zlib is not actually a hard requirement

2017-09-28 18:00:38 -0300  Ezequiel Garcia <ezequiel@vanguardiasur.com.ar>

	* ext/jpeg/gstjpegdec.c:
	  jpeg: Fixup frames without an EOI marker
	  Some cameras fail to send an end-of-image marker (EOI)
	  and can't be properly decoded by either JPEG or libjpeg.
	  This commit parses the frame, making sure it has an EOI.
	  If there isn't one, the EOI gets added to the buffer.
	  A similar fixup is done in the rtpjpegdepay element,
	  and it makes sense to do it in jpegdec as well.
	  Signed-off-by: Ezequiel Garcia <ezequiel@vanguardiasur.com.ar>
	  https://bugzilla.gnome.org/show_bug.cgi?id=791988

2017-12-26 13:50:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: skip translations if gettext is not available

2017-12-24 13:14:06 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-rtp.xml:
	  docs: add rtpL8pay/depay to docs

2017-12-24 13:11:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	  docs: update for recent changes

2015-05-15 17:00:26 +0100  Tim Allen <tim.allen@ge.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpL8depay.c:
	* gst/rtp/gstrtpL8depay.h:
	* gst/rtp/gstrtpL8pay.c:
	* gst/rtp/gstrtpL8pay.h:
	* gst/rtp/meson.build:
	  rtp: add L8 audio support

2017-12-23 12:45:17 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix typo in multicast join error message

2017-12-23 12:44:31 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: also proxy multicast-iface property to RTCP udpsrc

2015-11-02 00:41:28 +0100  Sebastian Rasmussen <sebrn@hotmail.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: don't try to set IPV6_TCLASS on IPV4 sockets
	  Avoids ERROR log message.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757449

2015-11-02 00:41:28 +0100  Sebastian Rasmussen <sebrn@hotmail.com>

	* tests/check/Makefile.am:
	* tests/check/elements/udpsink.c:
	  tests: udpsink: add check that sets QoS on IPv4/6 sockets
	  https://bugzilla.gnome.org/show_bug.cgi?id=757449

2017-12-22 10:21:28 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2deviceprovider: Don't do slow probes
	  This is problematic in the current design at it seriously slow down
	  startup of applications. As of now, no known application uses the
	  colorimetry and the interlace-modes for anything (the two fields that
	  won't be probed). So let's disable it, in the long term we'll try and
	  find a way to interact with the provider so applicaiton could opt-in
	  these slow probing methods for more advance configuration.

2017-12-22 10:15:48 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't redefine mmap64
	  On Linux, there exist a case where mmap64 is already a define to mmap,
	  so avoid the redefine warning here.

2017-12-19 17:37:58 +0800  Ting-Wei Lan <lantw@src.gnome.org>

	* configure.ac:
	* meson.build:
	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't use mmap64 if off_t is 64-bit
	  The difference between mmap and mmap64 is the type of 'offset' argument.
	  mmap64 always uses a 64-bit interger as offset, while mmap uses off_t,
	  whose size can vary on different operating systems or architectures.
	  However, not all operating systems support mmap64. Fortunately, although
	  FreeBSD only has mmap, its off_t is always 64-bit regardless of
	  architectures, so we can simply use mmap when sizeof(off_t) == 8.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791779

2017-12-22 09:17:04 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  Revert "v4l2object: Use mmap64 to match libv4l2 signature"
	  This reverts commit b61bba48488c0a627d90f04cc9917d8c4f3f0d9b.

2017-12-19 17:37:58 +0800  Ting-Wei Lan <lantw@src.gnome.org>

	* configure.ac:
	* meson.build:
	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Check for mmap64 before using it
	  mmap64 is not available on FreeBSD.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791779

2017-12-20 15:23:26 -0500  Vincent Penquerc'h <vincent.penquerch@collabora.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flv: flvmux ported to the GstAggregator
	  This makes it possible to create a flv file from a live source and not stop
	  when there are packet drops.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782920

2017-12-19 16:47:52 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Call gst_base_sink_wait_preroll on unlock
	  This means that packets will not be lost on fast pause/playing cycles.
	  Also refactor the code a little to simplify it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774945

2017-12-19 16:22:52 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/examples/gtk/Makefile.am:
	  gtk example: Fix cflags in Makefile.am

2017-12-19 15:46:52 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Remove unused variable

2017-12-19 13:03:28 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: don't include uninstalled header

2017-12-17 20:54:06 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/qt/Makefile.am:
	  gl: update plugins to use GstGL from -base

2017-12-17 20:54:06 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/gtk/Makefile.am:
	* ext/gtk/meson.build:
	* tests/examples/gtk/Makefile.am:
	  gl: update plugins to use GstGL from -base

2017-12-19 11:57:52 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix two leaks
	  * gst_event_new_stream_start() does not take ownership of the stream_id
	  * the pipeline_request_id string that is created was not being freed

2017-12-07 22:08:42 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: Add GstVideoCropMeta support
	  If downstream supports this meta, it will add or update it from
	  the GstBuffer in-place rather then copying.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791453

2017-12-13 09:22:17 +0000  Sean DuBois <sean@siobud.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/webm-mux.c:
	  Add AV1 to matroska plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=784160

2017-12-15 14:48:09 +0100  fengalin <fengalin@free.fr>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-read-common.c:
	* tests/check/elements/matroskademux.c:
	* tests/check/elements/matroskamux.c:
	  matroska: fix memory leaks due to toc related updates
	  https://bugzilla.gnome.org/show_bug.cgi?id=790686

2017-12-15 11:40:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/matroskamux.c:
	  matroskamux: Fix various memory leaks in the unit test
	  https://bugzilla.gnome.org/show_bug.cgi?id=790686

2017-12-14 19:05:36 +0100  fengalin <fengalin@free.fr>

	* tests/check/elements/matroskademux.c:
	* tests/check/elements/matroskamux.c:
	  matroska-mux: migrate test to gst_harness
	  ... following the guide lines from Håvard Graff (see https://gstconf.ubicast.tv/videos/moar-better-tests/).
	  https://bugzilla.gnome.org/show_bug.cgi?id=790686

2017-12-01 18:17:06 +0100  fengalin <fengalin@free.fr>

	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	* tests/check/elements/matroskademux.c:
	* tests/check/elements/matroskamux.c:
	  matroska: re-activate and update TOC support
	  TOC support in mastroskamux has been deactivated for a couple of years. This commit updates it to recent GstToc evolutions and introduces toc unit tests for both matroska-mux and matroska-demux.
	  There are two UIDs for Chapters in Matroska's specifications:
	  - The ChapterUID is a mandatory unsigned integer which internally refers to a given chapter. Except for title & language which use dedicated fields, this UID can also be used to add tags to the Chapter. The tags come in a separate section of the container.
	  - The ChapterStringUID is an optional UTF-8 string which also uniquely refers to a chapter but from an external perspective. It can act as a "WebVTT cue identifier" which "can be used to reference a specific cue, for example from script or CSS".
	  During muxing, the ChapterUID is generated and checked for unicity, while the ChapterStringUID receives the user defined UID. In order to be able to refer to chapters from the tags section, we maintain an internal Toc tree with the generated ChapterUID.
	  When demuxing, the ChapterStringUIDs (if available) are assigned to the GstTocEntries UIDs and an internal toc mimicking the toc is used to keep track of the ChapterUIDs and match the tags with the appropriate GstTocEntries.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790686

2017-12-14 18:28:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/v4l2/v4l2src-renegotiate.c:
	  v4l2src: Fix compiler error in example caused by re-declaring `index`
	  ../tests/examples/v4l2/v4l2src-renegotiate.c:57:13: error: ‘index’ redeclared as different kind of symbol
	  static gint index = 0;
	  ^

2017-12-14 14:49:01 +1100  Matthew Waters <matthew@centricular.com>

	* common:
	  Automatic update of common submodule
	  From e8c7a71 to 3fa2c9e

2017-12-13 14:39:47 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2object: Use a debug object for tracing
	  This way we can pass the pad name instead of the element for tracing
	  which helps identifying which v4l2object is used withing M2M element
	  like decoder, encoder and transform. For the reference, pads are name
	  <parent-name>:<pad-name>.

2017-12-13 12:06:21 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Push a GAP event if there's a second *or more*
	  And not "more than a second"

2017-12-13 11:35:37 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't push GAP event if first buffer is within 1s
	  If we saw empty segments, we previously unconditionally pushed a
	  GAP event downstream regardless of the duration of that empty
	  segment.
	  In order to avoid issues with initial negotiation of downstream elements
	  (which would negotiate to something before receiving any data due to
	  that initial GAP event), check if there's at least a second of difference
	  (like we do for other GAP-related checks in qtdemux) before
	  deciding to push a GAP event downstream.

2017-12-13 10:21:17 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't set pared=True on underspecified audio/mpeg
	  This *really* needs to go through a parser to figure out what the
	  exact content type is.

2017-12-11 15:27:08 -0600  Michael Catanzaro <mcatanzaro@igalia.com>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Fix -Wincompatible-pointer-types warning
	  This is caused by the new type propagation for g_object_ref.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791494

2017-12-09 16:15:24 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/.gitignore:
	  tests: ignore rtph264 test binary

2017-08-25 15:19:37 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/udpsrc.c:
	  tests: udpsrc: verify the correct amount of bytes is sent to the socket
	  https://bugzilla.gnome.org/show_bug.cgi?id=786799

2017-08-25 14:59:06 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/udpsrc.c:
	  tests: udpsrc: ensure test won't timeout if the buffers are already received
	  Sometimes all the buffers are received before the time we lock the
	  check_mutex, in which case g_cond_wait will wait forever for another
	  one. Just check if this is the case before waiting.
	  https://bugzilla.gnome.org/attachment.cgi?id=358397

2017-08-25 14:45:52 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/udpsrc.c:
	  tests: udpsrc: fix test_udpsrc to actually run and fix locking
	  Previously this would silently be skipped because 1600 != 1400
	  and there is no assertion on this call.
	  Also unlock check_mutex after use.
	  https://bugzilla.gnome.org/show_bug.cgi?id=786799

2017-09-21 18:23:54 +0300  John Nikolaides <jnikolaides@toolsonair.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: added a "split now" action signal
	  Now, the video file can be split at an arbitrary time chosen by the user.
	  https://bugzilla.gnome.org/show_bug.cgi?id=787922

2017-12-08 00:31:32 +0000  Alvaro Margulis <alvaro.margulis@cirpack.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: fix bind address leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=790986

2017-12-07 11:15:19 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  Revert "flacparse: fix header rewriting being ignored"
	  This caused broken metadata and also looks a bit dodgy.
	  Revert until we can figure out a solution that works for
	  all cases and doesn't break anything.
	  This reverts commit adeee44b07a173b9ab4253216caba8f66dd43abb.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727802
	  https://bugzilla.gnome.org/show_bug.cgi?id=785558

2017-12-05 15:14:04 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Handle drivers that only round up height
	  Commit 1f31715c9861 ("v4l2videodec: use visible size, not coded size,
	  for downstream negotiation filter") added support for removing the
	  padding obtained as the difference between width/height from G_FMT and
	  visible width/height from G_SELECTION from the probed caps obtained
	  via TRY_FMT.
	  This patch fixes the padding removal for drivers that only round up
	  height, but not width, to the padded frame size. This might happen
	  because horizontal padding can be handled by line stride (bytesperline),
	  but there is no such thing as plane stride in the V4L2 API for
	  single-buffer planar formats.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791271

2017-11-01 08:21:37 -0600  Matt Staples <staples255@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Add a signal to allow outgoing messages to be modified or dropped
	  This feature allows applications to implement extensions to the RTSP
	  protocol, such as those defined in the ONVIF Streaming Specification.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762884

2017-08-25 11:57:26 +0200  Haakon Sporsheim <haakon@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Handle zero length feedback packets
	  https://bugzilla.gnome.org/show_bug.cgi?id=791074

2017-07-10 15:19:34 +0200  Florian Zwoch <fzwoch@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix debug log for 'hvcC' codec_data
	  https://bugzilla.gnome.org/show_bug.cgi?id=784749

2017-12-01 13:04:41 +0100  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpsession.c:
	  tests: rtpsession: refactor tests to use GstHarness
	  This patch simplifies the tests (44% less code) and
	  makes them much more readable.
	  The provided SessionHarness also makes it much easier
	  to write new tests for rtpsession.
	  https://bugzilla.gnome.org/show_bug.cgi?id=791070

2017-11-24 10:36:01 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Request at least the full header size when parsing headers
	  Otherwise baseparse will incrementally send us bigger buffers until the
	  full header size is reached, which is not only pointless but also means
	  that baseparse will reallocate and copy into a bigger buffer for every
	  input buffers. In pull mode that's done in 64kb increments, in push mode
	  usually in much smaller increments, causing a lot of overhead for
	  example when parsing high-quality coverart.

2017-11-29 11:29:31 +0100  Florent Thiéry <florent.thiery@ubicast.eu>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix dmabuf support detection
	  This resulted in improper selection of dmabuf on unsupported drivers.
	  The checked ioctl errno was not correct.
	  https://bugzilla.gnome.org/show_bug.cgi?id=790940

2017-11-27 20:10:51 +1100  Matthew Waters <matthew@centricular.com>

	* common:
	  Automatic update of common submodule
	  From 3f4aa96 to e8c7a71

2017-11-27 14:44:58 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  gl/caopengllayer: use public GstGLContext instead of Cocoa-specific one
	  Allows keeping the GstGLCAOpenGLLayer public but not the winsys-specific
	  context/display/window.

2017-11-26 15:13:15 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	  autotools: stop controlling symbol visibility with -export-symbols-regex
	  Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
	  This should result in consistent behaviour for the autotools and
	  Meson builds.

2017-11-24 15:37:44 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Do more checks for seekability
	  When receiving a seek event, check whether we can actually seek based
	  on the information the server provided.
	  Also add more documentation on what the seekable field means

2017-11-25 00:53:42 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Always update reserved-duration-remaining
	  If a reserved-max-duration is set, we should always track
	  and update the reserved-duration-remaining estimate, even
	  if we're not sending periodic moov updates downstream for
	  full robust muxing.

2015-04-07 23:53:19 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* tests/check/elements/splitmux.c:
	  splitmuxsink: Use muxer reserved space properties if present.
	  If the use-robust-muxing property is set, check if the
	  assigned muxer has reserved-max-duration and
	  reserved-duration-remaining properties, and if so set
	  the configured maximum duration to the reserved-max-duration
	  property, and monitor the remaining space to start
	  a new file if the reserved header space is about to run out -
	  even though it never ought to.

2017-11-24 08:00:21 +0100  Edward Hervey <edward@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: Fix possibility of NULL variable
	  It's quite unlikely since it's initialized in instance initialization.
	  CID #1417721

2017-11-24 16:56:03 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* tests/check/elements/splitmux.c:
	  splitmux: Fix file switch-on-caps-change.
	  Switching to a new fragment because the input caps have
	  changed didn't properly end the previous file. Use the normal
	  EOS sequence to ensure that happens. Add a test that it works.

2017-11-24 16:53:40 +1100  Jan Schmidt <jan@centricular.com>

	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpegenc: Update output caps on input caps change
	  If the input changes width/height that should be reflected
	  in the output caps, so make sure they get updated

2017-11-23 22:58:40 +1100  Jan Schmidt <jan@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  Revert "gl: Use GstGLDisplayEGL directly instead of creating a GstGLDisplayVIVFb subclass"
	  This reverts commit 47fd4d391e775c11f529705bb0f457a9d25ba5e7.
	  This patch is incorrect. It doesn't actually compile, and causes a crash
	  because the viv-fb window implementation needs a native EGL handle
	  to pass to fbCreateWindow, but the GstGLDisplayEGL handleis actually
	  an EGLDisplay now (and gets cast to the wrong type)

2017-09-05 15:55:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: don't insert SPS/PPS inline for hvc1 output
	  Only for byte-stream or hev1. For hvc1 the SPS/PPS are in the
	  caps as codec_data field and in this case they shouldn't be in
	  the stream data as well. The output caps should be updated with
	  the new codec_data if needed, for hvc1.

2017-09-05 15:47:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	  rtph265depay: store negotiated output format as enum
	  We keep the boolean byte_stream around since it's nicer for
	  readability and most of the code just cares about byte_stream
	  or not. This is useful for future-proofing the code for when
	  we add support for hev1 output as well.

2017-08-29 17:05:51 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: add support for hvc1 as output format

2017-08-08 18:58:11 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: don't add trailing zeros to VPS/PPS/SPS
	  This would happen if input is byte-stream with four-byte
	  sync markers instead of three-byte ones. The code that
	  scans for sync markers will place the start of the NALU
	  on the third-last byte of the NALU sync marker, which
	  means that any additional zeros may be counted as belonging
	  to the previous NALU instead of being part of the next sync
	  marker. Fix that so we don't send VPS/SPS/PPS with trailing
	  zeros in this case.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=732758

2017-06-16 12:41:49 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: assemble AUs into downstream-allocated memory
	  When merging NALs into AUs, use downstream-provided allocator
	  to allocate memory and copy NALs directly into that memory when
	  assembling them.

2017-06-16 12:30:13 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	  rtph265depay: try to negotiate an allocator with downstream

2017-06-16 12:13:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: simplify buffer accumulation control flow
	  There is no difference between pushing out a buffer directly
	  with gst_rtp_base_depayload_push() and returning it from the
	  process function. The base class will just call _depayload_push()
	  on the returned buffer as well.
	  So instead of marshalling buffers through three layers and back,
	  just push them from one place in handle_nal() and always return
	  NULL from the process vfunc. This simplifies the code a little.
	  Also rename _push_fragmentation_unit() to _finish_fragmentation_unit()
	  for clarity. Push sounds like it means being pushed out, whereas
	  it might just be pushed into an adapter.
	  This change has the side-effect that multiple NALs in a single STAP
	  (such as SPS/PPS) may no longer be pushed out as a single buffer if
	  we output NALs in byte-stream format (i.e. not aggregate AUs), but
	  that shouldn't really make any difference to anyone.

2017-06-16 11:18:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: fix crash with empty sprops-parameters
	  https://bugzilla.gnome.org/show_bug.cgi?id=780040

2017-06-16 12:20:34 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: minor clean-up
	  Declutter caps update code a bit.

2017-08-08 13:10:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: add unit test for rtph264pay codec_data
	  Make sure no trailing zero bytes sneak into our SPS or PPS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732758

2014-07-05 06:21:48 +0000  Philip Craig <phil@blackmoth.com.au>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: don't add trailing zeros to PPS/SPS
	  This would happen if input is byte-stream with four-byte
	  sync markers instead of three-byte ones. The code that
	  scans for sync markers will place the start of the NALU
	  on the third-last byte of the NALU sync marker, which
	  means that any additional zeros may be counted as belonging
	  to the previous NALU instead of being part of the next sync
	  marker. Fix that so we don't send SPS/PPS with trailing
	  zeros in this case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732758

2017-05-20 15:50:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/rtph264.c:
	* tests/files/Makefile.am:
	* tests/files/h264.rtp:
	  tests: rtph264depay: add test for using downstream memory allocator

2017-06-03 00:58:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: assemble AUs into downstream-allocated memory
	  When merging NALs into AUs, use downstream-provided allocator
	  to allocate memory and copy NALs directly into that memory when
	  assembling them.

2017-06-02 21:27:40 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: try to negotiate an allocator with downstream

2017-06-02 20:54:20 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: minor clean-up
	  Declutter caps update code a bit.

2017-11-23 08:00:58 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Run gst-indent

2017-11-23 07:59:07 +0100  Edward Hervey <edward@centricular.com>

	* gst/replaygain/rganalysis.c:
	  rganalysis: Fix left shift of signed values
	  left shifting signed values is undefined.
	  Instead of doing "x << offs" which is undefined, do the equivalent
	  "x * (1 << offs)" which is well defined

2017-11-23 07:57:44 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check presence of bitrate tags
	  Check whether the tag was present before printing it out
	  CID #1418501

2017-11-21 09:33:49 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use the proper maximum value for seekable
	  it's a gfloat, not a gdouble

2017-11-18 02:27:50 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use new GST_SEQNUM_INVALID constant

2017-11-18 02:01:58 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxsrc: Don't return FALSE from event handling.
	  Returning FALSE because we drop an event means that
	  internal sources like qtdemux might throw an error
	  and break the whole pipeline. The only time it can
	  happen is either flushing or shutdown, and those
	  will be handled anyway.

2017-10-22 18:26:12 +0800  Jun Xie <jun.xie@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: reset reused QtDemuxStream while parsing a new 'trak'
	  if QtDemuxStream is reused, then we need to reset it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=788759

2017-11-13 10:43:11 +0900  Seungha Yang <pudding8757@gmail.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	  isomp4: Add official fourcc for VP8 codec
	  fourcc for VP8 codec is "vp08" defined by spec. To follow it,
	  add it to demux and change legacy VP8 fourcc "VP80" to "vp08" in mux.
	  Also, enable sync table in case of VP8 codec.
	  See also https://www.webmproject.org/vp9/mp4/
	  https://bugzilla.gnome.org/show_bug.cgi?id=790026

2017-11-13 10:38:06 +0900  Seungha Yang <pudding8757@gmail.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/qtdemux.c:
	  isomp4: Add support VP9 codec
	  Add fourcc for VP9 codec and support it by qtdemux and qtmux
	  See also https://www.webmproject.org/vp9/mp4/
	  https://bugzilla.gnome.org/show_bug.cgi?id=790026

2017-11-13 13:51:20 +0100  Edward Hervey <edward@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Remove bogus error message
	  It's just informational

2017-11-10 15:51:05 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtp/gstrtpmpvpay.c:
	  rtpmpvpay: Don't create empty buffer list
	  If there's nothing to send, just return

2017-03-13 18:14:12 +0900  paul.kim <paul.hyunil@lge.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Remove range header when seek to 0
	  This fixes the previous range header is remained if seek to 0 is
	  attempted.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779957

2017-11-08 16:34:01 +0100  Edward Hervey <edward@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Fix seeking back to 0
	  This is a regression introduced by "03db374 - souphttpsrc: retry
	  request on early termination from the server"
	  The problem was that when seeking back to 0, we would not end up calling
	  add_range_header() which in addition to adding range headers *ALSO* sets
	  the read_position to the requested one.
	  This would result in a wide variety of later failures, like reading
	  again and again instead of stopping properly.

2017-11-07 18:03:53 +0900  Seungha Yang <pudding8757@gmail.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Add parsing Colour element
	  ... and forward colorimetry to downstream. The Colour element describes
	  various color information (similar to 'colr' box in isobmff).
	  Note that, due to the comparatively limited syntax for color information
	  in vpx codecs, the color information in mkv/wemb container level
	  should be used for sophisticated color handling (e.g., HDR video).
	  https://bugzilla.gnome.org/show_bug.cgi?id=790023

2017-10-19 14:02:37 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2deviceprovider: Ignore touch sensing devices
	  With GST_V4L2_USE_LIBV4L2=1, my laptop's touchpad shows up as a video
	  source device in gst-device-monitor, but attempting to stream from it
	  fails because the device doesn't actually support any video formats.
	  name  : Synaptics RMI4 Touch Sensor
	  class : Video/Source
	  caps  : video/x-raw, format=(string)I420, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)0, height=(int)0, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1;
	  video/x-raw, format=(string)YV12, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)0, height=(int)0, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1;
	  video/x-raw, format=(string)BGR, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)0, height=(int)0, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1;
	  video/x-raw, format=(string)RGB, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)0, height=(int)0, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1;
	  properties:
	  udev-probed = true
	  device.bus_path = /sys/devices/rmi4-00/rmi4-00.fn54/video4linux/v4l-touch0
	  sysfs.path = /sys/devices/rmi4-00/rmi4-00.fn54/video4linux/v4l-touch0
	  device.subsystem = video4linux
	  device.product.name = "Synaptics\ RMI4\ Touch\ Sensor"
	  device.capabilities = :capture:
	  device.api = v4l2
	  device.path = /dev/v4l-touch0
	  v4l2.device.driver = rmi4_f54
	  v4l2.device.card = "Synaptics\ RMI4\ Touch\ Sensor"
	  v4l2.device.bus_info = rmi4:rmi4-00.fn54
	  v4l2.device.version = 265480 (0x00040d08)
	  v4l2.device.capabilities = 2501902337 (0x95200001)
	  v4l2.device.device_caps = 354418689 (0x15200001)
	  gst-launch-1.0 v4l2src device=/dev/v4l-touch0 ! ...
	  v4l2-ctl -d /dev/v4l-touch0 --list-formats reports:
	  ioctl: VIDIOC_ENUM_FMT
	  Index       : 0
	  Type        : Video Capture
	  Pixel Format: 'TD16'
	  Name        : 16-bit signed deltas
	  Index       : 1
	  Type        : Video Capture
	  Pixel Format: 'TD08'
	  Name        : 8-bit signed deltas
	  Index       : 2
	  Type        : Video Capture
	  Pixel Format: 'TU16'
	  Name        : 16-bit unsigned touch data
	  https://bugzilla.gnome.org/show_bug.cgi?id=789197

2017-11-03 13:27:50 -0400  Youness Alaoui <kakaroto@kakaroto.homelinux.net>

	* gst/rtp/gstrtpg722pay.c:
	  rtpg722pay: Add encoding-params to the src caps template
	  The G722 payload only accepts G722 audio with channels=1, so it must
	  specify the encoding-params=1 in its src caps, otherwise it causes issues
	  with farstream which thinks it supports 2 channels G722 and when
	  confronted with a remote that has G722/8000/2, it will negotiate it
	  and error out with a not-negotiated when the caps don't intersect
	  at runtime.
	  https://bugzilla.gnome.org/show_bug.cgi?id=789878

2017-10-06 17:36:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2allocator: Add support for data_offset
	  In MPLANE mode, the driver may set data_offset, which represent some
	  padding at the start of the buffer used internally. This portion of the
	  data need to be skipped, though it is included in bytesused.
	  This patch removes frame size sanity check as the method used will no
	  longer work. This check was simply there to help detect broken kernel
	  drivers. It would be re-implement by estimating the plane size, which is
	  not totally trivial and may be too much work for a simple debug check.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733501

2017-07-17 17:09:18 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Add "accept-certificate" signal for manually checking a TLS certificate for validity
	  https://bugzilla.gnome.org/show_bug.cgi?id=785024

2017-10-30 19:15:56 +0900  Sangkyu Park <sk1122.park@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Print RTSP/SDP messages to gstreamer log instead of stdout
	  - 'debug' property is deprecated
	  - All RTSP messages are printed to gstreamer log with 'log' level.
	  https://bugzilla.gnome.org/show_bug.cgi?id=788917

2017-11-01 15:29:58 +0900  Justin Kim <justin.kim@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsesson: downgrade message level to debug when detected XR
	  When XR packet is detected, warning message leads to misunderstandings.
	  Until RFC3611 is implemented in gst-plugins-base, the level needs to
	  be downgraded to avoid confusion.
	  https://bugzilla.gnome.org/show_bug.cgi?id=789746

2017-10-24 20:12:29 +0530  Ashish Kumar <kr.ashish@samsung.com>

	* gst/isomp4/atomsrecovery.c:
	  gst-plugins-good: atoms_recovery: Handled buffer mapping failure
	  https://bugzilla.gnome.org/show_bug.cgi?id=789413

2017-07-08 22:11:49 -0700  Thiago Santos <thiagossantos@gmail.com>

	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/atomsrecovery.h:
	* gst/isomp4/gstqtmoovrecover.c:
	  atomsrecovery: read from mdat only what is on headers
	  It is possible that the mdat has more data than what was stored in the
	  headers file. If we put that to the output the file will have bogus data
	  at the end and some players will complain.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784258

2017-07-05 22:23:21 -0700  Thiago Santos <thiagossantos@gmail.com>

	* gst/isomp4/atomsrecovery.c:
	  isomp4: atomsrecovery: handle common and large atom headers
	  Do not assume all files are large files. Check and use the short or
	  extended atom size field only if needed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784258

2017-10-20 11:08:24 +0200  Andreas Frisch <afrisch@make.tv>

	* configure.ac:
	  pngdec: fix build with libpng versions between 1.2 and 1.5.1 (revised)
	  https://bugzilla.gnome.org/show_bug.cgi?id=765927

2017-10-19 18:23:34 +0200  Andreas Frisch <fraxinas@dreambox.guru>

	* configure.ac:
	* ext/libpng/gstpngdec.c:
	  pngdec: fix build with libpng versions between 1.2 and 1.5.1
	  https://bugzilla.gnome.org/show_bug.cgi?id=765927

2017-10-19 16:17:45 +0200  Andreas Frisch <fraxinas@dreambox.guru>

	* ext/libpng/gstpngdec.c:
	  pngdec: Extract icc profiles and send them downstreams for colormanagement elements
	  https://bugzilla.gnome.org/show_bug.cgi?id=765927

2017-10-16 14:20:47 +0200  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: Add missing Since marker

2017-10-13 12:25:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/qt/qtplugin.pro:
	  qt: update qmake .pro file
	  Update for renaming of plugin file, and add some
	  missing source files.

2017-06-13 18:51:32 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	  gstgdkpixbufdec: stop pretending to decode gifs.
	  If you can't decode an animated gif, you can't decode a gif,
	  so stop squatting GST_RANK_SECONDARY for that format, libav
	  does a better job.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784683

2017-09-28 22:51:57 +0200  Philippe Renon <philippe_renon@yahoo.fr>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: simplify how DirecSoundBuffer is cleared
	  we always want to clear the whole buffer so no need to
	  start from offset even if the offset is always zero.
	  https://bugzilla.gnome.org/show_bug.cgi?id=788847

2017-09-28 22:49:31 +0200  Philippe Renon <philippe_renon@yahoo.fr>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: fix comment
	  https://bugzilla.gnome.org/show_bug.cgi?id=788847

2017-09-28 22:48:41 +0200  Philippe Renon <philippe_renon@yahoo.fr>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: don't call set_volume with private scaled volume
	  use get_volume() instead to get unscaled volume
	  https://bugzilla.gnome.org/show_bug.cgi?id=788847

2017-09-28 22:46:23 +0200  Philippe Renon <philippe_renon@yahoo.fr>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: remove duplicate volume initialization
	  https://bugzilla.gnome.org/show_bug.cgi?id=788847

2017-10-10 18:04:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix compiler warning
	  qtdemux.c: In function ‘gst_qtdemux_configure_stream’:
	  qtdemux.c:7764:34: error: suggest parentheses around ‘&&’ within ‘||’ [-Werror=parentheses]
	  if ((stream->n_samples == 1) && (stream->first_duration == 0)
	  ~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

2017-09-22 18:41:52 +0200  Nael Ouedraogo <nael.ouedraogo@crf.canon.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix assert when moof containing one sample
	  Avoid computing frame rate when a stream contain moof with only one
	  sample, to avoid an assert. The moof is considered as still picture.
	  The same is already done for one sample given in the moov.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782217

2017-10-09 14:17:25 +0200  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Avoid potentially dereferencing NULL pointer
	  CID 1418986

2017-10-08 00:07:43 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix debug message on pt mismatch

2017-10-07 21:11:41 +0000  Nicolas Dufresne <nicolas@ndufresne.ca>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Fix driver capability dectection
	  Use the right set of caps when checking if caps intersect. That makes
	  the check only select the supported devices.

2017-09-20 01:46:15 +0000  Nicolas Dufresne <nicolas@ndufresne.ca>

	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc/dec: Don't leak template caps

2017-10-07 21:17:53 +0000  Nicolas Dufresne <nicolas@ndufresne.ca>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videodec: Protect against null pool in _stop
	  This may happen if the negotiation fails, as we will have never
	  created the pools.

2017-10-07 15:55:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtsp/gstrtspsrc.c:
	  rtpbin, rtspsrc: fix compiler warnings about 64-bit integer signednes
	  "warning: this decimal constant is unsigned only in ISO C90" with
	  gcc 4.8.4 (Ubuntu/Linaro 4.8.4-2ubuntu1~14.04.3)

2017-10-07 15:39:18 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix build without libv4l
	  https://bugzilla.gnome.org/show_bug.cgi?id=779466

2017-10-07 14:06:38 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpsbcdepay.c:
	  rtpsbcdepay: Fix potential NULL pointer dereference
	  CID 1418864

2017-10-07 01:21:19 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/audioecho.c:
	  audioecho: Micro-optimize
	  Gives 1.28x speedup in surround-delay=false mode

2017-10-06 23:59:43 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/audioecho.c:
	  audioecho: Don't do linear interpolation between samples
	  Linear interpolation adds quite some noise, and it's unlikely that
	  anybody will ever need sub-sample accurate delays. Proper resampling
	  before that will lead to better results.

2017-09-29 22:19:42 -0400  Enrico Jorns <ejo@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: auto-detect dmabuf export for V4L2_IO_AUTO on capture side
	  Issue an invalid VIDIOC_EXPBUF ioctl to the driver to check if the
	  driver supports dmabuf export. If the driver does not implement the
	  IOCTL, the error is ENOTTY. Any other error codes mean that the driver
	  implements VIDIOC_EXPBUF.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779466

2017-09-24 14:35:01 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Only set pixel-aspect-ratio if specified
	  If it's not specified, we should let the decoder figure it out.
	  Apparently the code was already in place, all was to make the code
	  conditional.
	  https://bugzilla.gnome.org/show_bug.cgi?id=787795

2017-09-23 15:44:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't pull passed the EOS
	  When a truncated FLV is provided and processed in pull mode, we
	  may endup trying to pull passed EOS, causing a rather confusing
	  warning as the pull offset is an integer overflow.
	  https://bugzilla.gnome.org/show_bug.cgi?id=787795

2017-09-23 15:41:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Ignore invalid H.264 codec data
	  This code basically skip over codec_data with empty payload. In
	  this case, the codec_data variable is the size of the header for
	  the CODEC part of Video Tag. The remaining is supposed to be the
	  H.264 codec data, hence should not be empty.
	  https://bugzilla.gnome.org/show_bug.cgi?id=787795

2017-09-23 15:38:07 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Avoid integer overflow on invalid CTS
	  If the CTS is negative an would lead to a negtive PTS, clip
	  the CTS so the PTS will be 0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=787795

2017-10-05 14:36:28 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-isomp4.xml:
	  docs: Update for git changes

2017-10-05 14:35:27 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix build

2017-07-13 14:46:55 -0400  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Handle TCP as lower transport with RTSP 2.0
	  Meaning that the interleave fields have to be updated as
	  if streams setup was working when using pipelined setup
	  request. Otherwise there is a mismatch between the server
	  channel count and our own.
	  This also makes RTSP 2.0 over HTTP working.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781446

2017-04-20 17:45:39 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtsp: Start implementing support for RTSP 2.0
	  - Handle version negotation:
	  Added a `default-version` property so that the user can configure
	  what to use in case the server does not support version negotation
	  (which actually exist)
	  - Handle pipelined requests, which allow avoiding full round trip to
	  setup the RTP streams (request are sent in a raw, and response are
	  handled as they arrive).
	  - Handle the new Media-Properties header
	  - Handle the new Seek-Style header
	  - Handle the new Accept-Ranges header
	  Handling of IPV6 should already be OK.
	  We are still missing (at least) the following features (which do not
	  seem really mandatory as they require a "persistent connection between
	  server and client"):
	  - Server to Client TEARDOWN command (Not so usefull fmpov)
	  - PLAY_NOTIFY (not needed for our server yet)
	  - Support for the new REDIRECT features
	  and probably some more protocol changes might not be handled yet.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781446

2017-05-03 11:19:03 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use a macro to debug RTSP messages
	  Simplifying the code a little.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781446

2017-10-03 16:30:10 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska-mux.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/spectrum/gstspectrum.c:
	  Use proper GtkDoc notation for NULL/FALSE/TRUE

2017-10-02 12:35:48 -0700  Cassandra Rommel <cassandra.rommel@gmail.com>

	* ext/qt/gstqtglutility.cc:
	  gl: Use GstGLDisplayEGL directly instead of creating a GstGLDisplayVIVFb subclass
	  This simplifies the code a lot without any functional changes apart from
	  not closing the display connection. Closing the display connection is
	  not safe to do as it is shared between all other code in the same
	  process and no reference counting or anything happens at the platform
	  layer.

2017-10-01 16:09:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Ignore medias marked as sendonly
	  We're never going to receive anything from them, so don't create pads
	  for them. These medias are destinations where *we* could send something.

2017-09-05 11:41:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsbcdepay.h:
	  sbcdepay: Add property to ignore input timestamps
	  This then just counts samples and calculates the output timestamps based
	  on that and the very first observed timestamp. The timestamps on the
	  buffers are continued to be used to detect discontinuities that are too
	  big and reset the counter at that point.
	  When receiving data via Bluetooth, many devices put completely wrong
	  values into the RTP timestamp field. For example iOS seems to put a
	  timestamp in milliseconds in there, instead of something based on the
	  current sample offset (RTP clock-rate == sample rate).
	  https://bugzilla.gnome.org/show_bug.cgi?id=787297

2017-09-21 13:59:00 +0530  Ponnam Srinivas <p.srinivas@samsung.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: Fix Memory leak in error case
	  https://bugzilla.gnome.org/show_bug.cgi?id=787937

2017-09-22 16:55:21 +0530  Deepak Srivastava <srivastava.d@samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fixed memory leak in error code path
	  https://bugzilla.gnome.org/show_bug.cgi?id=788041

2017-09-20 09:37:59 +0530  Ponnam Srinivas <p.srinivas@samsung.com>

	* ext/libpng/gstpngenc.c:
	  pngenc: fix memory leak in error code path
	  Don't leak row_pointers if frame can't be mapped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=787885

2017-09-19 17:55:58 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Don't leak codec name

2017-08-05 12:23:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2bufferpool: Don't stop streaming when pool is flushing
	  The purpose of being able to flush the buffer pool is only to
	  unlock any blocked operation. Doing streamoff/streamon had the
	  side effect of turning off and on the camera. As we do a flush_start
	  / flush_stop sequence when shutting down, that would cause a really
	  quick sequence of streamoff/streamon/streamoff/close which was
	  causing some cameras to stop working.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783945

2017-09-17 16:18:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: implement basic chain_list function
	  Doesn't do anything fancy yet, but still avoids lots of
	  unnecessary locking/unlocking that would happen if the
	  default chain_list fallback function in GstPad got invoked.

2017-09-17 12:50:30 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: use new gst_buffer_list_calculate_size()

2017-09-14 13:00:56 +0200  Patrick Radizi <patrickr@axis.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtpbin: add option for sanity checking timestamp offset
	  Timestamp offsets needs to be checked to detect unrealistic values
	  caused for example by NTP clocks not in sync. The new parameter
	  max-ts-offset lets the user decide an upper offset limit. There
	  are two different cases for checking the offset based on if
	  ntp-sync is used or not:
	  1) ntp-sync enabled
	  Only negative offsest are allowed since a positive offset would
	  mean that the sender and receiver clocks are not in sync.
	  Default vaule of max-ts-offset = 0 (disabled)
	  2) ntp-sync disabled
	  Both positive and negative offsets are allowed.
	  Default vaule of max-ts-offset = 3000000000
	  The reason for different default values is to be backwards
	  compatible.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785733

2017-09-14 11:20:17 +0200  Patrick Radizi <patrickr@axis.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtpbin: add option for increasing ts_offset gradually
	  Instant large changes to ts_offset may cause timestamps to move
	  backwards and also cause visible effects in media playback. The new
	  option max-ts-offset-adjustment lets the application control the rate to
	  apply changes to ts_offset.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784002

2017-09-06 07:59:56 +0000  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  qmlglsink: Expose itemInitialized as property
	  Instead of just signalling when ready exposing the state
	  as a property allows us to bind at any time if player is
	  loaded async.

2017-09-13 16:05:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Ensure all caps a fixated
	  The code relied on the list compare function to fixate the caps
	  but if the caps only has one structure, the compare function will
	  never get called. Capture device for which there is only one
	  structure in the caps would then get some assertion and later
	  fail badly.
	  Instead, fixate before inserting into the list and split the reading
	  and the fixation of the structures.

2017-09-13 11:52:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't leak the par value

2017-09-13 11:38:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/examples/v4l2/v4l2src-renegotiate.c:
	  v4l2-renegotiate: Don't leak the option context

2017-09-13 11:33:33 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/examples/v4l2/v4l2src-renegotiate.c:
	  v4l2src-renegotiate: Don't leak pipeline desc string

2017-09-13 11:32:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/examples/v4l2/v4l2src-renegotiate.c:
	  v4l2-renegotiate: Change --enable-dmabuf into --io-mode=
	  This gives allow testing dmabuf importation but also exportation buy
	  letting user pick anything from the io-mode property on v4l2src.

2017-09-11 20:24:27 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: search_cluster should find preceding cluster before target
	  ... since failing this constraint takes search_pos by surprise which might
	  then end up in an infinite loop.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=787538

2017-09-07 14:33:57 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtx{send,receive}: improve the debug messages
	  * use INFO/DEBUG/LOG/TRACE equaly and meaningfully;
	  previously rtprtxsend:LOG and rtprtxreceive:LOG would generate
	  a totally different amount of log traffic and sometimes it was
	  impossible to see the information you wanted without useless
	  spam being printed around
	  * improve the wording, give a reasonable and self-explanatory
	  amount of information
	  * print SSRCs in hex
	  * avoid G_FOO_FORMAT for readability (we are just printing integers)

2017-09-07 09:39:13 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/qt/gstplugin.cc:
	* ext/qt/qtplugin.pro:
	  qt: fix build with qmake
	  Move the package defines for GST_PLUGIN_DEFINE from the
	  command line into the source file to avoid quoting issues
	  (-DPACKAGE_NAME="foo" means the quotes won't actually make
	  it to the compiler and then it no longer gets a string constant).

2017-09-05 16:20:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	  Request minimum buffer even if need_pool is FALSE
	  When tee is used, it will not request a pool, but still it wants to
	  know how many buffers are required.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730758

2017-09-05 16:20:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/qt/gstqtsink.cc:
	  Request minimum buffer even if need_pool is FALSE
	  When tee is used, it will not request a pool, but still it wants to
	  know how many buffers are required.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730758

2017-09-05 15:30:40 +0100  Ian Jamison <ian.dev@arkver.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Handle BT2020 for colorspace and transfer
	  This was not fully handled in switches and
	  ub gst_v4l2_object_get_colorspace();
	  https://bugzilla.gnome.org/show_bug.cgi?id=787313

2017-09-05 15:29:24 +0100  Ian Jamison <ian.dev@arkver.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix colorimetry transfer lookup for 4K video
	  https://bugzilla.gnome.org/show_bug.cgi?id=787160

2017-09-06 11:25:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Check if caps have changed after try_fmt
	  try_fmt will update the caps colorimetry and interlace-mode. Before this
	  call, those field are missing. The caps equality check was always
	  failing when a spurious reconfigure event was received.

2017-09-06 23:55:38 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Allow MPEG layer 1/2, AC3 and Opus in qtmux
	  qtmux is supposed to be the muxer that allows all formats,
	  with others (mp4mux and friends) being profile-restricted.

2017-09-05 12:56:44 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: fix keyunit detection
	  https://bugzilla.gnome.org/show_bug.cgi?id=787254

2017-09-05 15:42:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Fix decoding of streams that don't signal exactly twice the height
	  ... and also progressive streams.

2017-09-05 13:28:16 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Handle interlaced MJPEG streams
	  These come with two JPEG images per buffer of half height than signalled
	  in the container.
	  Changes based on Tim-Philipp Müller's 0.10 branch:
	  https://cgit.freedesktop.org/~tpm/gst-plugins-good/log/?h=jpegdec-interlaced
	  https://bugzilla.gnome.org/show_bug.cgi?id=568555

2017-09-01 15:00:12 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gtkgstglwidget.c:
	  gtkglsink: expose the created display and context correctly
	  1. Propagate the GstGLDisplay we create
	  2. Add the created GstGLContext to the propagated GstGLDisplay
	  Otherwise with multi-branch GL pipelines involving gtkglsink, things
	  will fall apart and errors will be genarated somewhere.

2017-09-04 17:06:39 +0200  Edward Hervey <edward@centricular.com>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: Really fix "usage before unmap"
	  Previous patch would try to unref a buffer that was pushed downstream.
	  Instead only unref when/if needed and keep usage of the cleanup: goto
	  block

2017-09-03 15:23:10 +0530  Arun Raghavan <arun@arunraghavan.net>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: Don't unmap buffer before accessing data from it
	  The previous patch added a check for a substream header after
	  gst_buffer_unmap(), which is incorrect.

2017-06-24 18:47:14 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: preserve DTS HD substream

2017-09-01 15:56:04 +0200  Edward Hervey <edward@centricular.com>

	* ext/qt/gstqtgl.h:
	  qt: Only include qtgui-config.h on qt >= 5.9.0
	  The file does not exist in previous versions

2017-08-31 14:40:44 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtgl.h:
	  qt: the defines for QT_OPENGL_ES_2 have moved
	  Update the includes to account for that

2017-04-26 13:50:41 +0200  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* ext/qt/qtwindow.cc:
	  qt: ensure GL_DRAW_FRAMEBUFFER

2017-08-14 18:18:07 +0530  Arun Raghavan <arun@arunraghavan.net>

	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: Fix some tabs that crept in somehow

2017-08-29 19:13:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Also log local and SR RTP running times when doing ntp-sync=true

2017-08-24 17:06:38 +1000  Matthew Waters <matthew@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: also create session when creating the send_rtcp_src_%u pad
	  If one requests the send_rtcp_src_%u pad before a recv_rtcp_sink_%u pad,
	  the session/pad would never be created and NULL was returned.
	  Switching the request order would work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=786718

2017-08-26 12:59:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/files/Makefile.am:
	* tests/files/cbr_stream.mp3:
	* tests/files/stream.mp2:
	* tests/files/vbr_stream.mp3:
	  tests: mpg123audiodec: add files needed by unit tests

2017-08-26 10:10:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/gst-plugins-good.supp:
	* tests/check/pipelines/.gitignore:
	* tests/check/pipelines/lame.c:
	* tests/check/pipelines/twolame.c:
	  tests: add basic unit test for twolame as well

2017-08-26 09:59:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/pipelines/lame.c:
	  tests: lame: fix build

2017-08-26 09:52:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/v4l2/.gitignore:
	  tests: ignore another binary

2017-08-26 09:41:13 +0100  Tim-Philipp Müller <tim@centricular.com>

	* REQUIREMENTS:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-twolame.xml:
	* ext/Makefile.am:
	* ext/meson.build:
	* ext/twolame/meson.build:
	* po/POTFILES.in:
	  twolame: hook up to build system
	  https://bugzilla.gnome.org/show_bug.cgi?id=774252

2017-08-26 09:21:44 +0100  Tim-Philipp Müller <tim@centricular.com>

	  Moving twolame mp2 encoder plugin from -ugly
	  https://bugzilla.gnome.org/show_bug.cgi?id=774252

2017-08-26 09:03:08 +0100  Tim-Philipp Müller <tim@centricular.com>

	* REQUIREMENTS:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-lame.xml:
	* ext/Makefile.am:
	* ext/lame/Makefile.am:
	* ext/lame/meson.build:
	* ext/meson.build:
	* po/POTFILES.in:
	* tests/check/Makefile.am:
	* tests/check/gst-plugins-good.supp:
	* tests/check/meson.build:
	  lame: hook up to build system
	  https://bugzilla.gnome.org/show_bug.cgi?id=774252

2017-08-25 21:13:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	  Moving lame mp3 encoder plugin from -ugly
	  https://bugzilla.gnome.org/show_bug.cgi?id=774252

2017-08-22 12:39:43 +0100  Julien Isorce <jisorce@oblong.com>

	* ext/qt/gstqsgtexture.cc:
	* ext/qt/gstqtglutility.cc:
	* ext/qt/gstqtsink.cc:
	* ext/qt/qtwindow.cc:
	  qt: fix broken build due to commit 2fd84a6c for gstgl
	  https://bugzilla.gnome.org/show_bug.cgi?id=784779

2017-07-07 16:15:12 +0100  Julien Isorce <jisorce@oblong.com>

	* ext/gtk/Makefile.am:
	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gtkgstglwidget.c:
	* tests/examples/gtk/glliveshader.c:
	  gl: do not include GL headers in public gstgl headers
	  Except for gst/gl/gstglfuncs.h
	  It is up to the client app to include these headers.
	  It is coherent with the fact that gstreamer-gl.pc does not
	  require any egl.pc/gles.pc. I.e. it is the responsability
	  of the app to search these headers within its build setup.
	  For example gstreamer-vaapi includes explicitly EGL/egl.h
	  and search for it in its configure.ac.
	  For example with this patch, if an app includes the headers
	  gst/gl/egl/gstglcontext_egl.h
	  gst/gl/egl/gstgldisplay_egl.h
	  gst/gl/egl/gstglmemoryegl.h
	  it will *no longer* automatically include EGL/egl.h and GLES2/gl2.h.
	  Which is good because the app might want to use the gstgl api only
	  without the need to bother about gl headers.
	  Also added a test: cd tests/check && make libs/gstglheaders.check
	  https://bugzilla.gnome.org/show_bug.cgi?id=784779

2017-08-20 20:41:19 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/check/meson.build:
	  meson: Link mpeg123audiodec test against gstfft
	  Fixing build error:
	  /run/build/gst-plugins-good/_flatpak_build/../tests/check/elements/mpg123audiodec.c:150: undefined reference to `gst_fft_s32_new'
	  /run/build/gst-plugins-good/_flatpak_build/../tests/check/elements/mpg123audiodec.c:151: undefined reference to `gst_fft_s32_window'
	  /run/build/gst-plugins-good/_flatpak_build/../tests/check/elements/mpg123audiodec.c:151: undefined reference to `gst_fft_s32_fft'
	  /run/build/gst-plugins-good/_flatpak_build/../tests/check/elements/mpg123audiodec.c:147: undefined reference to `gst_fft_s32_free'

2017-08-20 17:15:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/pipelines/tagschecking.c:
	  tests: tagschecking: remove gst-check-xmp-* temp files when done
	  Also fix temp file creation a bit.

2017-08-20 15:49:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	  docs: update for changes in git

2017-08-20 15:48:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-mpg123.xml:
	  mpg123: add to docs

2017-08-20 13:56:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* REQUIREMENTS:
	* configure.ac:
	* ext/Makefile.am:
	* ext/meson.build:
	* ext/mpg123/meson.build:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/meson.build:
	  mpg123: hook up to build system
	  https://bugzilla.gnome.org/show_bug.cgi?id=774252

2017-08-20 13:48:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	  Moving mpg123 plugin from -ugly

2017-08-17 12:23:25 +0100  Tim-Philipp Müller <tim@centricular.com>

	* README:
	* common:
	  Automatic update of common submodule
	  From 48a5d85 to 3f4aa96

2017-08-14 15:28:22 +0800  Sky Juan <skyjuan@realtek.com>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: fix not-linked handling causing glitches when selecting stream
	  Fix chain function not handling not-linked from baseparse.
	  When an input data is separated into 2 buffers, the second buffer
	  would not be pushed into the adapter if baseparse returns not-linked
	  for first buffer.
	  This caused glitches when switching streams and selecting
	  a stream that was previously unselected.
	  https://bugzilla.gnome.org/show_bug.cgi?id=786268

2017-08-16 13:57:50 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/goom2k1/filters.c:
	* gst/goom2k1/filters.h:
	* gst/goom2k1/goom_core.c:
	  goom2k1: Convert source files to UTF-8
	  Causes problems with the new gtk-doc 1.26 otherwise,
	  but is a good idea in any case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=786364

2017-08-14 03:08:41 -0500  Eduard Sinelnikov <eduard@reporty.com>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: Add support for growing WAV files
	  With some fixes by me.

2017-08-14 17:39:15 +0530  Arun Raghavan <arun@arunraghavan.net>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Fix compile error

2017-05-21 16:01:14 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  qmlglsink: Add itemInitialized signal to QML item
	  This is useful for autoplay for example. With autoplay, it is necessary to
	  wait until the scene graph is fully set up. This signal is emitted once the
	  QML item node is ready. So, inside a connected slot, the pipeline's state
	  can be set to PLAYING to automatically start playback as soon as the QML
	  script is loaded.
	  https://bugzilla.gnome.org/show_bug.cgi?id=786246

2017-08-14 10:36:56 +0000  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: fix if buffer size exceeds MTU
	  The plugin queued buffer data if not all buffer data fit
	  into a single RTP packet. Now RTP packets are pushed as long
	  as enough data is available.

2017-07-27 17:21:48 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/vpx/gstvpxenc.c:
	  vpxenc: discard frames that have been dropped by libvpx
	  This fixes a memory leak. When dropframe-threshold has been set,
	  libvpx may output less frames than the input ones, which causes
	  some GstVideoCodecFrames to queue up in GstVideoEncoder's internal
	  frame queue with no chance of ever being all released. And because
	  the frames keep references to the input buffers, the input buffer
	  pool keeps allocating new buffers and memory usage grows very fast.
	  For example the following pipeline's memory usage grows at a rate
	  of about 1GB per minute!
	  videotestsrc ! capsfilter caps=video/x-raw,width=1920,height=1080,framerate=30/1,format=I420 ! \
	  vp8enc target-bitrate=1000000 end-usage=cbr dropframe-threshold=95 ! fakesink
	  https://bugzilla.gnome.org/show_bug.cgi?id=783086

2017-08-08 13:11:58 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpstats: fix unsigned integer comparisons.
	  Callers of the API (rtpsource, rtpjitterbuffer) pass clock_rate
	  as a signed integer, and the comparison "<= 0" is used against
	  it, leading me to think the intention was to have the field
	  be typed as gint32, not guint32.
	  This led to situations where we could call scale_int with
	  a MAX_UINT32 (-1) guint32 as the denom, thus raising an
	  assertion.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785991

2017-08-10 14:44:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/taglib/meson.build:
	  taglib: use -fvisibility=hidden with this C++ plugin in meson too
	  Also pass args as cpp_args.

2017-03-22 15:25:17 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/isomp4/qtdemux.c:
	  qtdemux: allow larger files
	  For really long files such as contiguous recordings of a whole day, the
	  50MB limit is not sufficient.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781458

2017-08-10 16:08:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix offsets for reading lpcm specific fields
	  We were reading at the completely wrong positions, 16 bytes later in the
	  data.
	  Also add support for high-aligned samples.

2017-08-10 14:01:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: don't export symbols by default
	  Only plugin entry points should be exported.
	  Currently plugins might export more symbols with
	  the meson build, as we don't have the exports
	  regexp there that we pass to libtool.

2017-08-10 15:14:31 +0530  Deepak Srivastava <srivastava.d@samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix memory leak in wavparse element
	  Fixing of leaking the text field of the GstWavParseNote and
	  GstWavParseLabl structure.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785429

2017-08-08 10:37:12 +0000  Cyril Lashkevich <notorca@gmail.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Don't mark jpeg frames as deltas
	  JPEG formats are encoded, but they never have keyframe flag. But in
	  fact they are keyframes
	  https://bugzilla.gnome.org/show_bug.cgi?id=785990

2017-08-06 13:06:45 +0100  Philippe Normand <philn@igalia.com>

	* sys/osxvideo/Makefile.am:
	  osxvideo: rename library according to the plugin name
	  https://bugzilla.gnome.org/show_bug.cgi?id=785880

2017-08-02 17:16:21 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Don't drop buffer ref on qbuf
	  This function no longer take ownership of the buffer.
	  CID 1414800

2017-08-02 17:13:55 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Enable VP9 format
	  This was missing, preventing the encoder and decoder to work
	  properly. This also adds support for camera that would produce
	  VP9 (if that exists).

2017-08-02 12:28:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2h263enc.h:
	* sys/v4l2/gstv4l2h264enc.h:
	* sys/v4l2/gstv4l2mpeg4enc.h:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2transform.h:
	* sys/v4l2/gstv4l2videodec.h:
	* sys/v4l2/gstv4l2videoenc.h:
	* sys/v4l2/gstv4l2vp8enc.h:
	* sys/v4l2/gstv4l2vp9enc.h:
	  v4l2: Remove spurious CATEGORY_EXTERN
	  These have been copy pasted all over the place and are not used anymore.
	  All object have it's own category now. This fixes build warning since
	  the VP9 decoder had vp8 category declared.

2017-08-02 10:39:46 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2h264enc.c:
	* sys/v4l2/gstv4l2mpeg4enc.c:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/gstv4l2videoenc.h:
	* sys/v4l2/gstv4l2vp8enc.c:
	* sys/v4l2/gstv4l2vp9enc.c:
	  v4l2videoenc: Move the profile/level negotation in the base class
	  This removes duplicated code across different codec.

2017-08-02 09:36:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2h263enc.c:
	* sys/v4l2/gstv4l2h264enc.c:
	* sys/v4l2/gstv4l2mpeg4enc.c:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/gstv4l2videoenc.h:
	* sys/v4l2/gstv4l2vp8enc.c:
	* sys/v4l2/gstv4l2vp9enc.c:
	  v4l2videoenc: Turn gst_v4l2_is_video_enc into a helper
	  This reduces the amount of code needed in each codec class.

2017-08-01 16:01:11 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2vp8enc.c:
	* sys/v4l2/gstv4l2vp8enc.h:
	* sys/v4l2/gstv4l2vp9enc.c:
	* sys/v4l2/gstv4l2vp9enc.h:
	* sys/v4l2/meson.build:
	  v4l2: Add VP8/9 encoder support

2017-07-31 11:56:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Use mmap64 to match libv4l2 signature
	  https://bugzilla.gnome.org/show_bug.cgi?id=785628

2017-08-01 09:22:43 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Copy flags and timestamp when importing
	  Whenever we import from downstream pool (userptr or dmabuf-import), we
	  should copy over the flags and timestamp, otherwise downstream will not
	  get proper synchronization or will not be able to notice frames that has
	  corruption in it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785680

2017-07-31 16:09:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2h263enc.c:
	* sys/v4l2/gstv4l2h263enc.h:
	* sys/v4l2/meson.build:
	  v4l2: Add H263 Encoder support

2017-07-27 13:51:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	  v4l2: Add missing no-inst header

2017-07-26 15:18:01 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2mpeg4enc.c:
	* sys/v4l2/gstv4l2mpeg4enc.h:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/gstv4l2videoenc.h:
	* sys/v4l2/meson.build:
	  v4l2: Add interface for MPEG4 encoding

2017-07-27 10:51:07 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2h264enc.c:
	* sys/v4l2/gstv4l2h264enc.h:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2transform.h:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videodec.h:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/gstv4l2videoenc.h:
	  v4l2: Ignore register issue and keep probing
	  Don't stop registering the other dynamic plugins if one registration
	  fails.

2017-07-27 14:21:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/law/mulaw-decode.c:
	  mulawdec: Unmap input buffer if failing to map the output buffer

2017-07-27 09:22:25 +0530  Satya Prakash Gupta <sp.gupta@samsung.com>

	* gst/law/alaw-decode.c:
	  alawdec: Fix Memory leak in error case
	  https://bugzilla.gnome.org/show_bug.cgi?id=785435

2017-07-26 20:36:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/v4l2-common.h:
	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	  v4l2: Update external files with latest
	  This is copied from the linux kernel with only some include changes so
	  it works outside the kernel headers.

2017-07-18 10:41:40 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: For audio tracks, take the default duration from the first buffer
	  ... if we don't have any better idea from the caps. This allows writing
	  SimpleBlocks for a majority of audio streams where the duration of
	  frames is usually fixed. And as a side effect, allows VLC to play
	  streams with Opus as it only works with SimpleBlocks currently:
	  https://trac.videolan.org/vlc/ticket/18545
	  https://bugzilla.gnome.org/show_bug.cgi?id=784969

2017-07-24 16:45:40 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Fix compilation without libv4l2

2017-07-24 16:13:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Keep ref to element in allocator/pool
	  Removes the FIXME/Question in the buffer pool and add a ref to the
	  element in the GstAllocator too. This ref is strictly required to keep
	  the GstV4l2Object structure around.

2017-07-24 14:27:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Removed unused members

2017-07-24 14:19:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2h264enc.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: Add run-time environment to enable libv4l2
	  The library has started preventing a lot of interesting use cases,
	  like CREATE_BUFS, DMABuf, usage of TRY_FMT. As the libv4l2 is totally
	  inactive and not maintained, we decided to disable it. As a convenience
	  we added a run-time environment that let you enable it for testing.
	  GST_V4L2_USE_LIBV4L2=1
	  This of course only works if you have enabled libv4l2 at build time.

2017-07-17 10:04:02 +0200  Nicola Murino <nicola.murino@gmail.com>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: declare quality property changeable in PLAYING state
	  https://bugzilla.gnome.org/show_bug.cgi?id=785012

2017-07-21 23:34:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix colorimetry validation
	  While not documented, gst_video_colorimetry_matches() only accepts well
	  known names. Looking at the code and unit test, this seems to be on
	  purpose, so fixing by parsing the string and compating the colorimetry
	  structures.

2017-07-21 15:40:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2encoder: Fix negotiation error handling
	  The subclass negotiated function will call set_format, if that fails the
	  pool will not be created. We ended up with an assertion.
	  GStreamer-CRITICAL **: gst_buffer_pool_set_active: assertion 'GST_IS_BUFFER_POOL (pool)' failed

2017-07-19 22:25:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Speedup camera startup by skipping try_fmt
	  In this commit, we enabled skip_try_fmt_probes quirk in order to speed
	  up the start which is known to be disastrously slow with certain USB
	  cameras.
	  This has the side effect that we needed to rewrite the entire
	  negotiation process in a way that we iterate over the possible caps
	  until we find one that works.
	  The new negotiation method consist of extracting a preferred structure
	  from the peer caps and using this to fixate and sort the caps. To
	  reflect the old behaviour, we sort all resolution strictly bigger
	  to the preferred one with the closes one first. The rest is appended,
	  keeping the same order. We then normalize the caps in case there was
	  some list of interlace-mode or colorimetry left. We finally iterate
	  over all fixed caps and try it. 99% of the time, the first or the
	  second one should work, whit the result of a single S_FMT being issues.
	  From there, it will be relatively easy to introduce new negotiation
	  algorithm. The current algorithm is made for optimal image quality
	  with a scaling sink that sets it's window resolution as preference.
	  This the case if for:
	  v4l2src ! videoconvert ! videoscale ! ximagesink
	  Other strategy would be needed to optimize for non-scaling sink like
	  ximagesink or kmssink when the driver does not scale.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785156

2017-07-19 22:09:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Introduce quirk to skip slow probes
	  skip_try_fmt_probes quirk is set, V4L2 object will not probe for
	  interlace-mode and colorimetry to avoid relying on try_fmt. This quirk
	  will be used by v4l2src to avoid desastrous startup time with slow
	  USB webcams.
	  When this quirk is enabled, caller will have to iterate over the
	  negotiated caps as it may contains unsupported formats. If the peer
	  didn't choose a specific interlace-mode, or colorimetry, the value
	  chosen by the driver is set into the caps. For this reason, when this
	  mode is enabled, gst_v4l2_object_set_format() will require writable
	  caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785156

2017-07-19 22:07:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: always set the GstV4l2Error on error
	  Some of the error case were conditional to using try_fmt or not.
	  This is slightly unexpected, always set the error so the caller
	  can decide.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785156

2017-07-19 22:05:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Minor style fix and useful trace
	  https://bugzilla.gnome.org/show_bug.cgi?id=785156

2017-07-19 22:03:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix try/s_fmt errors
	  According to the spec,TRY_FMT cannot return EBUSY, though it can
	  return EINVAL if it was not possible to update the format to
	  something supported.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785156

2017-07-19 22:01:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Validate colorimetry in S/TRY_FMT
	  This is in preparation for removing slow TRY_FMT probes for
	  colorimetry. As we won't have tried that colorimetry we cannot
	  assume the driver will accept it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785156

2017-07-19 21:56:14 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Validate field in S/TRY_FMT
	  This is in preparation from removing the slow TRY_FMT probes for
	  interlacing. As we won't have tried that interlace-mode already
	  we need to validate that the driver isn't refusing it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=785156

2017-07-21 19:01:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/test-accurate-seek.c:
	  tests: icles: fix build
	  Can't do additions/subtractions on void* pointers.

2017-07-21 11:04:17 -0400  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/icles/test-accurate-seek.c:
	  tests:icles: Fix previous patch by implementing our memmem
	  Using the string version of it will fail on '\0'.

2017-07-21 10:17:00 -0400  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/icles/test-accurate-seek.c:
	  tests:icles: Do not use memmem GNU extension function
	  As it is not avalaible on windows/msvc and we can use pure GLib for that

2017-07-20 17:21:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/directsound/Makefile.am:
	  directsound: Fix .c file name in Makefile
	  This was broken by accident, bad search and replace.

2017-07-20 11:02:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* Makefile.am:
	* sys/waveform/Makefile.am:
	  waveform: Fix DLL name to match plugin name
	  https://bugzilla.gnome.org/show_bug.cgi?id=785168

2017-07-20 10:38:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* Makefile.am:
	* sys/directsound/Makefile.am:
	* sys/directsound/meson.build:
	  directsound: Fix DLL name to match plugin name
	  https://bugzilla.gnome.org/show_bug.cgi?id=785168

2017-07-19 12:38:03 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: preferably send open-ended segment rather than repeated segment events

2017-07-19 11:27:32 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix seeking in fragmented file without mfra random access info
	  ... which no longer worked due to unconditionally clearing sample info and
	  ending up in inconsistent state.  Let's tread a bit more carefully and also
	  allow for the old seek handling that resorts to scanning if no mfra info
	  is available.

2017-07-19 10:42:46 +0200  Nicolas Dechesne <nicolas.dechesne@linaro.org>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: add some useful debug messages
	  Add a couple of useful debug traces , they happened to be useful to
	  debug/investigate a 4K video playback issue with v4l2, so let's make these
	  changes more permanent.
	  Signed-off-by: Nicolas Dechesne <nicolas.dechesne@linaro.org>
	  https://bugzilla.gnome.org/show_bug.cgi?id=785109

2017-07-18 11:28:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Fix 4K colorimetry
	  Since 1.6, the transfer function for BT2020 has been changed from BT709
	  to BT2020_12. It's the same function, but with more precision. As a side
	  effect, the V4L2 colorpsace didn't match GStreamer colorspace. When
	  GStreamer ended up making a guess, it would not match anything supported
	  by V4L2 anymore. This this by using BT2020_12 for BT2020 colorspace and
	  BT2020 transfer function in replacement of BT709 whenever a 4K
	  resolution is detected.

2017-07-14 16:21:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Only check CROPCAP for par once
	  The pixel aspect ratio is documented to not change unless the TV
	  Standard is changed. So this mean that this will be uniform across all
	  possible format and resolutions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784674

2017-07-18 10:01:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/matroskamux.c:
	  Revert "matroskamux: adjust unit test to modified behaviour"
	  This reverts commit 8fe478c8a7746cd2c63f20d23e97e26e1a0e6192.
	  We're back to previous behaviour

2017-07-18 00:26:11 +0200  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: add properties to control cluster duration
	  https://bugzilla.gnome.org/show_bug.cgi?id=784971

2017-07-17 20:47:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: UVC driver is named uvcvideo these days
	  The quirk to avoid probing interlacing didn't work anymore as the driver
	  is now name uvcvideo. This should slightly speed up camera startup.

2017-07-12 21:02:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Remove unused defines

2017-07-12 20:53:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: Make gst_v4l2_get_capabilities static
	  It's not used outside of v4l2_calls.c

2017-07-12 20:49:47 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/gstv4l2h264enc.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/gstv4l2vidorient.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	  v4l2: Merge v4l2_calls.h into gstv4l2object.h
	  First step of a larger cleanup, all function from v4l2_calls are in fact
	  methods on GstV4l2Object. This split makes the code really confusing.
	  This also remove no longer unused macros.

2017-07-15 14:57:49 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123audiodec: fix caps leak
	  The pad template takes its own ref, so we should unref the caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784982

2017-07-15 12:48:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* po/meson.build:
	  meson: po: use glib preset and read language list from LINGUAS
	  Supported since meson 0.37, so we can use it now.

2017-07-14 12:12:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Trace unknown fourcc as text
	  This makes it easier to find out what is not supported.

2017-07-14 11:54:57 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: Don't probe for unneeded format
	  For v4l2videodec/enc, we generate elements per formats, and in
	  this case we can speed up the start up by only probing the format
	  we care about.

2017-07-13 12:32:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Implement stable element names
	  Before that, each m2m node would be wrapped as a single, multi-format
	  decoder element. As a unique name was needed, we where using the device
	  name, which changes between re-boots. This led to unpredictable element
	  names. In this patch, we generate an element per codec, using
	  v4l2<codec>dec name. If there is multiple decoder for the same format,
	  the following elements will be named v4l2<node><codec>dec.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784908

2017-07-13 14:50:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Post an element message with the HTTP headers on the bus too
	  Instead of just sending a sticky event with them downstream. This allows
	  getting the HTTP headers easily in the application, and especially also
	  on errors.

2017-07-13 12:47:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix parsing of RLE depth
	  Regression introduced by 86b427dc70562f891a551ffc9f96cefe1cafcddd
	  https://bugzilla.gnome.org/show_bug.cgi?id=784812

2017-07-12 15:29:32 +1000  Jan Schmidt <jan@centricular.com>

	* ext/qt/gstqtsink.cc:
	* ext/qt/gstqtsink.h:
	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  qt: Use a proxy object for access to the QML widget
	  QML can destroy the video widget at any time, leaving
	  us with a dangling pointer. Use a lock and a proxy
	  object to cope with that, and block in the widget
	  destructor if there are ongoing calls into the widget.

2017-07-10 18:57:11 +0200  Philippe Renon <philippe_renon@yahoo.fr>

	* ext/shout2/gstshout2.h:
	  shout2: use gint and guint in place of int and uint
	  this fixes a compilation error with gcc 7.1.0 on mys2 where uint is not defined
	  https://bugzilla.gnome.org/show_bug.cgi?id=784758

2017-07-07 21:15:57 +0900  Yasushi SHOJI <yashi@atmark-techno.com>

	* gst/rtp/gstrtpgsmpay.c:
	  rtpgsmpay: fix accidental garbage data before actual payload
	  Do not allocate payload size outbuf if appending payload buffer.
	  The commit 137672ff1824948bda4b1b1967de8c24a0055b67 attached payload
	  to the output buffer but forgot to remove payload allocation.  That
	  effectively doubled payload size and add zero'ed or random bytes.
	  Makes the following pipeline work again:
	  gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay ! gsmdec ! autoaudiosink
	  https://bugzilla.gnome.org/show_bug.cgi?id=784616

2017-07-01 18:57:47 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: segment seek position is expressed in buffer time
	  ... so it need not be corrected again for stream start

2017-07-09 10:54:27 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavidemux.c:
	  avidemux: provide average bitrate tag

2017-07-07 23:49:44 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* tests/examples/v4l2/v4l2src-renegotiate.c:
	  examples: v4l2: fix wrong initializations brought by 4e8ad583022671c5
	  https://bugzilla.gnome.org/show_bug.cgi?id=682770

2015-02-27 13:03:42 -0300  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/examples/v4l2/Makefile.am:
	* tests/examples/v4l2/meson.build:
	* tests/examples/v4l2/v4l2src-renegotiate.c:
	  examples: v4l2: add example for v4l2src renegotiation
	  Based on work from Thiago Santos <thiagoss@osg.samsung.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=682770

2017-07-07 11:58:10 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: find python3 via python3 module
	  https://bugzilla.gnome.org/show_bug.cgi?id=783198

2017-07-05 14:44:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	  tests: rtpbin: fix build in uninstalled setup

2017-07-04 17:42:25 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpbin.c:
	  rtpsession: Send EOS if all internal sources sent bye
	  The ones which are not internal should not matter, and we should
	  wait for all sources to have sent their BYEs.
	  And add unit test
	  https://bugzilla.gnome.org/show_bug.cgi?id=773218

2017-07-04 12:24:41 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Only send EOS if all sources have been marked bye
	  Now that multiple sender RTPSource can share the same RTPSession, we
	  must not send an EOS unless they're all marked bye.

2017-07-04 11:49:29 -0400  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* ext/libcaca/gstcacasink.c:
	  caca: Do not include, unused, sys/time.h
	  Which moreover makes building on windows (mingw/msvc) fail:
	  https://ci.appveyor.com/project/thiblahute/gst-build-ge9m5

2017-07-03 11:47:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtprtxreceive: Add memory and boudary checks
	  This element was not checking if mapping the RTP buffer and the payload
	  worked, and was not checking if the RTX payload was large enough.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784484

2017-07-04 14:58:00 +0900  Seungha Yang <sh.yang@lge.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Unset limit on the number of connection if soup session sharing is used
	  Soup allows only up to two connections per host in a session,
	  if we use default value. When session sharing is used, however,
	  more connections might be required in a session.
	  (e.g., multi-audio adaptive streaming case)
	  https://bugzilla.gnome.org/show_bug.cgi?id=784495

2017-07-03 20:27:29 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: fix use-after-free on seek event
	  Get seqnum before unreffing the seek event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784486

2017-07-01 18:59:14 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/gstqtmux.c:
	  qtmux: robustify time tracking for sparse subtitle stream

2017-07-01 18:59:07 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/gstqtmux.c:
	  qtmux: correctly track chunk size of subtitle stream
	  ... thereby ensuring correct chunk offset tracking for all streams.

2017-06-27 15:59:18 +0100  Julien Isorce <jisorce@oblong.com>

	* gst/rtpmanager/rtpstats.h:
	  rtpstats: fix assertion 'denom > 0' failed
	  gst_util_uint64_scale_int takes a gint as denom parameter
	  whereas ctx->clock_rate is a guint32.
	  It happens when gst_rtp_packet_rate_ctx_reset set clock_rate
	  to -1.
	  So just define clock_rate as gint like it is done in rtpsource.h
	  https://bugzilla.gnome.org/show_bug.cgi?id=784250

2017-06-28 14:05:27 -0500  Matt Fischer <matt.fischer@garmin.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Block recursive calls to resurect_buffer
	  When resurrecting a buffer, the subsequent free call can result
	  in the group-released handler being called again, which causes
	  a recursive loop.  This patch blocks the signal handler during
	  the time that it executes, ensuring that the loop will not occur.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759292

2017-06-20 16:39:36 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: Avoid deprecated ssl-ca-file property
	  SoupSession's ssl-ca-file property is deprecated. Use the recommended
	  tls-database property.
	  This is a bit more complex as it requires creating a GTlsFileDatabase
	  object for an absolute (!) path to the CA certificates file.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784005

2017-06-20 16:37:55 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: Avoid deprecated server ssl properties
	  The ssl-cert-file and ssl-key-file properties are deprecated. Use the
	  soup_server_set_ssl_cert_file function to load the files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784005

2017-06-20 16:34:41 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: Make ssl_cert/key_file static
	  Just a bit of cleanup.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784005

2017-06-20 16:28:35 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* tests/files/test-cert.pem:
	  tests: souphttpsrc: Update test-cert.pem
	  Recent GnuTLS disregards the Common Name and only looks at the Subject
	  Alternative Name extension. Since our test-cert has no SAN extension,
	  validation fails.
	  Generate a new certificate with SAN. In addition to 127.0.0.1, for good
	  measure make it valid for localhost and ::1, too.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784005

2017-06-29 15:22:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Allow any type of proxy
	  Currently we only allowed HTTP proxy. Don't filter for the scheme, just check
	  if it looks like an URI. Soup will warn if the URI is invalid or if
	  proxy protocol is not supported. This enables using SOCKS 4/5 which is
	  directly implemented into GIO.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783012

2017-05-24 15:07:51 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: increase by one the number of allocated buffers
	  Increasing this number fix a buffer starvation problem I'm hitting
	  with a "v4l2src ! kmssink" pipeline.
	  kmssink requests 2 buffer as it keeps a reference on the last rendered
	  one. So we were allocating 3 buffers for the pipeline.
	  Once the first 2 buffers have been pushed we ended up with:
	  - one buffer queued in v4l2
	  - one being pushed
	  - one kept as last rendered
	  If this 3rd buffer is released after that v4l2 used the first one to
	  capture we end up with a buffer starvation problem as no buffer is currently
	  queued in v4l2 for capture.
	  Fixing this by adding one extra buffer to the pipeline so when one
	  buffer is being pushed downstream the other can already be queued to
	  capture the next frame.
	  We were already adding 3 buffers if downstream didn't reply to the
	  allocation query. I reduced this number to 2 to compensate the extra
	  buffer which is now always added.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783049

2017-06-29 18:59:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Create send/recv mutexes once, not on every connect()
	  Also fixes a crash caused by freeing an uninitialized mutex in an error
	  case.
	  https://bugzilla.gnome.org//show_bug.cgi?id=784282

2017-06-27 18:20:17 -0500  Matt Fischer <matt.fischer@garmin.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix memory leak with dmabuf
	  This patch fixes a memory leak that is caused if the dmabuf file
	  descriptor dup fails.  Previously, _cleanup_failed_alloc() would
	  not unref the memory because mems_allocated had not yet been
	  incremented.
	  https://bugzilla.gnome.org/show_bug.cgi?id=784302

2017-06-28 19:46:04 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/qtdemux_types.c:
	  qtdemux: specify '_swr' atom as a container atom
	  ... so it is parsed as an mp4 style metadata atom as written by muxer

2017-06-27 20:14:57 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/atoms.c:
	  qtmux: initialize mdhd language code as undefined

2017-06-22 15:34:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: Add a faststart-min-packets property
	  When set this property will allow the jitterbuffer to start delivering
	  packets as soon as N most recent packets have consecutive seqnum. A
	  faststart-min-packets of zero disables this feature. This heuristic is
	  also used in rtpsource which implements the probation mechanism and a
	  similar heuristic is used to handle long gaps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769536

2017-06-23 16:18:57 -0400  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* meson.build:
	  meson: Allow using glib as a subproject

2017-06-26 11:09:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/examples/audiofx/meson.build:
	* tests/examples/cairo/meson.build:
	* tests/examples/equalizer/meson.build:
	* tests/examples/jack/meson.build:
	* tests/examples/level/meson.build:
	* tests/examples/meson.build:
	* tests/examples/rtp/meson.build:
	* tests/examples/shapewipe/meson.build:
	* tests/examples/spectrum/meson.build:
	* tests/examples/v4l2/meson.build:
	* tests/meson.build:
	  meson: build examples
	  https://bugzilla.gnome.org/show_bug.cgi?id=784134

2017-06-26 09:47:55 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: fix with-package-name option
	  https://bugzilla.gnome.org/show_bug.cgi?id=784082

2017-06-26 09:38:46 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/meson.build:
	  meson: tests: icles: simplify build file

2017-06-26 00:22:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/meson.build:
	* tests/meson.build:
	  meson: build tests/icles/
	  https://bugzilla.gnome.org/show_bug.cgi?id=784134

2017-06-19 21:13:42 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: correctly calculate overall first_ts to ensure stream sync
	  ... by minding and compensating for the dts_adjustment that may have
	  been introduced in the PTS timeline.

2017-06-10 15:14:41 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: track highest known cluster position and time
	  ... to use as a fallback initial duration estimate and to provide for
	  interpolation when scanning for position.

2017-06-10 13:46:20 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: improve and simplify searching for cluster and position
	  ... avoiding inefficiency proportional to file size

2017-06-08 16:55:29 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: increase chunk size when scanning for cluster

2017-06-08 16:39:06 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: maintain variable state when searching for position
	  ... so skipping to next cluster happens efficiently

2017-06-24 00:21:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/meson.build:
	* ext/raw1394/meson.build:
	  meson: build raw1394 plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=784134

2017-06-23 23:50:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/aalib/meson.build:
	* ext/meson.build:
	  meson: build aalib plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=784134

2017-06-23 23:38:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/libcaca/meson.build:
	* ext/meson.build:
	  meson: build caca plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=784134

2017-06-23 20:01:59 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update for git master

2017-06-23 19:52:04 +0100  Tim-Philipp Müller <tim@centricular.com>

	* README:
	* configure.ac:
	* meson.build:
	* po/POTFILES.in:
	* sys/Makefile.am:
	* sys/meson.build:
	* sys/sunaudio/Makefile.am:
	* sys/sunaudio/gstsunaudio.c:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiomixer.h:
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixeroptions.c:
	* sys/sunaudio/gstsunaudiomixeroptions.h:
	* sys/sunaudio/gstsunaudiomixertrack.c:
	* sys/sunaudio/gstsunaudiomixertrack.h:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosink.h:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/sunaudio/gstsunaudiosrc.h:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/sunaudio.c:
	* tests/check/meson.build:
	  sys: remove sunaudio plugin
	  Even though hooked up to the build system, it's clear that no one
	  has ever built or used this with GStreamer 1.x. It wants to link
	  against libgstinterfaces, which no longer exists. And uses 0.10-style
	  raw audio caps. And the last meaningful change was done in 2009.
	  Let's just remove it.

2017-06-23 19:35:28 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/meson.build:
	* sys/oss4/meson.build:
	  meson: build oss4 plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=784134

2017-06-23 19:23:52 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/meson.build:
	* sys/oss/meson.build:
	  meson: build oss plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=784134

2017-06-22 11:38:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Actually use the receive lock when receiving, not the send lock

2017-06-22 01:01:40 +1000  Jan Schmidt <jan@centricular.com>

	* tests/examples/qt/qmlsink/CMakeLists.txt:
	  qmlsink example: Add CMakeLists.txt
	  Make it possible to build using cmake instead of qmake

2017-06-22 01:01:40 +1000  Jan Schmidt <jan@centricular.com>

	* ext/qt/qtitem.cc:
	  qt: Remove misleading reference to GTK in qtitem.cc

2017-06-15 11:46:54 -0400  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* ext/flac/gstflactag.c:
	  flactag: Fix warning with the newly added GstStateChange values
	  https://bugzilla.gnome.org/show_bug.cgi?id=783798

2017-06-15 19:09:26 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: do not checksum the stream id
	  https://bugzilla.gnome.org/show_bug.cgi?id=783307

2017-06-15 23:31:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/qtdemux.c:
	  qtmux: add support for muxing PNG
	  Demuxer already supported it.

2017-06-15 10:40:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Use a mutex for protecting against concurrent send/receives
	  We currently send data to the RTSP connection from multiple threads:
	  whenever a command is to be handled and whenever RTCP is generated. This
	  can cause data corruption or worse if both happen at the same time.
	  As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive()
	  calls with a mutex. While this means that we hold a mutex during the IO
	  operation, this is not actually a problem as the IO operation can be
	  interrupted (gst_rtsp_connection_flush()) at any time and is blocking by
	  itself anyway.

2017-06-15 11:50:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: Un-merge the last two stsc entries after serializing
	  The last entry will most likely get new samples added to it in "robust"
	  muxing mode, changing the samples_per_chunk and thus making it wrong to
	  keep the last two entries merged. It will run into an assertion later
	  when adding a new sample to the chunk.
	  Thanks to gdiener@cardinalpeak.com for the analysis of the bug and
	  proposal for a solution.

2017-06-14 00:09:25 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Actually clip to upstream size instead of size of the data chunk
	  There might be other chunks after the data chunk, so clipping the chunk
	  size with the data size can lead to a negative number and all following
	  calculations go wrong and cause crashes or worse.
	  This was introduced in 3ac119bbe2c360e28c087cf3852ea769d611b120.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783760

2017-06-13 17:40:19 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: Drop allocation queries
	  They can cause us to deadlock, while we're waiting for a new frame and
	  upstream is waiting for the allocation query to be answered before
	  sending a frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=783753

2017-06-01 02:03:27 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: uniquify stream ids
	  https://bugzilla.gnome.org/show_bug.cgi?id=783307

2017-06-07 12:47:59 -0400  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/check/meson.build:
	  meson: Do not use path separator in test names
	  Avoiding warnings like:
	  WARNING: Target "elements/audioamplify" has a path separator in its name.

2017-06-06 11:29:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/examples/v4l2/camctrl.c:
	  Fix v4l2 example

2017-06-05 16:55:13 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: remove not needed code
	  remove not needed code about res variable.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783422

2017-06-02 14:01:17 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Make sure min_buffers is valid
	  When upstream does no use the v4l2videoenc pool, we need to activate
	  that internal pool. Though, we relied the driver to provide a minimum
	  required buffer, which Qualcomm Venus driver don't currently provide.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783361

2017-06-02 11:30:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: fix caps leak

2017-05-26 16:30:06 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: simplify buffer accumulation control flow
	  There is no difference between pushing out a buffer directly
	  with gst_rtp_base_depayload_push() and returning it from the
	  process function. The base class will just call _depayload_push()
	  on the returned buffer as well.
	  So instead of marshalling buffers through three layers and back,
	  just push them from one place in handle_nal() and always return
	  NULL from the process vfunc. This simplifies the code a little.
	  Also rename _push_fragmentation_unit() to _finish_fragmentation_unit()
	  for clarity. Push sounds like it means being pushed out, whereas
	  it might just be pushed into an adapter.
	  This change has the side-effect that multiple NALs in a single STAP
	  (such as SPS/PPS) may no longer be pushed out as a single buffer if
	  we output NALs in byte-stream format (i.e. not aggregate AUs), but
	  that shouldn't really make any difference to anyone.

2017-05-30 22:23:10 +0200  Juan Navarro <juan.navarro@gmx.es>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: print value of unknown RTCP Payload Type
	  This adds printing the actual value of any unknown RTCP PT
	  to the already existing WARNING log message.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783248

2017-05-26 17:52:19 +0200  Edward Hervey <edward@centricular.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Don't leak VideoCodecState
	  CID #1409852

2017-05-26 17:48:01 +0200  Edward Hervey <edward@centricular.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Remove un-needed variable check
	  if pad wasn't present by now everything would have broken before
	  CID #1409854

2017-05-25 15:26:37 +0200  Piotr Drąg <piotrdrag@gmail.com>

	* po/POTFILES.in:
	  po: update POTFILES
	  https://bugzilla.gnome.org/show_bug.cgi?id=783093

2017-05-25 10:09:04 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/qt/qtwindow.cc:
	  glframebuffer: check frame buffer status need use specific fbo target
	  https://bugzilla.gnome.org/show_bug.cgi?id=783065

2017-05-24 14:19:27 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2videoenc: Remove unused function

2017-05-21 15:29:11 +0200  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/types-compat.h:
	  v4l2: Don't redefine __bitwise if already set
	  https://bugzilla.gnome.org/show_bug.cgi?id=728438

2017-05-23 14:40:56 -0400  Ayaka <ayaka@soulik.info>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2h264enc.c:
	* sys/v4l2/gstv4l2h264enc.h:
	* sys/v4l2/gstv4l2videoenc.c:
	* sys/v4l2/gstv4l2videoenc.h:
	* sys/v4l2/meson.build:
	  v4l2: Add Video Encoder support
	  This implements H264 encoding support using generic V4L2 interface. It is
	  reported to work with Samsung MFC driver, IXM.6 CODA driver and
	  Qualcomm mainline Venus driver. Other platform should be supported as
	  none of this work is platform specific.
	  The implementation consist of a GstV4l2VideoEnc base class, which
	  implements the core streaming functionality. This base class is implemented
	  by GstV4l2H264Enc class that implements the caps negotiation specific to
	  H264 profiles and level. This implementation supports hardware with multiple
	  H264 encoder. Though, to make it simplier to use, the first discovered H264
	  encoder will be named v4l2h264enc. Other encoder found during discovery will
	  have a unique name like v4l2video0h264enc.
	  This work is the combined work of multiple developpers in the last 3
	  years. Thanks to all of the contributors:
	  Ayaka <ayaka@soulik.info>
	  Frédéric Sureau <frederic.sureau@vodalys.com>
	  Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
	  Nicolas Dufresne <nicolas.dufresne@collabora.com>
	  Pablo Anton <pablo.anton@vodalys-labs.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=728438

2017-05-23 14:36:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Remove unused forward declaration
	  https://bugzilla.gnome.org/show_bug.cgi?id=728438

2015-10-05 16:30:46 +0100  Ayaka <ayaka@soulik.info>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2pool: Fix wrong error message
	  https://bugzilla.gnome.org/show_bug.cgi?id=728438

2015-10-05 16:20:19 +0100  Ayaka <ayaka@soulik.info>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: increase pre-allocated encoded buffer size
	  As of today, the MFC encoder often need to exceed that 1 MB
	  size for encoded buffer we fixed earlier for decoding.
	  https://bugzilla.gnome.org/show_bug.cgi?id=728438

2017-05-24 16:32:30 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpopusdepay.c:
	  rtpopusdepay: minor perf improvements
	  Use the ::process_rtp_packet() vfunc to avoid mapping the
	  RTP buffer twice.
	  gst_rtp_buffer_get_payload_buffer() returns a new sub-buffer
	  which will always be writable, so no need to make it writable.

2017-05-24 16:14:54 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  rtp: opus: use existing utility funcs for copying/dropping metas
	  We had our own copies of those while the code was in -bad, but now
	  we can use the existing utility functions instead of re-implementing
	  them.

2017-05-24 12:57:10 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtputils.c:
	* gst/rtp/gstrtputils.h:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp9depay.c:
	* gst/rtp/gstrtpvp9pay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: cache meta tag quarks and add more utility functions for metas
	  Every g_quark_from_static_string() is a hash table lookup serialised
	  on the global quark lock in GLib. Let's just look up the two quarks
	  we need once and cache them locally for future use. While we're at it,
	  add new utility functions for the two most commonly used tags
	  (audio + video). Make first argument a gpointer so we don't have to
	  cast and make the code ugly. These are used for logging purposes
	  only anyway.

2017-05-24 11:33:05 +0530  vijay <vijay.palaniswamy@in.bosch.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse : Fix, Caps were not set while reusing aacparse
	  While reusing aacparse caps were not set.This fix enables aacparse to reuse in same pipeline.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783027

2017-05-21 17:45:34 +0100  Tim-Philipp Müller <tim@centricular.com>

	* Makefile.am:
	* config.h.meson:
	* meson.build:
	  meson: don't need config.h.meson any longer

2017-05-21 15:26:12 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/qt/gstqsgtexture.cc:
	* ext/qt/gstqsgtexture.h:
	  qmlglsink: Add dummy texture that is shown as placeholder for NULL buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=782917

2017-04-24 16:55:22 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2send: use non-blocking I/O and a configurable network operations timeout
	  This allows timing out on network errors much earlier
	  (currently it takes ~15min to timeout) and we can still
	  unlock and change state in the meantime.
	  https://bugzilla.gnome.org/show_bug.cgi?id=571722

2017-05-21 10:37:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/taglib/meson.build:
	* meson.build:
	  meson: make C++ compiler optional
	  It's only needed for the taglib plugin which is optional.

2017-05-21 10:33:43 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/multifile.vproj:
	  multifile: remove some cruft

2017-05-20 17:09:52 +0200  Josep Torra <jtorra@oblong.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: fixes playback of mono streams with no channel-mask field in caps
	  Fixes a negotiation error seen when trying to playback of a .MOV file with
	  a mono AAC audio stream decoded by avcdec_aac that doesn't set channel-mask
	  field but sink was requiring channel-mask=0x3.

2015-09-06 20:49:59 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: Push tag event to both pads
	  Tags are pushed to "videosrcpad"/"audiosrcpad" in
	  gst_dvdemux_add_pad() method, however they will be NULL
	  in this method, hence tags are not pushed.
	  Instead, send tag event to "pad" created gst_dvdemux_add_pad().
	  Signal no-more-pads when both pads are created
	  https://bugzilla.gnome.org/show_bug.cgi?id=743657

2017-05-20 14:53:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* meson_options.txt:
	* tests/check/elements/autodetect.c:
	  meson: add options to set package name and origin
	  https://bugzilla.gnome.org/show_bug.cgi?id=782172

2017-05-20 11:40:33 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: fix property name in example pipeline
	  Since the move from CVS the property name of the documentation example
	  has been filename instead of location. Users trying the gst-launch
	  command as is will get:
	  no property name "filename" in element
	  Fixing it.

2017-05-20 11:13:40 +0200  Josep Torra <jtorra@oblong.com>

	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideo: fix macOS 10.12 deprecation warnings
	  Add #defines to allow older versions of macOS to use the new constant names.

2017-05-13 09:05:57 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  isomp4: Safely ignore [skip] atoms
	  Instead of warning about them

2017-05-18 15:23:14 +0300  Simon Himmelbauer <shimmelbauer@toolsonair.com>

	* ext/qt/gstqtglutility.cc:
	  qt: Use GST_GL_HAVE_PLATFORM_CGL instead of GST_GL_HAVE_PLATFORM_COCOA
	  The latter is not used/available anymore since years. Also fix a typo
	  in the include path for the Cocoa GL display header.

2017-05-18 15:10:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Make session sharing thread-safe on our side
	  https://bugzilla.gnome.org/show_bug.cgi?id=780140

2017-05-18 10:53:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audiofx/gststereo.c:
	  stereo: fix typo in plugin description

2017-05-18 10:43:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/shout2/gstshout2.c:
	* gst/audiofx/gstscaletempoplugin.c:
	  Fix up package name and origin in some plugins

2017-05-15 19:51:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	  gst: Clear floating flag in constructor of all GstObject subclasses that are not owned by any parent
	  https://bugzilla.gnome.org/show_bug.cgi?id=743062

2017-05-15 14:22:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/raw1394/gst1394clock.c:
	  1394: Sink the clock reference in the constructor
	  This is now needed as GstClock does not do that internally anymore,
	  because that broke bindings.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743062

2017-05-17 10:58:05 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/qt/gstqtglutility.cc:
	  qml: Add EGL platform support for x11 backend
	  Add support for EGL platform when x11 is available. This can work
	  e.g. on imx6 platform.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782718

2017-04-28 23:05:35 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/pulse/pulseutil.h:
	  pulse: Accept MPEG 1 layer 3 version 2.5
	  https://bugzilla.gnome.org/show_bug.cgi?id=781929

2017-05-16 13:50:16 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	* ext/aalib/Makefile.am:
	* ext/cairo/Makefile.am:
	* ext/dv/Makefile.am:
	* ext/flac/Makefile.am:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/jack/Makefile.am:
	* ext/jpeg/Makefile.am:
	* ext/libcaca/Makefile.am:
	* ext/libpng/Makefile.am:
	* ext/pulse/Makefile.am:
	* ext/raw1394/Makefile.am:
	* ext/shout2/Makefile.am:
	* ext/soup/Makefile.am:
	* ext/speex/Makefile.am:
	* ext/taglib/Makefile.am:
	* ext/vpx/Makefile.am:
	* ext/wavpack/Makefile.am:
	* gst/alpha/Makefile.am:
	* gst/apetag/Makefile.am:
	* gst/audiofx/Makefile.am:
	* gst/audioparsers/Makefile.am:
	* gst/auparse/Makefile.am:
	* gst/autodetect/Makefile.am:
	* gst/avi/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debugutils/Makefile.am:
	* gst/deinterlace/Makefile.am:
	* gst/dtmf/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/equalizer/Makefile.am:
	* gst/flv/Makefile.am:
	* gst/flx/Makefile.am:
	* gst/goom/Makefile.am:
	* gst/goom2k1/Makefile.am:
	* gst/icydemux/Makefile.am:
	* gst/id3demux/Makefile.am:
	* gst/imagefreeze/Makefile.am:
	* gst/interleave/Makefile.am:
	* gst/isomp4/Makefile.am:
	* gst/law/Makefile.am:
	* gst/level/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/monoscope/Makefile.am:
	* gst/multifile/Makefile.am:
	* gst/multipart/Makefile.am:
	* gst/replaygain/Makefile.am:
	* gst/rtp/Makefile.am:
	* gst/rtpmanager/Makefile.am:
	* gst/rtsp/Makefile.am:
	* gst/shapewipe/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/spectrum/Makefile.am:
	* gst/udp/Makefile.am:
	* gst/videobox/Makefile.am:
	* gst/videocrop/Makefile.am:
	* gst/videofilter/Makefile.am:
	* gst/videomixer/Makefile.am:
	* gst/wavenc/Makefile.am:
	* gst/wavparse/Makefile.am:
	* gst/y4m/Makefile.am:
	* sys/directsound/Makefile.am:
	* sys/oss/Makefile.am:
	* sys/oss4/Makefile.am:
	* sys/osxaudio/Makefile.am:
	* sys/osxvideo/Makefile.am:
	* sys/sunaudio/Makefile.am:
	* sys/v4l2/Makefile.am:
	* sys/waveform/Makefile.am:
	* sys/ximage/Makefile.am:
	  Remove plugin specific static build option
	  Static and dynamic plugins now have the same interface. The standard
	  --enable-static/--enable-shared toggle are sufficient.

2017-05-16 14:07:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/twolame/Makefile.am:
	  Remove plugin specific static build option
	  Static and dynamic plugins now have the same interface. The standard
	  --enable-static/--enable-shared toggle are sufficient.

2017-05-16 14:07:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/lame/Makefile.am:
	  Remove plugin specific static build option
	  Static and dynamic plugins now have the same interface. The standard
	  --enable-static/--enable-shared toggle are sufficient.

2017-05-16 14:07:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/mpg123/Makefile.am:
	  Remove plugin specific static build option
	  Static and dynamic plugins now have the same interface. The standard
	  --enable-static/--enable-shared toggle are sufficient.

2017-05-16 14:05:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/Makefile.am:
	  Remove plugin specific static build option
	  Static and dynamic plugins now have the same interface. The standard
	  --enable-static/--enable-shared toggle are sufficient.

2017-05-16 14:05:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/qt/Makefile.am:
	  Remove plugin specific static build option
	  Static and dynamic plugins now have the same interface. The standard
	  --enable-static/--enable-shared toggle are sufficient.

2017-05-12 17:53:57 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Add alignment-threshold argument
	  If a non-reference stream is behind the reference stream by an amount of
	  time smaller than the alignment threshold (in nsec), it counts as being
	  after it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782563

2017-05-16 12:56:15 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Do not check timecode data for mp4 container
	  Timecode trak is only supported for mov right now, not for mp4. That
	  code would otherwise create an invalid trak if the muxed video contained
	  timecode metadata.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782684

2017-05-11 20:01:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: When accepting renegotiation, just return TRUE and change nothing
	  We only accept new caps if they are basically the same. We don't want to
	  reset anything as if the caps are new, otherwise various state could get
	  out of sync with the current run.

2017-05-11 19:21:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: In prefill mode, only pad buffers with > 0 sized memories as needed
	  Adding a 0-byte memory has not much effect.
	  Also add some debug output.

2017-05-10 15:58:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Lateness is in QT timescale, diff in GstClockTime
	  Print the right one in debug output to get meaningful numbers.

2017-05-10 14:31:40 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Error out if a gap edit list has to be written in prefill mode
	  We don't have any space reserved for this in the moov and the
	  pre-finalized moov would have broken A/V synchronization. Error out here
	  now

2017-05-10 11:42:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Calculate with reserved moov size instead of last moov size
	  We have some padding added after the initial moov, so a bigger updated
	  moov can be handled to some degree and is expected. Previously we just
	  ignored the padding and errored out in cases when the padding would've
	  just been enough.

2017-05-10 11:12:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Error out directly if sending filler data results in a flow error
	  CID 1405994

2017-05-09 16:02:43 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: In prefill mode, handle the case when only the first chunk was ever used

2017-05-09 09:47:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/qt/gstplugin.cc:
	  qmlgl: Make the plugin name match the pugin file name

2017-03-16 15:12:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Use a in-memory cookie jar by default in sessions we created
	  This ensures that cookies are stored and used as set by the server, and
	  shared with other souphttpsrc that use the same SoupSession.
	  https://bugzilla.gnome.org/show_bug.cgi?id=780140

2017-03-16 13:58:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Implement soup session sharing
	  souphttpsrc now shares its SoupSession with other elements in the
	  pipeline via GstContext if possible (session-wide settings are all the
	  defaults), or if the context was forced by the application.
	  This allows multiple souphttpsrcs to reuse connections, cookies, etc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=780140

2017-03-09 10:15:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Add new prefill recording mode
	  This sets up a moov with the correct sample positions beforehand and
	  only works with constant framerate, I-frame only streams.
	  Currently only support for ProRes and raw audio is implemented but
	  adding new codecs is just a matter of defining appropriate maximum frame
	  sizes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781447

2017-03-29 14:01:25 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Error out on discontinuities/gaps when muxing raw audio
	  When muxing raw audio, we have no way of storing timestamps but are just
	  storing a continuous stream of audio samples. If the difference between
	  the expected and the real timestamp becomes to big, we should error out
	  instead of silently creating files with wrong A/V sync.
	  https://bugzilla.gnome.org/show_bug.cgi?id=780679

2017-05-09 11:41:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: Set fb->priv to NULL after freeing just in case
	  https://bugzilla.gnome.org/show_bug.cgi?id=782359

2017-05-08 15:22:00 +0000  Dustin Spicuzza <dustin@virtualroadside.com>

	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: Use GstClock API instead of Sleep() for waiting
	  It's more accurate and allows cancellation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773681

2017-05-08 15:05:45 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/gstvp9dec.c:
	  vpx: fix build against older libvpx versions
	  Such as 1.3.0 as on raspbian.

2017-05-03 23:23:10 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix corner case causing large CPU usage
	  We were unnecessarily looping/goto-ing repeatedly when we had exactly
	  the amount of data as the free space, and also when the free space was
	  too small. This, as it turns out, is a very common scenario with
	  Directsound on Windows.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=773681
	  We have to do polling here because the event notification API that
	  Directsound exposes cannot be used with live playback since all events
	  must be registered in advance with the capture buffer, you cannot
	  add/remove them once playback has begun. Directsoundsrc had the same
	  problem.
	  See also: https://bugzilla.gnome.org/show_bug.cgi?id=781249

2017-05-03 23:31:00 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Clean up some debug logging
	  Don't need to print the function name, gstreamer does it for you.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773681

2017-05-06 22:30:20 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-ids.h:
	  matroskademux: improve index memory usage
	  Re-arrange order of index entry struct members to avoid padding
	  bytes in the middle of the struct, thus potentially reducing the
	  overall size of the struct and reducing memory used by the index.
	  On Linux x86_64 the size goes down from 32 bytes to 24 bytes for
	  each index entry.

2017-05-04 18:59:14 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	* meson.build:
	  Back to development

=== release 1.12.0 ===

2017-05-04 15:38:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.12.0

2017-05-04 15:07:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2017-05-04 13:47:20 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	  po: Update translations

2017-05-02 10:32:30 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix crash on mss stream caused by invalid stsd entry access
	  Since mss has no moov, default stsd entry should be created with media-caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782042

=== release 1.11.91 ===

2017-04-27 17:29:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.11.91

2017-04-27 15:58:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2017-04-27 15:28:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/LINGUAS:
	* po/el.po:
	* po/fur.po:
	  po: Update translations

2017-04-27 12:56:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't crash in debug output if stream==NULL
	  That case is correctly handled below but not in the debug output.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781270

2017-04-25 17:11:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't perform seeks with inconsistent seek values
	  If gst_segment_do_seek() fails, we shouldn't try seeking on that
	  resulting segment but just error out. Crashes further down the line
	  otherwise.

2017-04-24 20:27:49 +0100  Tim-Philipp Müller <tim@centricular.com>

	* common:
	  Automatic update of common submodule
	  From 60aeef6 to 48a5d85

2017-04-24 17:31:04 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: add test for rtph264depay avc/byte-stream output
	  Make sure avc output doesn't contain SPS/PPS inline, but
	  byte-stream output does.

2017-04-24 17:29:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: don't insert SPS/PPS inline for AVC output
	  SPS/PPS are in the caps in this case and shouldn't be in
	  the stream data.

2017-04-21 19:09:14 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Chain up to the parent class' provide_clock() implementation
	  If no clock was provided directly by rtspsrc. This behaviour was removed
	  by f8013487c91a6ffc552a4b25aa1a70f0bd5377f8 and results in rtspsrc not
	  providing the system clock via the rtpjitterbuffer.
	  As a result, if another element like an audio sink, provides a clock,
	  the pipeline would select that (when going to PAUSED/PLAYING again later).
	  Audio clocks usually don't progress in PAUSED, and thus our live source
	  won't be able to use the clock to produce data, making the sink never
	  preroll and everything is stuck.

2017-04-20 11:22:15 +0200  Jürgen Sachs <juergen.sachs@metz-ce.de>

	* gst/isomp4/qtdemux.c:
	  qtdemux: reset sample_description_id to default
	  Fixes stream where sample_description_id is specified in the tfhd
	  https://bugzilla.gnome.org/show_bug.cgi?id=778337

2017-04-20 13:16:24 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't use an explicit name for requesting audio pads
	  ... unless the muxer uses the same audio pad template name as
	  splitmuxsink. We can't request a pad called "audio_0" on a muxer that
	  wants pads to be "sink_%d".

2017-02-23 09:31:36 +0900  ChangBok Chae <changbok.chea@gmail.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: remove duplicated segment initialization
	  It's also done in gst_flv_demux_cleanup().
	  https://bugzilla.gnome.org/show_bug.cgi?id=779106

2017-04-20 20:17:35 +1000  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Correctly catch FLUSH events in probes
	  https://bugzilla.gnome.org/show_bug.cgi?id=767498

2017-04-19 12:28:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  Revert "rtpbin: pipeline gets an EOS when any rtpsources byes"
	  This reverts commit eeea2a7fe88a17b15318d5b6ae6e190b2f777030.
	  It breaks EOS in some sender pipelines, see
	  https://bugzilla.gnome.org/show_bug.cgi?id=773218#c20

2017-04-14 17:01:49 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Reset adapter in more discontinuity cases
	  In push mode we process as much as possible in the adapter. When we receive
	  a DISCONT buffer which we can't match to an actual sample (based on the existing
	  sample table) and there is still data remaining in the incoming adapter,there is
	  one of two cases happening:
	  1) We are doing reverse playback, in which case we should flush out all pending
	  data
	  2) We have leftover data from the previous incoming buffer... which we can't do
	  anything about.
	  For the second case, make sure we flush out the remaining data so that we can start
	  parsing again from scratch.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781319

2017-04-14 10:56:41 +0200  Edward Hervey <edward@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use GST_ELEMENT_ERROR_WITH_DETAILS
	  Allows the application to know the exact status code that was returned
	  by the server in a programmatic fashion.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781304

2017-04-16 18:47:56 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix leak on QtDemuxStreamStsdEntry
	  Fix unit test failure
	  https://bugzilla.gnome.org/show_bug.cgi?id=781362

2017-04-14 13:38:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix timescale of timecode tracks
	  They should have ideally the same timescale of the video track, which we
	  can't guarantee here as in theory timecode configuration and video
	  framerate could be different. However we should set a correct timescale
	  based on the framerate given in the timecode configuration, and not just
	  use the framerate numerator.

2017-04-13 13:25:06 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Properly reset demuxer when all streams are EOS
	  Make sure offset and neededbytes are properly resetted when all
	  streams are EOS in push-mode.
	  Avoids cases when some data might still be pushed by upstream (because
	  it didn't yet see the resulting GST_FLOW_EOS yet) and qtdemux gets
	  completely lost.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781266

2017-04-13 08:00:30 +0200  Edward Hervey <edward@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Make more usage of error macro
	  And make sure we actually use the provided soup_msg argument in the macro

2017-03-08 15:01:13 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* gst/audiofx/gststereo.c:
	  docs: Port all docstring to gtk-doc markdown

2017-03-08 15:01:13 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtksink.c:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstwidget.c:
	  docs: Port all docstring to gtk-doc markdown

2017-04-12 18:46:53 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/meson.build:
	  meson: Print message when disabling taglib on MSVC

2017-04-12 13:26:59 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't forget to update pad->last_buf
	  buf is the current pad->last_buf value. If ever it gets copied/unreffed,
	  we need to make sure to write back the new  pointer to the last_buf
	  variable.
	  Fixes using wrong pointer values in the case of decrasing DTS value

2017-04-12 11:33:05 +0200  Edward Hervey <edward@centricular.com>

	* tests/check/elements/.gitignore:
	  tests: Add vp9enc to gitignore

2017-04-11 13:41:48 +0200  Jürgen Sachs <juergen.sachs@metz-ce.de>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix: sample description index override in tfhd not evaluated
	  https://bugzilla.gnome.org/show_bug.cgi?id=778337

2017-04-12 11:03:24 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add out-of-bound check
	  Make sure we don't read invalid memory

2016-04-27 12:17:37 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: move parsing of tkhd out of stsd entry loop
	  It needs only to be read once.

2016-04-07 12:23:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check for a different stsd entry before pushing a sample
	  Before pushing a sample, check if there was a change in the current
	  stsd entry. This patch also assumes that the first stsd entry is
	  used as default for the first sample. It might cause an uneeded
	  caps renegotiation when this isn't the case.

2016-04-06 12:55:18 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: parse all stsd entries
	  stsd can have multiple format entries, parse them all.
	  This is required to play DVB DASH profile that uses multiple entries
	  to identify the different available bitrates/options on dash streams
	  The stream format-specific data is not stored into QtDemuxStreamStsdEntry

2016-04-05 14:34:00 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: rework stsd sample entries access
	  Instead of using the stsd as a base pointer, use the actual stsd
	  entry as the stsd can have multiple entries. This is rarely used
	  for file playback but is a possible profile with in DVB DASH specs.
	  This still doesn't support stsd with multiple entries but makes it
	  easier to do so.

2016-04-05 18:00:10 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: get stsd child by index instead of type
	  There might be multiple children with the same type

2017-04-07 16:33:18 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests/check/rtprtx: add checks for rtprtxqueue's max-size-{time,packets} properties
	  https://bugzilla.gnome.org/show_bug.cgi?id=780867

2017-04-04 17:33:31 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtprtxqueue: implement handling of the max-size-time property
	  https://bugzilla.gnome.org/show_bug.cgi?id=780867

2017-04-10 23:49:06 +0100  Tim-Philipp Müller <tim@centricular.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From 39ac2f5 to 60aeef6

2017-04-10 08:56:00 +0000  Todor Tomov <todor.tomov@linaro.org>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2object: Copy timestamp when importing buffers
	  This is needed for V4L2_OUTPUT interface, and is harmless of
	  V4L2_CAPTURE interfaces. This will fix timestamp in cases like:
	  v4l2src io-mode=dmabuf ! v4l2videoNenc output-io-mode=dmabuf-import !  ...
	  Same apply for userptr.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781119

2017-04-10 15:55:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix last_dts tracking for raw audio and similar formats
	  Accumulate the durations directly and don't scale yet another time by
	  the number of samples.

2017-04-07 10:48:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/elements/splitmux.c:
	  tests: fix leak in splitmux test
	  https://bugzilla.gnome.org/show_bug.cgi?id=781025

2017-04-07 15:29:43 +0800  Lyon Wang <lyon.wang@nxp.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Scale GAP event timestamp and duration like for buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=781008

2017-02-17 10:01:08 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videodec.h:
	  v4l2dec: Fix race when going from PAUSED to READY
	  Running `gst-validate-launcher -t validate.file.playback.change_state_intensive.vorbis_vp8_1_webm`
	  on odroid XU4 (s5p-mfc v4l2 driver) often leads to:
	  ERROR:../subprojects/gst-plugins-good/sys/v4l2/gstv4l2videodec.c:215:gst_v4l2_video_dec_stop: assertion failed: (g_atomic_int_get (&self->processing) == FALSE)
	  This happens when the following race happens:
	  - T0: Main thread
	  - T1: Upstream streaming thread
	  - T2. v4l2dec processing thread)
	  [The decoder is in PAUSED state]
	  T0. The validate scenario runs `Executing (36/40) set-state: state=null repeat=40`
	  T1- The decoder handles a frame
	  T2- A decoded frame is push downstream
	  T2- Downstream returns FLUSHING as it is already flushing changing state
	  T2- The decoder stops its processing thread and sets `->processing = FALSE`
	  T1- The decoder handles another frame
	  T1- `->process` is FALSE so the decoder restarts its streaming thread
	  T0- In v4l2dec-> stop the processing thread is stopped
	  NOTE: At this point the processing thread loop never started.
	  T0- assertion failed: (g_atomic_int_get (&self->processing) == FALSE)
	  Here I am removing the whole ->processing logic to base it all on the
	  GstTask state to avoid duplicating the knowledge.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778830

=== release 1.11.90 ===

2017-04-07 16:31:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.11.90

2017-04-07 15:18:11 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2017-04-07 15:06:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	  po: Update translations

2017-04-06 12:01:00 +0200  Edward Hervey <edward@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: streamline and improve AudioSpecificConfig parsing
	  AudioSpecifigConfig is used in a variety of AAC streams but was
	  being parsed differently. Instead, make everyone use the same parsing.
	  * Remove unused 'bits' field (it was always set to 0 if present)
	  * Add proper GAConfig parsing (to know the  number of samples per frame
	  if present).
	  Fixes wrong rate/channels configuration in streams coming from qtdemux
	  https://bugzilla.gnome.org/show_bug.cgi?id=780966

2017-04-05 09:46:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Fix 32bit only printf format
	  The previous patch was using %llu for 64bits printf, which is 32bit
	  specific. We also trace the latency in time human readable form now.

2016-03-16 16:22:48 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: set streamparm for outputs that support it
	  Without a specified framerate from the sink, the decoder frame interval
	  should be set using the framerate of the encoded video stream.
	  Therefore, the v4l2object should be able to change the framerate on the
	  output if the V4L2 device accepts it.
	  This is also necessary for mem2mem encoders so that their bitrate
	  calculation code may work correctly and they may report the correct
	  frame duration on the capture queue.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779466

2016-03-16 16:24:55 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: only set latency if the frame duration is valid
	  If the duration of the v4l2object is GST_CLOCK_TIME_NONE, because the
	  sink did not specify a framerate in the caps and the driver accepts the
	  framerate, the decoder element uses GST_CLOCK_TIME_NONE to calculate and
	  set the element latency.
	  While this is a bug of the capture driver, the decoder element should
	  not use the invalid duration to calculate a latency, but print a warning
	  instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779466

2016-11-23 12:17:55 -0500  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Block in preroll_wait on unlock
	  The correct behaviour of anything stuck in the ->render() function
	  between ->unlock() and ->unlock_stop() is to call
	  gst_base_sink_wait_preroll() and only return an error if this returns an
	  error, otherwise, it must continue where it left off!
	  https://bugzilla.gnome.org/show_bug.cgi?id=774945

2017-04-05 15:55:20 +1000  Jan Schmidt <jan@centricular.com>

	* ext/vpx/gstvp9dec.c:
	  vp9dec: Add warnings for unsupported frame formats
	  At least output an element warning on the bus when we
	  encounter a frame format GStreamer doesn't currently support.

2017-04-04 17:55:13 +0200  Edward Hervey <edward@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Handle Parametric Stereo with HE-AAC(v2)
	  According to ISO/IEC:14496-2:2009 , in the case of HE-AACv2 (audioObjecType
	  29) parametric stereo is used (a single mono track is used and then
	  transformations are applied to it to provide a stereo output).
	  We therefore report two channels in the case where there is one reported
	  in the audioChannelConfiguration.
	  Fixes the various issues where a demuxer would report two channels, but
	  then the parser would say there's only one channel, and then the decoder
	  would output two channels.

2017-04-04 15:22:25 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Simplify buffer refcounting in add_buffer() and remove unneeded NULL checks

2017-04-04 15:08:33 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Select the best pad based on the cached last_buf if any
	  last_buf is the one we're going to write next, not buf. As such we
	  should check timestamps against that one if there is one to select the
	  earliest pad.
	  Also remember the currently selected pad in the very beginning when
	  storing the first last_buf.
	  This both solves some edge cases where not the correct next pad was
	  selected corresponding to the target interleave.

2017-04-04 15:07:40 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Error out immediately if a timecode is to be written but downstream return not-OK

2017-04-03 11:34:49 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Update variables before early exit
	  This is an update of d78d5896272d78df41e696fac929e7dfb3bb3dfa
	  We still exit as early as possible in case of non-ok/non-unlinked combined
	  flow, but we first make sure that we update the internal position variables.
	  This ensures that if upstreams "ignores" the flow return (and carries on pushing),
	  we don't end up processing data with completely bogus variables/positions.

2017-03-24 00:11:13 +1300  Douglas Bagnall <douglas@halo.gen.nz>

	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	  interleave: avoid using uninitialised ordering_map
	  If self->channel_positions == NULL (which seems unlikely),
	  self->default_channels_ordering_map will be used unintialised.
	  We avoid that by keeping track of the channel_mask, which is set when
	  the ordering map is initialised.
	  https://bugzilla.gnome.org/show_bug.cgi?id=780331

2017-03-23 23:56:31 +1300  Douglas Bagnall <douglas@halo.gen.nz>

	* gst/interleave/interleave.c:
	  interleave: don't overflow channel map with >64 channels
	  When there are more than 64 channels, we don't want to exceed the
	  bounds of the ordering_map buffer, and in these cases we don't want to
	  rempa at all. Here we avoid doing that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=780331

2017-03-28 14:23:16 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/check/meson.build:
	  meson: Use get_pkgconfig_variable instead of calling pkg-config ourself
	  It is avalaible in meson 0.36 which is now are requirement

2017-03-28 14:22:41 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* pkgconfig/gstreamer-plugins-good.pc.in:
	* pkgconfig/meson.build:
	  pkgconfig: Do not ever build an installed .pc file

2017-03-28 11:15:53 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/check/meson.build:
	  meson: test: Fix environment object usage

2017-03-28 11:14:47 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* meson.build:
	* pkgconfig/gstreamer-plugins-good.pc.in:
	* pkgconfig/meson.build:
	  pkgconfig: Generate the pkg-config with meson too

2017-03-27 21:52:00 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: In gap mode, consider the mdat offset when calculating the remaining mdat size
	  The mdat generally does not start at offset 0, we have to include the
	  size of the moof and whatever else was in front of the mdat.

2017-03-27 11:43:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atomsrecovery.c:
	  atomsrecovery: Error out when fseek() fails instead of silently ignoring
	  CID 1403262

2017-03-23 22:13:05 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Also add videometa if there is padding to the right and bottom
	  https://bugzilla.gnome.org/show_bug.cgi?id=780478

2017-03-21 12:54:27 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: fix output segment and buffer DTS to correspond to the flattened PTS
	  https://bugzilla.gnome.org/show_bug.cgi?id=780347

2017-03-23 17:53:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Remove some unused variables

2017-03-23 15:01:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Remove a couple of unneeded levels of indentation

2017-03-22 18:18:40 +0000  Enrique Ocaña González <eocanha@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: distinguish TFDT with value 0 from no TFDT at all
	  TFDTs with time 0 are being ignored since commit 1fc3d42f. They're
	  mistaken with the case of not having TFDT, but those two cases
	  must be distinguished in some way.
	  This patch passes an extra boolean flag when the TFDT is present.
	  This is now the condition being evaluated, instead of checking for
	  0 time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=780410

2017-03-22 19:15:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Reset current chunk after writing out timecode
	  If we have multiple tracks with timecodes, or it's not the first track
	  that has timecodes, or not the first buffer, we already started a chunk
	  for media data. We now need to "close" that chunk because we wrote data
	  for the timecode track and a new chunk has to be started for the
	  original track the next time it has data.

2017-03-22 18:52:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Do timecode handling per track, not per muxer instance
	  There could be multiple video tracks with timecodes.

2017-03-22 00:38:51 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	  qtdemux: matroskademux: Ignore repeated seek events
	  Similar to what was done in adaptivedemux, ignore seek
	  events we've already handled - such as when they are received
	  on every srcpad of files with lots of streams.

2017-03-21 14:55:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  dashdemux: Update mdatleft from overall mdatsize and offset when observing a gap
	  Otherwise mdatleft will have a value calculated from the initial
	  mdatsize minus the parts of the stream that we saw, which is not
	  including all the parts of the stream that might've been skipped.

2017-03-20 17:03:32 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  docs: update two references to the removed 'mad' plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=776140

2017-03-20 12:03:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxqueue.c:
	  rtprtxqueue: add basic documentation and example pipelines
	  Mostly explaining the difference between rtprtxqueue and rtprtxsend.

2017-03-17 20:58:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/meson.build:
	  v4l2: Fix meson plugin shared object name
	  It didn't match between AutoMake and Meson, and the Meson name
	  didn't math the plugin name (video4linux2).

2017-03-16 18:20:54 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtprtxreceive: fix example pipelines and improve the documentation
	  https://bugzilla.gnome.org/show_bug.cgi?id=771383

2017-03-17 14:10:40 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: fix playback if sample number does not start at 0
	  This reverts commit 29b807685d3c962bbe8afe351c5dca97d59eb5e0, while
	  fixing the original breaking tests/check/pipelines/flacdec.

2017-03-17 11:30:04 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  Revert "flacparse: fix playback if sample number does not start at 0"
	  This breaks gst-validate on the build server (though not locally),
	  and a unit test, and I can't run unit tests right now for some
	  unrelated reason.
	  This reverts commit 0747b56f8e7f4731d67f8d13a4bdc453dde0fdf7.

2017-03-16 17:44:41 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: print the correct variable in debug statement
	  This debug statement is meant to print the time since the last (early)
	  RTCP transmission, not the last regular RTCP transmission (which also
	  happens to be set a few lines above to current_time, so the debug output
	  is just confusing)

2017-03-16 17:42:27 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: convert LOG message to TRACE
	  This is printed too often (for every chained buffer!) and just clutters the logs.

2017-03-16 14:58:45 +0100  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: fix warning message
	  https://bugzilla.gnome.org/show_bug.cgi?id=780105

2017-03-16 13:54:54 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: fix playback if sample number does not start at 0
	  https://bugzilla.gnome.org/show_bug.cgi?id=777738

2017-03-15 18:58:55 +0100  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsource: get clock-rate from pt if needed to generate SR
	  https://bugzilla.gnome.org/show_bug.cgi?id=780105

2017-03-16 13:52:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Include GStreamer souphttpsrc version in default User-Agent string

2017-03-16 00:41:44 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: fix crash with empty sprops-parameters
	  https://bugzilla.gnome.org/show_bug.cgi?id=780040

2017-03-11 21:20:40 -0800  Thiago Santos <thiagossantos@gmail.com>

	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/atomsrecovery.h:
	  atomsrecovery: also handle extra atoms after 'mdia' in a 'trak'
	  Take into account the atoms at the end of the 'trak' atom when
	  recovering it. So that its size (already computed and added in the trak
	  size) isn't making offsets wrong.
	  https://bugzilla.gnome.org/show_bug.cgi?id=771478

2017-03-11 12:56:33 -0800  Thiago Santos <thiagossantos@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: avoid fallthrough to moovrecovery failure section
	  Return before that to preserve our successfull results, otherwise no
	  moov recovery information would be written
	  https://bugzilla.gnome.org/show_bug.cgi?id=771478

2017-03-11 12:27:28 -0800  Thiago Santos <thiagossantos@gmail.com>

	* gst/isomp4/atomsrecovery.c:
	  atomsrecovery: expect more atom types at the headers
	  Skip more atoms at the header until it finds the 'mdat' to continue the
	  moov recovery
	  https://bugzilla.gnome.org/show_bug.cgi?id=771478

2017-03-14 16:42:25 -0400  Olivier Crête <olivier.crete@collabora.com>

	* Makefile.am:
	* configure.ac:
	* tests/examples/Makefile.am:
	* tests/examples/pulse/.gitignore:
	* tests/examples/pulse/Makefile.am:
	* tests/examples/pulse/pulse.c:
	  pulse example: Remove
	  That example only tested the property probe interface, which has been removed.
	  The same kind of thing can now be done with the generic gst-device-monitor tool.

2017-03-14 16:38:02 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	  v4l2: Remove unused macro

2017-03-14 16:35:25 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Remove unused definitions

2017-03-14 10:10:19 +0100  Emeric Grange <egrange@gopro.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtmux: add CineForm support
	  https://bugzilla.gnome.org/show_bug.cgi?id=780024

2017-03-14 15:09:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Only create new chunks if we have more than a single stream
	  There's no point in creating multiple chunks otherwise, it only wastes
	  some bytes for storing the chunk offsets.

2017-03-14 10:09:46 +0100  Emeric Grange <egrange@gopro.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add S16L support
	  https://bugzilla.gnome.org/show_bug.cgi?id=780022

2017-03-14 15:48:08 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/splitmux.c:
	  splitmux test: Use passed first/last timestamps
	  Don't hard-code the expected timestamp range, use the
	  values the caller is passing in.

2017-03-14 14:15:00 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gl: GL_ARRAY_BUFFER is not a part of VAO state
	  As a result we need to bind it on every draw in order to have the
	  correct state in the GL state machine.

2017-03-13 14:28:47 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtsrc.cc:
	  gl/format: use our own GL format enum's instead of gstvideo's
	  They can describe in more detail (such as component sizes) the requested format.

2017-03-12 11:42:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* Makefile.am:
	* docs/plugins/inspect/plugin-soup.xml:
	  Add old plugin names to cruft list
	  This will help fixing uninstalled setup. Also fix missing path
	  correction in one of the plugin xml.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779344

2016-12-15 12:38:40 +0100  Michael Dutka <mail@michael-dutka.de>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph265depay.c:
	  rtph264depay, rtph265depay: remove stray g_debug()
	  https://bugzilla.gnome.org/show_bug.cgi?id=779858

2017-03-10 11:24:14 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: init fourcc
	  Initialize the fourcc to 0 so that we can detect failure later.

2017-03-08 22:50:52 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/Makefile.am:
	* tests/check/elements/level.c:
	* tests/check/elements/rglimiter.c:
	  tests: Add missing LDADD for libm in tests using math.h
	  Also, remove the math.h include for the one that just prentend to need
	  it.

2017-03-08 22:15:46 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  Fix shout2 plugin doc generation
	  In the previous patch, we also renamed shout2send to shout2, so it does
	  not clash with it's feature. Though we forgot to rename it in the doc
	  reference. This patch also add a cruft detection on the xml that made me
	  miss this error.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779344

2017-03-04 11:03:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/Makefile.am:
	* ext/gtk/gstplugin.c:
	  Rename plugin filesnames to match plugin names
	  - libgstgtksink.so -> libgstgtk.so
	  - libgstteletextdec.so -> libgstteletex.so
	  - libgstcamerabin2.so -> libgstcamerabin.so
	  - libgstonvif.so -> libgstrtponvif.so (meson only)
	  - sdp -> sdpelem (avoid clash with libgstsdp)
	  - gstsiren -> siren
	  - libgstkmssink.so -> libgstkms.so
	  https://bugzilla.gnome.org/show_bug.cgi?id=779344

2017-03-04 10:52:47 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-shout2.xml:
	* ext/pulse/Makefile.am:
	* ext/pulse/meson.build:
	* ext/shout2/gstshout2.c:
	* ext/soup/Makefile.am:
	* ext/soup/meson.build:
	* sys/oss4/Makefile.am:
	  Fix plugin filenames to match plugin names
	  - libgstpulse.so becomes libgstpulseaudio.so
	  - libgstsouphttpsrc.so becomes libgstsoup.so
	  - libgstoss4audio.so becomes libgstoss4.so
	  https://bugzilla.gnome.org/show_bug.cgi?id=779344

2017-03-08 16:01:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: Free EDTS instead of just clearing it and setting it to NULL

2017-03-08 15:27:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix some memory leaks related to timecode tracks

2017-03-04 00:34:44 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/splitmux.c:
	  splitmux: Add unit test for reverse playback
	  Ensure that reverse playback works and generates the range
	  of timestamps (0-3s) we expect, in monotonically descending order.

2017-02-28 11:50:45 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Fix reverse playback
	  Fix the check for whether the start time of the segment has
	  been reached when playing in reverse. Otherwise, playback
	  stops after reaching the start of any file part, instead of
	  continuing until all parts within the segment have played

2017-02-22 03:01:31 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't lose crypto info on a new moof
	  We parse the next moof in advance of having pushed
	  all samples from the previous one in some cases, and
	  we'll still need the crypto info from the previous
	  fragment so keep around any unused crypto info entries
	  when adding new ones

2017-02-27 13:55:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Update modification times when sending the moov
	  https://bugzilla.gnome.org/show_bug.cgi?id=779422

2017-03-01 16:11:47 -0800  Michael Smith <mlrsmith@gmail.com>

	* gst/audioparsers/gstsbcparse.h:
	  sbcparse: Fix up values for allocation enumeration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779389

2017-02-28 13:10:50 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtprtxreceive: fix potential leak of old, unassociated, association requests
	  https://bugzilla.gnome.org/show_bug.cgi?id=722560

2017-02-28 15:47:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Don't increment -1 / unset indices
	  CID 1398545

2017-02-28 15:20:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Protect against NULL pointer dereference for streams without caps
	  CID 1363332

2017-02-28 12:57:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: Free mac on errors
	  CID 1212149

2017-02-28 12:45:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: Add missing break to for loop

2017-02-28 11:02:54 +0100  Edward Hervey <edward@centricular.com>

	* tests/check/Makefile.am:
	  check: Fix splitmux test CFLAGS
	  Needs to know where the gstapp headers are

2017-02-27 21:02:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix compilation with gcc 7
	  qtdemux.c: In function ‘qtdemux_parse_samples’:
	  qtdemux.c:8450:39: error: ‘*’ in boolean context, suggest ‘&&’ instead [-Werror=int-in-bool-context]
	  if (stream->samples_per_frame * stream->bytes_per_frame) {
	  ~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~

2017-02-27 21:01:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: Fix compilation with gcc 7
	  gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_reset’:
	  gstmpegaudioparse.c:209:3: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size]
	  memset (mp3parse->xing_seek_table_inverse, 0, 256);
	  ^~~~~~
	  gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_handle_first_frame’:
	  gstmpegaudioparse.c:951:7: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size]
	  memset (mp3parse->xing_seek_table_inverse, 0, 256);
	  ^~~~~~

2017-02-27 19:31:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: When getting new headers, replace the old version of them
	  This prevents storing an infinite amount of e.g. comment headers if they
	  come without a new initialization header in front of them. There can
	  only be one header of each type.

2017-02-27 19:25:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/rtp-payloading.c:
	  rtp-payloading: Add new test for Vorbis renegotiation
	  Check if encoding, payloading, depayloading and decoding works if the
	  stream configuration (and thus the headers) change.

2017-02-27 19:24:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvorbispay.c:
	  vorbispay: Only replace headers when receiving a new config header
	  If we also replace all headers when receiving any possibly following
	  comments header, we would throw away the config header before being able
	  to make use of it.

2017-02-23 12:11:15 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/Makefile.am:
	* tests/check/elements/splitmux.c:
	  tests: splitmux: add unit test for content with sparse streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=761086

2017-02-22 11:23:19 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxpartreader: ignore sparse streams when calculating the end offset of a part
	  A sparse stream's ending timestamp can be considerably smaller
	  than the ending timestamps of the other streams, which can lead
	  to skipping considerable time from the next part.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761086

2017-02-22 11:21:06 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxpartreader: identify sparse streams

2017-02-17 14:37:08 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qml: Add support for Vivante EGL FS windowing system
	  https://bugzilla.gnome.org/show_bug.cgi?id=778825

2017-02-25 21:47:03 -0300  Edgard Lima <edgard.lima@gmail.com>

	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* gst/audioparsers/gstamrparse.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726depay.h:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg726pay.h:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmadepay.h:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmudepay.h:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtppcmupay.h:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexdepay.h:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpspeexpay.h:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2colorbalance.c:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/gstv4l2vidorient.c:
	* sys/v4l2/gstv4l2vidorient.h:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	  Update Edgard Lima's email
	  https://bugzilla.gnome.org/show_bug.cgi?id=779230

2017-02-08 13:36:00 +0000  Andrew <nifigase@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: Don't always reset PTS to 0 after a gap
	  In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP
	  timestamps is more than (3 * jbuf->clock_rate) we call
	  rtp_jitter_buffer_reset_skew which resets pts to 0. So components down
	  the pipeline (playes, mixers) just skip frames/samples until pts becomes
	  equal to pts before gap.
	  In version 1.10.2 and before this checking was bypassed for packets with
	  "estimated dts", and gaps were handled correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778341

2017-02-24 15:59:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* meson.build:
	  meson: Update version

2017-02-24 15:37:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.11.2 ===

2017-02-24 15:07:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	  Release 1.11.2

2017-02-24 12:50:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2017-02-24 12:44:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	  po: Update translations

2017-02-10 20:50:17 +0900  Seungha Yang <sh.yang@lge.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Extract redirection uri on libsoup's restarted callback
	  Let libsoup handle redirection automatically.
	  And then, to figure out redirection uri, extract it on "restarted"
	  callback which will be fired before soup_session_send() is returned.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778428

2017-01-02 19:29:04 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Update image size when extrapolating
	  Update the image size according the amount of data we are going to
	  read/write. This workaround bugs in driver where the sizeimage provided
	  by TRY/S_FMT represent the buffer length (maximum size) rather then the expected
	  bytesused (buffer size).
	  https://bugzilla.gnome.org/show_bug.cgi?id=775564

2017-02-17 15:50:32 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix typo in _acquire_format() error messages
	  Fixes:
	  https://bugzilla.gnome.org/show_bug.cgi?id=778815

2017-02-07 17:27:56 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/matroskamux.c:
	* tests/check/elements/qtmux.c:
	  tests: matroskamux, qtmux: don't add codec_data buffers to template caps
	  streamheader and codec_data buffers fields are only meant to be
	  in the negotiated caps, not the template caps.
	  Fixes false-positive leaks of those buffers detected by the leaks
	  tracer, as template caps are static, and we decided to not include
	  code in gstreamer core to handle this unusual case of template caps
	  having buffers in them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768762

2017-02-09 12:46:54 +0000  Jochen Henneberg <jh@henneberg-systemdesign.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: Update and send out headers when new headers are received
	  The payloader needs to reset and update the vorbis config data which is
	  pushed on the network if it receives new headers, or at least, it may
	  have to do so.
	  Without this, the stream configuration could change without the
	  payloader sending the new configuration to the other side.

2017-02-15 14:48:58 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Change files on incompatible caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=761761

2017-02-15 16:35:01 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Reset ready_for_output on state change
	  https://bugzilla.gnome.org/show_bug.cgi?id=761761

2017-02-15 15:09:06 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Remove unused next_max_out_running_time
	  https://bugzilla.gnome.org/show_bug.cgi?id=761761

2017-02-15 15:07:32 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Remove unused muxed_out_time
	  https://bugzilla.gnome.org/show_bug.cgi?id=761761

2017-02-17 13:07:05 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  Revert "qtdemux: Always snap to the start of the keyframe"
	  This reverts commit 107902ec514bd826aa29d2298107e2c091e1c779.
	  This commit intended to ensure that keyframe seeks land at the
	  start timestamp of a keyframe, rather than in the middle of one,
	  but they cause trouble on files with sparse streams, or with
	  JPEG 'cover art' tracks that have only one or a few JPEG samples
	  with very long durations.
	  That's still desirable for doing seamless cutting of videos,
	  but needs a rethink for implementation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778690

2017-02-17 01:22:11 +1100  Jan Schmidt <jan@centricular.com>

	* gst/audiofx/audioecho.c:
	* gst/audiofx/audioecho.h:
	  audiofx/echo: added surround-delay and surround-mask
	  Add a new boolean surround-delay property that makes
	  audioecho just apply a delay to certain channels to create
	  a surround effect, rather than an echo on all
	  channels. This is useful when upmixing from stereo - for example.
	  Add a surround-mask property to control which channels
	  are considered surround sound channels when adding a
	  delay with surround-delay = true
	  Original patch from Jochen Henneberg <jh@henneberg-systemdesign.com>

2017-02-15 00:13:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Use IP_MULTICAST_ALL for filtering IPv4 packets if available
	  This goes around the inefficient control message based filtering and
	  does all the filtering kernel-side. Unfortunately this is Linux-only and
	  there is no IPv6 variant of it (yet).

2017-02-14 19:53:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	* Makefile.am:
	  meson: dist meson build files
	  Ship meson build files in tarballs, so people who use tarballs
	  in their builds can start playing with meson already.

2017-02-10 10:53:05 +0100  Søren Juul <zpon.dk@gmail.com>

	* gst/icydemux/gsticydemux.c:
	* tests/check/elements/icydemux.c:
	  icydemux: reset tags on empty value
	  Some radio streams uses StreamTitle='' to reset the title after a
	  track stopped playing, e.g. while the host talks between tracks or
	  during news segments.
	  This change forces an empty tag object to be distributed if
	  StreamTitle or StreamUrl is received with empty value, thus allowing
	  downstream elements to get notified about this.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778437

2017-02-13 11:17:25 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Properly notify missing elements
	  If the srtp elements are not present, post a message on the bus
	  informing about the missing plugins.

2017-02-10 10:32:57 -0300  Juan Pablo Ugarte <ugarte@endlessm.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: mark singleton caps as "may be leaked" objects.
	  Set MAY_BE_LEAKED flag on static pads returned by gst_v4l2_object_get_*_caps()
	  functions. Made functions thread safe by using g_once_init[enter|leave]
	  funtions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778453

2017-02-09 14:18:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Remove now unused done label

2017-02-09 12:55:32 +0100  Nick Kallen <nickkallen@me.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: do not cache caps
	  Upstream elements like videoflip can transform caps, such as changing width and height.
	  When an imagefreeze downstream receives an ACCEPT_CAPS query it will NOW return
	  all caps that it can accept.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778389

2017-02-09 11:29:43 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Add a comment about how atom_trak_set_elst_entry() works

2014-08-22 09:55:43 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux: demote some log messages to TRACE level
	  Don't spam debug log with uninteresting stuff.

2017-02-08 17:24:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Clear edit lists every time we recalculate them
	  We recalculate them, so any old information has to be forgotten.
	  Otherwise we write invalid edit lists when writing headers multiple
	  times.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778330

2017-02-07 13:10:18 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxsrc: Allow for buffers before the segment when measuring
	  Used signed calculations when measuring the max_ts of an input
	  fragment, so as to calculate the correct duration and offset
	  when buffers have timestamps preceding their segment

2017-02-02 12:55:25 +0100  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: relate received FIRs and PLIs to source
	  This is needed in order to:
	  - Avoid ignoring requests for different media sources.
	  - Add SSRC field in the GstForceKeyUnit event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778013

2017-01-30 20:20:08 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: sanity check number of segments in edit list
	  Fixes crash with fuzzed file.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777940

2017-01-02 22:16:39 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Skip seeking query if upstream format is time
	  Don't need to querying byte-format seeking for time-format
	  upstream case
	  https://bugzilla.gnome.org/show_bug.cgi?id=776715

2016-12-01 12:47:08 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use upstream's StreamFlags if there are
	  When multiple demuxer's are used, upstream might want to indicate
	  default streams using GST_STREAM_FLAG_{SELECT, UNSELECT}
	  https://bugzilla.gnome.org/show_bug.cgi?id=775440

2017-01-27 16:14:16 +0200  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/isomp4/atoms.c:
	  qtmux: Timecode track fixes for STSD entry
	  The n_frames field (frames per second) should follow the nominal frame
	  rate for drop-frame timecodes.
	  Also, the trak's timescale (and duration, accordingly) should follow the
	  STSD entry's timescale and frame duration (fps_n and fps_d accordingly),
	  not the other way around.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777832

2017-01-19 11:08:11 +0100  Arnaud Vrac <avrac@freebox.fr>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: retry request on early termination from the server
	  Fix a regression introduced by commit 183695c61a54f1 (refactor to use
	  Soup's sync API). The code previously attempted to reconnect when the
	  server closed the connection early, for example when the stream was put
	  in pause for some time.
	  Reintroduce this feature by checking if EOS is received before the
	  expected content size is downloaded. In this case, do the request
	  starting at the previous read position.
	  https://bugzilla.gnome.org/show_bug.cgi?id=776720

2017-01-10 09:40:56 -0700  Matt Staples <staples255@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: find_stream_by_channel should ignore unconfigured streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=777101

2017-01-25 18:43:00 +0000  Brendan Shanks <brendan.shanks@teradek.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix debug typo and remove misleading warning
	  https://bugzilla.gnome.org/show_bug.cgi?id=777362

2017-01-25 20:56:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/rtp/client-PCMA.c:
	  rtp: Remove unused variable in example
	  client-PCMA.c:84:22: warning: unused variable 'isrc' [-Wunused-variable]
	  GObject *session, *isrc, *osrc;
	  ^

2017-01-25 19:21:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/qt/Makefile.am:
	  qt: The code requires at least C++11
	  ... and clang requires this to be specified on the commandline while gcc
	  nowadays defaults to C++11 or even newer.

2017-01-09 11:32:35 +0530  Rahul Bedarkar <rahul.bedarkar@imgtec.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: check for not NULL before clearing adapter
	  In case wavparse receives a manually injected FLUSH_STOP event
	  while operating in pull mode we get criticals because we'd try
	  to clear a NULL adapter.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777123

2017-01-24 19:23:44 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/check/meson.build:
	  meson: Properly use ':' for defining keywords

2017-01-17 16:41:58 +0100  Jean-Christophe Trotin <jean-christophe.trotin@st.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: reference memory before the buffer is queued
	  In gst_v4l2_allocator_qbuf(), the memory is referenced after the
	  buffer is queued. Once queued (VIDIOC_QBUF), the buffer might be handled
	  by the V4L2 driver (e.g. decoded) and dequeued (gst_v4l2_allocator_dqbuf),
	  through a different thread, before the memory is referenced (gst_memory_ref).
	  In this case, in gst_v4l2_allocator_dqbuf(), the memory is unreferenced
	  (gst_memory_unref) before having been referenced: the memory refcount
	  reaches 0, and the memory is freed.
	  So, to avoid this crossing case, in gst_v4l2_allocator_qbuf(), the
	  memory shall be referenced before the buffer is queued.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777399

2017-01-24 17:59:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: Only write 4 byte zero padding to the Video Sample Description in MOV
	  For MP4 this is not defined, and it actually breaks things for MSE in
	  Chrome if we do this. For MOV this is required by some broken software
	  but the official specification says it's optional:
	  https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFChap3/qtff3.html
	  https://bugzilla.gnome.org/show_bug.cgi?id=777540

2017-01-02 13:42:04 +0100  Santiago Carot-Nemesio <scarot@twilio.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpstats: Keep number of nacks sent/received per source
	  Currently, the nack packets sent or received are kept at session level,
	  which makes it impossible to distinguish how many of these packages were
	  sent/received per ssrc when several sources are in the same session. This
	  patch is aligned with the https://www.w3.org/TR/webrtc-stats/#dom-rtcrtpstreamstats
	  https://bugzilla.gnome.org/show_bug.cgi?id=776714

2016-12-08 15:59:33 +0100  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Fix handling of config-interval
	  Insert VPS/SPS/PPS before the first NAL unit containing an I-frame in an
	  access unit only. If an access unit consists of several such NAL units
	  (tiles) VPS/SPS/PPS should only be inserted before the first of them so
	  that parameters are only updated between frames.
	  Do not insert VPS/SPS/PPS before P-frames when config-interval is -1.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775817

2017-01-19 12:29:44 +0100  Arnaud Vrac <avrac@freebox.fr>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: report a useful error message when soup_session_send fails
	  This helps to understand cases where libsoup doesn't set the message
	  status code after running soup_session_send.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777222

2017-01-19 11:05:00 +0100  Arnaud Vrac <avrac@freebox.fr>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: properly check that seek range was respected
	  This check must be done only when we are sure the request was
	  successfully sent. soup_session_send() might fail without setting the
	  status code. In this case status code is 0 so we would only catch the
	  error after the seek range check. In this case we would report an error
	  saying that the seek range was not respected, instead of reporting the
	  underlying error that triggered the soup_session_send() failure.
	  https://bugzilla.gnome.org/attachment.cgi?bugid=777222

2017-01-09 21:04:51 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	  gdkpixbufoverlay: add a positioning coefficient pair
	  ... so as to allow one clearly defined (absolute) positioning mode
	  that can cater for a variety of absolute but also relative positioning
	  with respect to edge or center.

2017-01-21 20:48:22 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: update composition in _before_transform
	  ... since we need to determine passthrough mode for buffer preparation before
	  calling into _transform_ip.

2017-01-07 20:11:13 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: handle setting NULL gdkpixbuf
	  ... which is a clearer way to clear any current overlay, other than
	  fiddling with alpha or positioning properties to make it virtually go away.

2017-01-20 17:16:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Stop reading a ncdt sub-tag if it goes behind the surrounding tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=777532

2017-01-20 07:58:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix various out of bounds reads when parsing ncdt tags
	  https://bugzilla.gnome.org/show_bug.cgi?id=777500

2017-01-19 13:46:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Increment current stts index whenever we finished one stts entry
	  Otherwise we could read more chunks than there are available, doing an
	  out of bounds read and potentially crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777469

2017-01-19 13:25:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  Revert "qtdemux: Increment current stts index in all code paths after reading one chunk"
	  This reverts commit 99d5d7570d0b53dad3bc8eb653b1320ee422aace. It broke
	  playback of various valid files.

2017-01-19 07:52:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Increment current stts index in all code paths after reading one chunk
	  Otherwise we could read more chunks than there are available, doing an
	  out of bounds read and potentially crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777469

2017-01-19 08:37:37 +0100  Edward Hervey <edward@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Initialize return variable
	  In the normal use-case we would end up with ret being unitialized
	  causing havoc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777222

2017-01-13 12:27:40 +0000  David Warman <dwarman@manglebit.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid XMP tag parsing fatal error.
	  qtdemux_handle_xmp_taglist() requires a writable taglist,
	  but qtdemux->tag_list can become non-writable, specifically
	  after sending global tags (qtdemux.c:958), which adds a
	  second reference.  Ensure the list is made writable before
	  calling (make_writable will copy the list if necessary).
	  https://bugzilla.gnome.org/show_bug.cgi?id=766177

2016-05-31 13:17:45 -0300  Thiago Santos <thiagossantos@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: rework taglist handling
	  Keep taglist around during element existance to avoid having to
	  create it at different places before usage. Makes code simpler to handle.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766177

2017-01-16 11:58:02 +0100  Arnaud Vrac <avrac@freebox.fr>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: make flow return values handling clearer
	  The flow return values was stored in the element before because the
	  result had to be set from callbacks. This is not the case anymore, we
	  can return the flow result directly from functions, making the code
	  easier to understand.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777222

2017-01-13 16:40:43 +0100  Arnaud Vrac <avrac@freebox.fr>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: properly track redirections
	  The current code configures libsoup to handle redirections
	  transparently, without informing the caller, thus preventing the element
	  to record the redirect code and location uri.
	  Fix this by always setting the SOUP_MESSAGE_NO_REDIRECT, preventing
	  libsoup from handling the redirection. When we receive a redirection
	  request and libsoup can safely handle it, return a custom error which
	  triggers a retry with the new URI.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777222

2017-01-17 10:53:39 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: add 4444 and 4444xq variants to video/x-prores pad template caps
	  They are handled since commit 7b565475bf551c53b8eed46f7086f3b372f1f6c4
	  (qt: Add support for ProRes 4444 XQ).
	  https://bugzilla.gnome.org/show_bug.cgi?id=777377

2017-01-17 10:48:57 +1100  Jan Schmidt <jan@centricular.com>

	* gst/matroska/ebml-read.c:
	  matroska: Quiet a WARN when parsing push mode
	  This warning was noisy when returning EOS, which is
	  just used to indicate more data is needed from upstream.

2017-01-16 14:50:22 +0100  Georg Lippitsch <glippitsch@toolsonair.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't write Sync Sample Atom for ProRes
	  https://bugzilla.gnome.org/show_bug.cgi?id=777331

2015-01-28 08:58:26 +0100  Enrico Jorns <ejo@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2_calls.h:
	  v4l2: Remove usage and definition of LOG_CAPS macro
	  Unlike former definitions of LOG_CAPS, the current implementation simply
	  expands to GST_DEBUG_OBJECT. The LOG_CAPS macro is rarely used and most
	  uses duplicate already existing GST_DEBUG_OBJECT lines. Therefore, the
	  caps are often printed twice which unnecessarily clutters the debug log.
	  Replace LOG_CAPS calls with GST_DEBUG_OBJECT, remove LOG_CAPS calls, and
	  delete the definition of LOG_CAPS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=776899

2017-01-16 15:40:43 +0100  Jean-Christophe Trotin <jean-christophe.trotin@st.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: remove duplicated line of code
	  https://bugzilla.gnome.org/show_bug.cgi?id=777330

2017-01-16 15:17:15 +0100  Jean-Christophe Trotin <jean-christophe.trotin@st.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: fix memory type in allocator probe
	  The buffer memory type provided to the VIDIOC_CREATE_BUFS ioctl shall
	  be set with the value ("memory") given as input parameter of the
	  gst_v4l2_allocator_probe() function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777327

2017-01-14 15:27:19 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/flac/gstflacenc.c:
	  flacenc: fix other icon counter check
	  It's never going to be 0 if we first increment and then check.

2017-01-14 15:16:53 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: boldly assume that first 'covr' image is the front cover

2017-01-14 15:09:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: extract cover art images into GST_TAG_IMAGE not PREVIEW_IMAGE
	  These are usually much bigger than icon size and required by
	  iTunes to be certain fairly large sizes. In qtmux it is also
	  the IMAGE tags which we write out as 'covr' atoms.

2017-01-14 15:05:36 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/flac/gstflacenc.c:
	  flacenc: also set PICTURE tag width and height if available

2017-01-14 14:58:52 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/flac/gstflacenc.c:
	  flacenc: fix encoder init error with some GST_TAG_PREVIEW_IMAGEs
	  The encoder fails to initialise when we try to set GST_TAG_PREVIEW_IMAGEs
	  sent to use by qtdemux from iTunes-generated m4a files. We should
	  not just blindly translate the PREVIEW tag to file icon image types,
	  but check if the specific conditions required are met (i.e. image
	  type 1 must be a 32x32 PNG icon, and what we're getting is 500x500).
	  https://bugzilla.gnome.org/show_bug.cgi?id=776962

2017-01-13 12:39:00 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: bump version

2017-01-11 10:32:23 -0300  Juan Pablo Ugarte <ugarte@endlessm.com>

	* tests/examples/gtk/glliveshader.c:
	  gl/examples/gtk: fixed compilation on systems without GL_GEOMETRY_SHADER
	  https://bugzilla.gnome.org/show_bug.cgi?id=777143

2017-01-12 21:35:25 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtsink.cc:
	* ext/qt/gstqtsrc.cc:
	  gl/utils: also take care of the local GL context in query functions
	  Simplifies a deduplicates a lot of code in elements retrieving/setting
	  the local OpenGL context.

2017-01-12 21:35:25 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	  gl/utils: also take care of the local GL context in query functions
	  Simplifies a deduplicates a lot of code in elements retrieving/setting
	  the local OpenGL context.

2016-12-22 17:40:40 +0200  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Add option for timecode-based split
	  If this option is given, it will calculate the next split point based on
	  timecode difference.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774209

2017-01-13 00:01:06 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't reset request pad numbering across uses
	  When reset, don't restart request pad numberings, as
	  request pads can survive across state changes. Only
	  restart at 0 if all request pads are handed back first.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777174

2017-01-11 18:52:28 +0100  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtxqueue: Expose basic statistics as properties.
	  Statistics about the total number of retransmission requests
	  and the actual number of retransmitted packets can be helpful
	  at application-level.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777182

2017-01-12 17:45:35 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: simplify video/x-h264 caps handling
	  'stream-format' and 'alignment' are defined in pad template caps so
	  there is no need to check them again here. Also remove bitrate parsing from
	  caps as bitrate in caps doesn't make sense but from tags, which is
	  actually the case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777181

2016-12-08 17:02:22 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: add basic HEVC/H.265 muxing support
	  https://bugzilla.gnome.org/show_bug.cgi?id=736752

2017-01-11 18:29:05 +0100  Georg Lippitsch <glippitsch@toolsonair.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Calculate clean aperture size
	  Calculate clean aperture dimensions by first guessing
	  display aspect ratio based on pixel aspect ratio and
	  frame size.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777100

2017-01-10 18:19:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtmux: Write tapt atom for MOV files if PAR not 1/1
	  Needed for QuickTime 7 to properly play files.
	  Also write the clap atom for MOV files always, not only when ProRes is
	  used as a video codec. It's mandatory for MOV.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777100

2017-01-12 16:32:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.11.1 ===

2017-01-12 15:31:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	  Release 1.11.1

2017-01-12 14:38:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2017-01-12 14:36:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	* po/hr.po:
	* po/id.po:
	* po/zh_CN.po:
	  po: Update translations

2017-01-11 17:53:32 -0800  Andre McCurdy <armccurdy@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: free seqh after calling qtdemux_parse_svq3_stsd_data()
	  The seqh buffer allocated in qtdemux_parse_svq3_stsd_data() needs to
	  be freed by the caller after use.
	  https://bugzilla.gnome.org/show_bug.cgi?id=777157
	  Signed-off-by: Andre McCurdy <armccurdy@gmail.com>

2017-01-10 16:01:35 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  isomp4: Don't spam debug log with knonw/padding atoms
	  Only output WARNING messages for atoms we don't know how to handle
	  instead of for padding/known atoms we don't need to do any processing
	  on
	  https://bugzilla.gnome.org/show_bug.cgi?id=777095

2017-01-10 16:54:48 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/qt/qtwindow.cc:
	* ext/qt/qtwindow.h:
	  qmlglsrc: use glBlitFramebuffer to copy texture for GLES3.0
	  If support glBlitFrameBuffer, use it for texture copy instead
	  of glCopyTexImage2D
	  https://bugzilla.gnome.org/show_bug.cgi?id=777078

2017-01-09 19:05:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtsp/gstrtspsrc.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	  Fix indentation

2017-01-09 19:04:04 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: rtpjitterbuffer: fix compiler warning due to c99-ism
	  rtpjitterbuffer.c:592:3: error: ‘for’ loop initial declarations are only allowed in C99 mode

2016-11-11 14:31:03 +1100  Matthew Waters <matthew@centricular.com>

	* gst/autodetect/gstautodetect.c:
	  autodetect: bring the element state down after success
	  Otherwise some messages that are emitted by the element on NULL->READY
	  will not reach the application.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764947

2017-01-08 01:13:32 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Write tfdt atom into fragmented files.
	  The DASH spec requires that tfdt atoms be present, so
	  write one out. ISO/IEC 23009-1:2014 6.3.4.2
	  https://bugzilla.gnome.org/show_bug.cgi?id=708221

2017-01-07 23:55:42 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't reset output timestamps when no tfdt
	  If a fragmented stream doesn't have a tfdt, don't
	  reset the output timestamps at each fragment boundary
	  by erroneously using the default value of 0. Introduced
	  by commit 69fc48
	  https://bugzilla.gnome.org/show_bug.cgi?id=754230

2016-12-16 16:51:48 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* ext/vpx/meson.build:
	* gst/equalizer/meson.build:
	* gst/isomp4/meson.build:
	* meson.build:
	  meson: Install presets files

2017-01-03 10:12:30 +0530  Garima Gaur <garima.g@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix some caps leaks
	  https://bugzilla.gnome.org//show_bug.cgi?id=776789

2016-12-22 17:34:08 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Return a bin with a "location" property as a sink
	  Splitmuxsink might be called with a custom bin as a sink. If it has a
	  "location" property, it can be used.

2016-11-18 22:42:18 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmux: Rewrite buffer collection and scheduling
	  Majorly change the way that splitmuxsink collects
	  incoming data and sends it to the output, so that it
	  makes all decisions about when / where to split files
	  on the input side.
	  Use separate queues for each stream, so they can be
	  grown individually and kept as small as possible.
	  This removes raciness I observed where sometimes
	  some data would end up put in a different output file
	  over multiple runs with the same input.
	  Also fixes hangs with input queues getting full
	  and causing muxing to stall out.

2016-11-17 23:40:27 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* tests/check/elements/splitmux.c:
	  splitmuxsink: Add format-location-full signal
	  Add a new signal for formatting the filename, which receives
	  a GstSample containing the first buffer from the reference
	  stream that will be muxed into that file.
	  Useful for creating filenames that are based on the
	  running time or other attributes of the buffer.
	  To make it work, opening of files and setting filenames is
	  now deferred until there is some data to write to it,
	  which also requires some changes to how async state changes
	  and gap events are handled.

2016-12-31 01:54:01 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Always snap to the start of the keyframe
	  When performing a key-unit seek, always snap to the start ts
	  of the keyframe buffer we landed on so that the keyframe is
	  entirely within the resulting outgoing segment. That seems
	  the most sensible result, since the user requested snapping
	  to the keyframe position.

2016-12-31 01:48:04 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Omit cslg_shift when snapping seeks
	  Segments times and seek requests are stored and handled
	  in raw 'PTS' time, without the cslg_shift - which only applies
	  to outgoing samples. Omit the cslg_shift portion when
	  extracting PTS to compare for internal seek snaps.
	  If the cslg_shift is included, then keyframe+snap-before seeks
	  generate a segment start/stop time that already includes the
	  cslg_shift, and it's then added a 2nd time, causing the
	  first buffer(s) to have timestamps that are out of segment.

2016-12-30 22:31:38 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: Remove bogus check in atom_stsc_add_new_entry()
	  Remove an old check from atom_stsc_add_new_entry() that
	  extends the last entry in the STSC if the samples per chunk
	  matches, as the new interleave merging logic requires that
	  the final entry by updateable. There's already code
	  below which simply merges the final entry into the previous
	  one when needed, so rely on that instead.
	  Fixes asserts like:
	  ERROR:atoms.c:2940:atom_stsc_update_entry: assertion failed:
	  (atom_array_index (&stsc->entries, len - 1).first_chunk == first_chunk)

2016-04-24 21:38:51 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix key_time in gst_qtdemux_adjust_seek()
	  time in segment should be PTS based (not DTS).
	  https://bugzilla.gnome.org/show_bug.cgi?id=765498

2016-12-28 22:49:27 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Pass seek flags when activating.
	  Pass all seek flags when activating a part
	  based on a seek, so that SNAP flags are preserved.

2016-11-26 01:13:19 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmux: Fix a small race in the splitmuxsrc
	  Make sure the state of the parser is set to
	  collecting streams before chaining up to the
	  parent change_state() method, to close a
	  small window that can cause playback to
	  never commence.

2017-01-02 15:06:33 +0100  Edward Hervey <edward@centricular.com>

	* tests/check/elements/amrparse.c:
	  check: Remove dead code

2016-12-31 09:52:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: refactor max_files handling a bit
	  Use GQueue instead of a GSList so we don't have to traverse
	  the whole list to append something every time. And it also
	  keeps track of the number of items in it for us.
	  Add a function to add filenames to the list of old files and
	  use it in more places, so that memory doesn't build up in
	  other modes either if no max_files limit is specified.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766991

2016-05-29 17:21:47 +0100  Ursula Maplehurst <ursula@kangatronix.co.uk>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: don't leak memory when no max-files limit is set
	  Technically we weren't leaking the memory, just storing it internally
	  and never using it until the element is freed. But we'd still use more
	  and more memory over time, so this is not good over longer periods
	  of time. Only keep track of files if there's actually a limit set,
	  so that we will prune the list from time to time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766991

2016-12-29 12:39:20 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: adjust segment stop for KEY_UNIT negative rate seeking

2016-12-29 12:25:35 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: implement pull mode SNAP flag seeking

2016-12-29 11:26:33 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavidemux.c:
	  avidemux: tweak KEY_UNIT SNAP seek handling
	  Previously, seeking to position y where y is (strictly) within a keyframe
	  would seek to that keyframe both with SNAP_BEFORE and SNAP_AFTER,
	  where the latter is now adjusted to really snap to the next keyframe.

2016-12-28 13:23:11 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavidemux.c:
	  avidemux: correctly perform pull mode KEY_UNIT seeking
	  Rather amazingly (and equally unnoticed), keyunit seeking resulted in segments
	  where start != time (which is bogus for simple avi timeline).  So, properly
	  adjust the segment (start) rather than fiddling with segment time (only).

2016-12-28 13:04:54 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavidemux.c:
	  avidemux: restore considering of pull mode KEY_UNIT seeking
	  ... by using the original seek event's flags rather than the corresponding
	  segment flags, which do not have such counterpart flags (and
	  do no longer have them covertly sneaking in nowadays).

2015-05-08 12:44:01 +0200  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: only drop actual streamheader buffers with xiph codecs
	  With Xiph codecs the stream header buffers are both in the caps and are
	  usually also at the beginning of each input stream, but it's perfectly
	  possible that the input stream does not have the stream header buffers
	  inline in the data. Matroskamux would drop the first N buffers assuming
	  they're stream headers, but this meant it would drop actual payload data
	  when the stream didn't contain the stream headers inline. Fix this by
	  only dropping leading buffers if they're flagged as stream headers. This
	  fixes issues with streams that are being tapped into after streaming
	  has started.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749098

2016-12-21 17:43:58 +0100  Nicola Murino <nicola.murino@gmail.com>

	* tests/check/elements/matroskamux.c:
	  matroskamux: adjust unit test to modified behaviour
	  Now matroskamux mark all packets of audio-only streams as keyframes so
	  in test_block_group after pushing the test audio data 4 buffers are produced
	  and not more 2. The last buffer is the original data and must match with what
	  pushed. The remaining ones are matroskamux headers
	  https://bugzilla.gnome.org/show_bug.cgi?id=754696

2016-05-30 01:15:31 +0200  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: mark all packets of audio-only streams as keyframes
	  This helps with streaming audio-only streams via multifdsink,
	  tcpserversink and such.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754696

2015-03-28 18:15:36 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: add G722 audio support
	  https://bugzilla.gnome.org/show_bug.cgi?id=746574

2016-12-13 11:11:07 +0900  Wonchul Lee <wonchul.lee@collabora.com>

	* gst/udp/gstudpsrc.c:
	  updsrc: Add to join multiple multicast interfaces
	  https://bugzilla.gnome.org/show_bug.cgi?id=776030

2015-03-25 13:51:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpklvdepay.c:
	  rtpklvdepay: add the SPARSE flag to the outgoing stream-start event

2016-12-17 13:42:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/qt/gstqtsink.cc:
	* ext/qt/gstqtsrc.cc:
	  qt: improve element and property descriptions a bit

2016-12-14 14:37:45 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpmanager: place content before Since-version API marker
	  Avoids confusing the parser

2016-12-14 14:16:53 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/shout2/gstshout2.c:
	  shout2: fix 404 in package origin

2016-12-14 21:45:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check if we have enough data available when parsing edit lists
	  Also consume the data entry by entry to get complicated indexing out of
	  the code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=776107

2016-12-14 19:15:03 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't check size in a non-list value
	  After commit 1ea9735a I see these error while using the webcam
	  integrated in my laptop:
	  GStreamer-CRITICAL **: gst_value_list_get_size: assertion 'GST_VALUE_HOLDS_LIST (value)' failed
	  The issue is gst_v4l2src_value_simplify() was doing its job of
	  generating a single value, rather than the original list. That why,
	  when getting the list size, a critical warning was raised.
	  This patch takes advantage of the compiler optimizations to verify
	  first if the list was simplified, thus use it directly, otherwise,
	  if it is a list, verify its size.
	  https://bugzilla.gnome.org/show_bug.cgi?id=776106

2016-12-14 10:39:12 +0100  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests/jitterbuffer: Major refactoring and cleanups
	  * Changed PCMU->TEST for common macros
	  * Changed verify-functions (lost & rtx) into macros.
	  * Remove option to add marker-bit for test-buffers (not used anywhere)
	  * Add new push_test_buffer function that makes sure there are correlation
	  between dts and the time on the clock. (classic test-mistake)
	  * Established a generic starting-point for tests with the
	  construct_deterministic_initial_state function and use it where
	  applicable, which removes lots of "boilerplate" everywhere.
	  * Add basic lost-event test
	  * Remove as much "magic constants" as possible.
	  * Remove 3 tests that no longer are testing anything that others don't,
	  and was completely unmaintainable.
	  * Remove unnecessary use of the testclock
	  * Verify each test is testing what it actually says it does (and modify
	  where it doesn't)
	  In general, make the tests much smaller, better, more maintainable and
	  readable.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774409

2016-12-14 09:54:11 +0000  Tim-Philipp Müller <tim@centricular.com>

	* .gitignore:
	* Makefile.am:
	* configure.ac:
	* gst-plugins-good.spec.in:
	  Remove generated .spec file
	  Likely extremely bitrotten, and we should not ship this anyway.

2016-12-14 10:15:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check that the XiTh size is big enough
	  https://bugzilla.gnome.org/show_bug.cgi?id=775794

2016-12-09 20:27:53 +0900  Heekyoung Seo <heekyoung.seo@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check node length of video sample description
	  Add check for node length of video sample description and its fields and
	  for the XiTh atom.
	  Also unify the code a bit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775794

2016-12-08 18:50:52 +0900  Heekyoung Seo <heekyoung.seo@lge.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Enable xvid/mp2 codec support
	  Add support for xvid video and mp2 audio, add m2v1 fourcc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775794

2016-12-13 22:32:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvp9depay.c:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/rtprtx.c:
	* tests/check/elements/vp9enc.c:
	  gst: Don't declare variables inside the for loop header
	  This is a C99 feature.

2016-12-11 13:27:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Ensure to reinit buffers whenever they were not allocated yet
	  That is, whenever we go through start/stop we have to ensure that on the
	  next opportunity the buffers are reallocated again. Otherwise the
	  buffers might be NULL because the element was reused with the same
	  configuration as before (i.e. set_caps() wouldn't have reinited the
	  buffers).
	  https://bugzilla.gnome.org/show_bug.cgi?id=775898

2016-12-10 12:52:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/design/Makefile.am:
	* docs/design/design-rtpauxiliary.txt:
	* docs/design/design-rtpcollision.txt:
	* docs/design/design-rtpretransmission.txt:
	  docs: design: remove, moved to gst-docs

2016-12-09 17:17:35 -0300  Thibault Saunier <tsaunier@gnome.org>

	* meson.build:
	  meson: Support building without Gst debug

2016-12-09 17:55:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flx/gstflxdec.c:
	* gst/flx/gstflxdec.h:
	  flxdec: Only send SEGMENT events after CAPS
	  I.e., don't just forward the event but delay it if we don't have caps on
	  the srcpad yet.

2016-12-09 17:49:40 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flx/gstflxdec.c:
	  flxdec: Unref and unmap buffers in all code paths as needed
	  https://bugzilla.gnome.org/show_bug.cgi?id=775888

2016-12-08 12:37:25 +0300  Sergey Borovkov <sergey.borovkov@wireload.net>

	* ext/qt/gstqtglutility.cc:
	  qml: Fix egl being deinitialized on display cleanup
	  Use the with_egl_display() variant in order to not destroy the
	  EGLDisplay on destruction.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775793

2016-12-06 17:42:31 +0530  Arun Raghavan <arun@osg.samsung.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't set empty interlace-mode list
	  If for some reason we fail to probe formats (all try_fmt calls fail, for
	  example), this is not a critical error, but we end up with an empty list
	  of interlace modes. This causes all subsequent negotiation to fail.
	  This patch fixes interlace-mode setting to be skipped if we failed to
	  detect any.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775702

2016-12-07 17:22:22 +0530  Garima Gaur <garima.g@samsung.com>

	* gst/monoscope/gstmonoscope.c:
	  monoscope: Unref allocation query after finished with it
	  https://bugzilla.gnome.org/show_bug.cgi?id=775752

2016-12-07 22:55:46 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  qml/item: also unref the display on destruction
	  Leaking objects (and a thread!) is never a good idea.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775746

2016-12-07 22:58:29 +1100  Matthew Waters <matthew@centricular.com>

	* tests/examples/qt/qmlsink/main.cpp:
	  tests/examples/qmlsink: scope QApplication/Engine
	  So they are destroyed before gst_deinit() is run and the leaks tracer
	  doesn't show false-positives.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775746

2016-12-06 07:48:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flx/gstflxdec.c:
	  flxdec: Allocate 0-initialized memory for the decoded frame
	  Otherwise we might leak arbitrary information from the uninitialized
	  memory if not every pixel is written.
	  https://scarybeastsecurity.blogspot.gr/2016/12/1days-0days-pocs-more-gstreamer-flic.html

2016-12-05 07:57:19 -0700  Matt Staples <staples255@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix session cleanup when handling redirect on PLAY
	  Redirect on PLAY wasn't doing the necessary session cleanup. Fixed by
	  removing code from gst_rtspsrc_send that changed the state varable upon
	  encountering a redirect. Better to let the redirect handlers in
	  gst_rtspsrc_retrieve_sdp and gst_rtspsrc_play do their own
	  state-dependent cleanup.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775543

2016-09-07 16:10:27 +0300  Aleix Conchillo Flaque <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: always send teardown request
	  Allow CMD_CLOSE to cancel all commands not only CMD_PAUSE
	  and ignore CMD_WAIT while closing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748360

2016-12-03 08:19:27 +0100  Edward Hervey <bilboed@bilboed.com>

	* README:
	* common:
	  Automatic update of common submodule
	  From f980fd9 to 39ac2f5

2016-12-01 17:08:09 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: Don't leak duplicate items
	  When providing items with a seqnum, there is a (very small) probability
	  that an element with the same seqnum already exists. Don't forget
	  to free that item if it wasn't inserted.
	  And avoid returning undefined values when dealing with duplicate items

2016-12-01 11:23:02 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Sanitize unknown codec caps
	  We might have non-printable characters in the unknown fourcc, replace
	  them with '_', in the same way we do it for unknown tags.

2016-12-01 20:04:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Free vprp chunk also if it existed but we made no use of it
	  https://bugzilla.gnome.org/show_bug.cgi?id=775479

2016-12-01 17:38:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: Fix memory leak when parsing attachments
	  gst_tag_image_data_to_image_sample() does not take ownership of the
	  passed memory, so don't set it to NULL to allow us to free it later.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775472

2016-12-01 14:56:18 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: Unify zlib/bzip2 decompress loops with the ones from qtdemux
	  Especially, simplify the code a bit.

2016-12-01 14:41:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Increase inflate buffer in bigger steps
	  1024 bytes is quite small, let's do 4096 bytes (or one page).
	  Also remove redundant if, we're always in that case when getting here.

2016-12-01 14:30:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Ensure that size of the pasp atom is as much as we need
	  https://bugzilla.gnome.org/show_bug.cgi?id=775455

2016-12-01 14:30:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Free compressed moov node and it's corresponding decompressed data
	  https://bugzilla.gnome.org/show_bug.cgi?id=775455

2016-12-01 14:29:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check size of compressed MOOV header against available data
	  And actually read the size of the cmvd atom from the right position.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775455

2016-12-01 14:27:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix zlib inflate loop
	  Handle errors cleanly, deallocate all memory and return the actual size
	  of the inflated data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775455

2016-12-01 13:38:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Make sure we have enough data in the codec_data to be able to parse it
	  Also error out cleanly if mapping the buffer failed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775450

2016-12-01 13:32:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix out of bounds read in tag parsing code
	  We can't simply assume that the length of the tag value as given
	  inside the stream is correct but should also check against the amount of
	  data we have actually available.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775451

2016-12-01 15:06:06 +0530  Garima Gaur <garima.g@samsung.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpsbcdepay.c:
	  rtp: Fix some memory leaks in usage of gst_pad_get_current_caps()
	  https://bugzilla.gnome.org/show_bug.cgi?id=775071

2016-11-30 17:56:02 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Read interlacing information from 'fiel' atom
	  Read interlacing and TFF/BFF information from the 'fiel' atom and pass it
	  into the caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=775414

2016-11-29 13:55:40 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix compiler warning
	  qtdemux.c: In function ‘qtdemux_parse_trak’:
	  qtdemux.c:10184:38: error: format ‘%lu’ expects argument of type ‘long unsigned int’, but argument 9 has type ‘gint {aka const int}’ [-Werror=format=]
	  GST_DEBUG_OBJECT (qtdemux, "Found jpeg: len %u, need %lu", len,
	  ^

2016-11-28 13:45:24 -0800  Scott D Phillips <scott.d.phillips@intel.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Change off_t type to gint
	  off_t is a signed integer type provided by sys/types.h on posix systems.
	  Replace with gint for building on non-posix systems (like windows).
	  https://bugzilla.gnome.org/show_bug.cgi?id=775287

2016-11-22 21:00:25 -0800  Scott D Phillips <scott.d.phillips@intel.com>

	* meson.build:
	  meson: add libm to has_function checks
	  The functions from math.h may be implemented in libm.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774876

2016-10-27 23:02:37 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/meson.build:
	  Revert "meson: dv plugin now works on MSVC"
	  This reverts commit 05a89613feff70cff416367f5aa807a1d5c68b63.
	  Let's not put in stuff that needs unreleased Meson. This can go in
	  for the next cycle.

2016-11-28 13:51:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Ensure that tags are valid UTF-8 before adding them to the taglist
	  https://bugzilla.gnome.org/show_bug.cgi?id=775219

2016-11-28 12:22:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: Post an error message on the bus if we got EOS without having added any pads

2016-11-28 12:00:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Handle non-UTF8 headers and error reasons more gracefully
	  Especially don't put them into GstStructures in one way or another, just
	  ignore them or error out cleanly depending on the importance of their
	  content.

2016-11-28 09:30:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvrawpay.c:
	  vrawpay: Error out cleanly if mapping the video frame fails
	  Instead of later dereferencing NULL and crashing.

2016-11-27 11:14:13 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: Update statistics before pushing
	  If an element queries the number of retransmission buffers pushed
	  *while* the push is still taking place (and before the object lock
	  is taken just after) it would end up with the wrong statistic
	  being reported.
	  Increment it just before the push, avoids races when getting statistics
	  https://bugzilla.gnome.org/show_bug.cgi?id=768723

2016-11-26 11:20:51 +0000  Tim-Philipp Müller <tim@centricular.com>

	* .gitmodules:
	  common: use https protocol for common submodule
	  https://bugzilla.gnome.org/show_bug.cgi?id=775110

2016-07-28 18:51:24 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  gstv4l2bufferpool: lock flush_stop against regular qbuf
	  These can be called from different threads and both manipulate the
	  pool->buffers array. Lock them properly and let flush_stop move the
	  array contents into a temporary array on the stack to avoid having
	  to call release_buffer under the object lock.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775015

2016-11-24 14:25:22 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  gstv4l2bufferpool: remove critical error message when process is called on an inactive pool
	  If the pool is inactive, it is guaranteed to also be flushing, so the
	  following check will return GST_FLOW_FLUSHING anyway.
	  This can happen if a v4l2src is blocking on DQBUF in create and is sent
	  an EOS event on another thread. In that case the pool is set to
	  flushing/inactive without locking, the v4l2src is unblocked, and may
	  call pool_process with a valid buffer on the already inactive pool.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775014

2016-11-24 14:41:52 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: release buffer if create fails
	  gst_base_src_get_range does not expect a buffer to be returned in
	  the error case, so we are leaking a reference here if create fails.
	  https://bugzilla.gnome.org/show_bug.cgi?id=775014

2016-11-23 18:34:04 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Handle create_session() returning NULL in bundle code
	  CID 1394492.

2016-11-22 16:42:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Make sure to only change DTS of writable buffers
	  And trivial cleanup
	  https://bugzilla.gnome.org/show_bug.cgi?id=774840

2016-11-22 16:42:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Error out much earlier if we don't have a valid PTS
	  https://bugzilla.gnome.org/show_bug.cgi?id=774840

2016-11-22 16:18:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Only use buffer durations if they are actually valid
	  https://bugzilla.gnome.org/show_bug.cgi?id=774840

2016-11-22 15:59:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Revert commits that set DTS and duration on buffers unconditionally
	  39f7e52266fde3b3c035e22cbcbb2bb1fa207b17 was setting the buffer duration
	  to 0 if is not valid, under the assumption that this is "the last"
	  buffer and no others are coming next. This is wrong, last_buf is the
	  previous buffer and not the very last one.
	  4e3c13c87c258c9c95e2217d32ab314d12b5fffc was setting DTS to 0 if there
	  was none. This will set DTS to 0 for all e.g. audio streams, completely
	  messing up calculations if streams don't start at 0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774840

2016-11-22 15:58:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Only write "gap" edit list if there is a non-zero gap
	  https://bugzilla.gnome.org/show_bug.cgi?id=774840

2016-11-23 07:09:06 +1100  Matthew Waters <matthew@centricular.com>

	* gst/flx/flx_color.c:
	* gst/flx/flx_fmt.h:
	* gst/flx/gstflxdec.c:
	* gst/flx/gstflxdec.h:
	  flxdec: rewrite logic based on GstByteReader/Writer
	  Solves overreading/writing the given arrays and will error out if the
	  streams asks to do that.
	  Also does more error checking that the stream is valid and won't
	  overrun any allocated arrays.  Also mitigate integer overflow errors
	  calculating allocation sizes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774859

2016-11-23 11:20:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flx/gstflxdec.c:
	  flxdec: Don't unref() parent in the chain function
	  We don't own the reference here, it is owned by the caller and given to
	  us for the scope of this function. Leftover mistake from 0.10 porting.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774897

2016-11-22 20:33:29 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: libvpx's release buffer is sometimes called with fb->priv==NULL
	  Don't assert on this but just ignore these cases.

2016-11-22 20:24:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix cluster searching if we search multiple times in one chunk
	  After finding a cluster id in the byte reader, we skip ahead the reader
	  position by one further byte to be able to continue searching from there
	  inside the same chunk if the cluster candidate was a false positive.
	  We have to accomodate for that additional byte when resuming the search,
	  otherwise all following pulls are off-by-one for every resume and we run
	  into an assertion.

2016-11-22 20:01:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-ids.c:
	  matroska: Add size checks to the parsing of FLAC headers

2016-11-22 23:46:00 +1100  Matthew Waters <matthew@centricular.com>

	* gst/flx/gstflxdec.c:
	  flxdec: fix some warnings comparing unsigned < 0
	  bf43f44fcfada5ec4a3ce60cb374340486fe9fac was comparing an unsigned
	  expression to be < 0 which was always false.
	  gstflxdec.c: In function ‘flx_decode_brun’:
	  gstflxdec.c:322:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
	  if ((glong) row - count < 0) {
	  ^
	  gstflxdec.c:332:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
	  if ((glong) row - count < 0) {
	  ^
	  https://bugzilla.gnome.org/show_bug.cgi?id=774834

2016-11-21 16:17:31 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Enable up to 16 unpositioned raw audio channels
	  https://bugzilla.gnome.org/show_bug.cgi?id=774789

2016-11-22 19:05:00 +1100  Matthew Waters <matthew@centricular.com>

	* gst/flx/gstflxdec.c:
	  flxdec: add some write bounds checking
	  Without checking the bounds of the frame we are writing into, we can
	  write off the end of the destination buffer.
	  https://scarybeastsecurity.blogspot.dk/2016/11/0day-exploit-advancing-exploitation.html
	  https://bugzilla.gnome.org/show_bug.cgi?id=774834

2016-11-21 15:25:23 +0000  David Evans <bbcrddave@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Be sure not to read off end of FLAC dfLa box
	  https://bugzilla.gnome.org/show_bug.cgi?id=773712

2016-11-21 11:48:58 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: add support for skipping invalid data in push mode
	  https://bugzilla.gnome.org/show_bug.cgi?id=774566

2016-11-21 11:48:29 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroskaparse: add support for skipping invalid data
	  https://bugzilla.gnome.org/show_bug.cgi?id=774566

2016-11-18 17:00:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Move to new helper function to parse authentication responses
	  https://bugzilla.gnome.org/show_bug.cgi?id=774416

2016-11-20 14:12:16 +0100  christophecvr <stefansat@telenet.be>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix wrong compiler warning with gcc 6.2
	  | ../../../git/gst/isomp4/qtdemux.c: In function 'qtdemux_parse_tree':
	  | ../../../git/gst/isomp4/qtdemux.c:10224:24: error: 'size' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  |                  offset += size;
	  |                         ^~
	  | ../../../git/gst/isomp4/qtdemux.c:10197:25: note: 'size' was declared here
	  |                  guint32 size, tag;
	  |                          ^~~~
	  https://bugzilla.gnome.org/show_bug.cgi?id=774747

2016-11-20 16:15:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* Makefile.am:
	* configure.ac:
	* win32/MANIFEST:
	* win32/common/config.h:
	  win32: remove copies of generated headers

2016-11-20 13:14:08 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Ensure that raw video have properly aligned buffers
	  That is, aligned to to 32 bytes for video. Fixes crashes if the raw
	  buffers are passed to SIMD processing functions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774428

2016-11-20 13:08:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Ensure that raw audio and video have properly aligned buffers
	  That is, aligned to the basic type for audio and to 32 bytes for video.
	  Fixes crashes if the raw buffers are passed to SIMD processing functions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774428

2016-11-14 14:44:11 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Always write edit lists for the tracks to give a more accurate duration
	  Always write an edit list for the whole track. In general this is not
	  necessary except for the case of having a gap or DTS adjustment but
	  it allows to give the whole track's duration in the usually more
	  accurate media timescale.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774403

2016-11-18 22:45:45 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Remove useless return variable
	  qtdemux_expose_streams() returns flow error immediately, if there is an error.
	  So, the variable for the flow return is not needed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774674

2016-11-17 13:59:48 +0000  David Evans <bbcrddave@gmail.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add support for FLAC encapsulated in ISOBMFF
	  As defined by
	  https://git.xiph.org/?p=flac.git;a=blob_plain;f=doc/isoflac.txt
	  https://bugzilla.gnome.org/show_bug.cgi?id=773712

2016-11-17 19:59:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Mark pad as needing reconfiguration again if it failed
	  And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774623

2016-11-17 19:59:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/monoscope/gstmonoscope.c:
	  monoscope: Mark pad as needing reconfiguration again if it failed
	  And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774623

2016-11-17 19:58:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Mark pad as needing reconfiguration again if reconfiguration failed
	  And consider negotiation failures on flushing pads as FLUSHING, not as
	  NOT_NEGOTIATED.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774623

2016-11-17 19:56:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdec.c:
	  dvdec: Fix handling of negotiation failures
	  Return NOT_NEGOTIATED if sending the caps event fails, or FLUSHING if
	  the pad was flushing at that point.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774623

2016-11-17 17:16:26 -0800  Scott D Phillips <scott.d.phillips@intel.com>

	* meson.build:
	  meson: add_global_arguments -> add_project_arguments
	  https://bugzilla.gnome.org/show_bug.cgi?id=774656

2016-11-16 10:53:51 +0530  Vinod Kesti <vinodkesti@yahoo.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: pad request fails for flvmux
	  splitmuxsink requests pad from element using pad template like "video_%u", "audio_%u" and "sink_%d". This is true for most of the muxers.
	  But splitmuxsink not able to request pad to flvmux as flvmux has "audio" and "video" as pad templates.
	  fix: splitmuxsink should fallback to "audio" and  "video" when template not found.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774507

2016-11-17 10:24:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: Add remaining relevant parts from a3a55305 to the parser
	  https://bugzilla.gnome.org/show_bug.cgi?id=774566

2016-11-16 22:39:01 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: ignore parsing errors at the end of the file
	  This is the same change as a3a55305 for the parser.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774566

2016-11-16 08:56:34 +0100  Philippe Normand <philn@igalia.com>

	* docs/plugins/gst-plugins-good-plugins.signals:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/rtpbundle.c:
	* tests/check/meson.build:
	* tests/examples/rtp/.gitignore:
	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-rtpbundle.c:
	* tests/examples/rtp/server-rtpbundle.c:
	  rtpbin: receive bundle support
	  A new signal named on-bundled-ssrc is provided and can be
	  used by the application to redirect a stream to a different
	  GstRtpSession or to keep the RTX stream grouped within the
	  GstRtpSession of the same media type.
	  https://bugzilla.gnome.org/show_bug.cgi?id=772740

2016-11-15 16:52:39 +0530  Vinod Kesti <vinodkesti@yahoo.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: assertion while converting ADTS stream to RAW
	  aacparse resizes input buffer while converting ADTS stream to RAW,
	  During buffer resize buffer write permission is not checked.
	  This throws gst_buffer_is_writable assertion and leads to AV sync issue some times.
	  It is corrected by making buffer writeable using gst_buffer_make_writable
	  https://bugzilla.gnome.org/show_bug.cgi?id=774129

2016-11-15 21:17:51 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't modify upstream TIME segment
	  TIME segment implies that stream/running time is being handled by upstream.
	  So, we shouldn't override it without any clue.
	  This patch is for fixing seek in DASH streaming.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774196

2016-11-14 22:33:27 +0530  Arun Raghavan <arun@osg.samsung.com>

	* config.h.meson:
	  meson: Add define for v4l2-probe config option

2016-11-14 17:37:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: Reset caps accumulator to ANY when resyncing the adapter, not EMPTY
	  The accumulator is filled by intersecting with all the pad caps, as such
	  it must be initialized with ANY (like it is before the iteration is
	  started) and not to EMPTY.
	  Fixes the CAPS query always returning EMPTY caps when resyncing happened
	  during the query, e.g. because pads were added/removed.

2016-11-14 12:13:14 +0100  Petr Kulhavy <brain@jikos.cz>

	* gst/udp/gstudpsrc.c:
	  udpsrc: remove redundant saddr unref
	  The g_object_unref (saddr) before receiving message seems to be redundant as it
	  is done just before jumping to retry
	  Though not directly related, part of
	  https://bugzilla.gnome.org/show_bug.cgi?id=772841

2016-11-12 23:34:23 +0100  Petr Kulhavy <brain@jikos.cz>

	* gst/udp/gstudpsrc.c:
	  udpsrc: receive control messages only in multicast
	  Control messages are used only in multicast mode - to detect if the destination
	  address is not ours and possibly drop the packet. However in non-multicast
	  modes the messages are still allocated and freed even if not used. Therefore
	  request control messages from g_socket_receive_message() only in multicast
	  mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=772841

2016-11-11 10:45:01 -0800  Scott D Phillips <scott.d.phillips@intel.com>

	* gst/matroska/matroska-mux.c:
	  Use intermediate guint when handling GstVideoMultiviewFlags
	  The underlying integer type of the enum GstVideoMultiviewFlags is
	  implementation defined and may not have the same size as guint.
	  https://bugzilla.gnome.org/show_bug.cgi?id=774293

2016-11-11 10:44:18 -0800  Scott D Phillips <scott.d.phillips@intel.com>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: update uri_get_type to match the prototype in GstURIHandlerInterface
	  https://bugzilla.gnome.org/show_bug.cgi?id=774293

2016-10-26 22:37:34 -0700  Scott D Phillips <scott.d.phillips@intel.com>

	* meson.build:
	  meson: don't add_global_arguments when being built as a subproject
	  https://bugzilla.gnome.org/show_bug.cgi?id=773568

2016-10-21 15:49:36 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: fix header rewriting being ignored
	  https://bugzilla.gnome.org/show_bug.cgi?id=727802

2016-11-09 06:25:27 +0000  Sean DuBois <sean@siobud.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Add metadatacreator property
	  Allow users to set metadatacreator value in the meta packet
	  https://bugzilla.gnome.org/show_bug.cgi?id=774131

2016-11-01 19:56:36 +0200  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Use first buffer TS as mux start time
	  Do not use last buffer TS + buffer duration because buffer duration
	  might be inaccurate, especially for frame rates like 30fps where a
	  rounding error is observed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773785

2016-11-07 14:47:22 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/qt/gstqtsrc.cc:
	* ext/qt/gstqtsrc.h:
	* ext/qt/qtwindow.cc:
	* ext/qt/qtwindow.h:
	  qmlglsrc: some enhancements for qmlglsrc
	  1. Need set use-default-fbo to qquickwindow during set property
	  to support change render target on the fly.
	  2. Calculate qmlglsrc refresh frame rate in qtglwindow
	  https://bugzilla.gnome.org/show_bug.cgi?id=774035

2016-11-03 15:03:59 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix timer-reuse bug
	  When doing rtx, the jitterbuffer will always add an rtx-timer for the next
	  sequence number.
	  In the case of the packet corresponding to that sequence number arriving,
	  that same timer will be reused, and simply moved on to wait for the
	  following sequence number etc.
	  Once an rtx-timer expires (after all retries), it will be rescheduled as
	  a lost-timer instead for the same sequence number.
	  Now, if this particular sequence-number now arrives (after the timer has
	  become a lost-timer), the reuse mechanism *should* now set a new
	  rtx-timer for the next sequence number, but the bug is that it does
	  not change the timer-type, and hence schedules a lost-timer for that
	  following sequence number, with the result that you will have a very
	  early lost-event for a packet that might still arrive, and you will
	  never be able to send any rtx for this packet.
	  Found by Erlend Graff - erlend@pexip.com
	  https://bugzilla.gnome.org/show_bug.cgi?id=773891

2016-10-09 15:59:05 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix lost-event using dts instead of pts
	  The lost-event was using a different time-domain (dts) than the outgoing
	  buffers (pts). Given certain network-conditions these two would become
	  sufficiently different and the lost-event contained timestamp/duration
	  that was really wrong. As an example GstAudioDecoder could produce
	  a stream that jumps back and forth in time after receiving a lost-event.
	  The previous behavior calculated the pts (based on the rtptime) inside the
	  rtp_jitter_buffer_insert function, but now this functionality has been
	  refactored into a new function rtp_jitter_buffer_calculate_pts that is
	  called much earlier in the _chain function to make pts available to
	  various calculations that wrongly used dts previously
	  (like the lost-event).
	  There are however two calculations where using dts is the right thing to
	  do: calculating the receive-jitter and the rtx-round-trip-time, where the
	  arrival time of the buffer from the network is the right metric
	  (and is what dts in fact is today).
	  The patch also adds two tests regarding B-frames or the
	  “rtptime-going-backwards”-scenario, as there were some concerns that this
	  patch might break this behavior (which the tests shows it does not).

2016-11-03 16:33:53 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix bug in reschedule_timer
	  The new timeout is always going to be (timeout + delay), however, the
	  old behavior compared the current timeout to just (timeout), basically
	  being (delay) off.
	  This would happen if rtx-delay == rtx-retry-timeout, with the result that
	  a second rtx attempt for any buffers would be scheduled immediately instead
	  of after rtx-delay ms.
	  Simply calculate (new_timeout = timeout + delay) and then use that instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773905

2016-11-03 13:27:51 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/wavparse.c:
	* tests/files/Makefile.am:
	* tests/files/audiotestsrc.wav:
	  tests: wavparse: add test for processing an actual .wav file
	  https://bugzilla.gnome.org/show_bug.cgi?id=773861

2016-11-03 12:34:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Don't set caps to NULL after setting them on the srcpad
	  We would like to check later on EOS if we found a known stream type or
	  not, to possibly post an error message.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773861

2016-10-05 12:19:12 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	  gl: GST_GL_TYPE -> GST_TYPE_GL
	  Some deprecated symbols are kept for backwards compatibility

2016-10-05 12:19:12 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtsink.cc:
	* ext/qt/gstqtsrc.cc:
	  gl: GST_GL_TYPE -> GST_TYPE_GL
	  Some deprecated symbols are kept for backwards compatibility

2016-11-02 14:33:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't deref NULL pads in debug output
	  That tends to crash.

2016-11-02 11:46:07 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: Don't use gst_video_colorimetry_to_string_full()
	  The API was reverted. Just use the plain
	  gst_video_colorimetry_to_string() function.

2016-11-02 11:00:13 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix GObject warnings on shutdown.
	  Commit 83e718 added a pad template to splitmux request
	  pads, which means that GstElement now releases the pads on
	  dispose, but after having removed all elements in the bin
	  and unlinked them. Make sure we can handle cleanup in that case
	  without throwing assertions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773784

2016-11-02 02:25:51 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multifile/gstsplitmuxsrc.h:
	  splitmuxsrc: Store seek seqnum and send it on EOS / segment events.
	  GES relies on the EOS event having the seqnum of the seek that
	  caused it.

2016-11-02 02:25:00 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Forward a not-linked error on the bus
	  Handle not-linked as for other fatal errors and post it
	  onto the bus so the app knows

2016-11-01 21:00:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix compiler warning
	  qtdemux.c: In function ‘qtdemux_parse_tree’:
	  qtdemux.c:10139:16: error: ‘color_table_id’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  if (color_table_id != 0) {
	  ^
	  qtdemux.c:10121:19: note: ‘color_table_id’ was declared here
	  guint16 color_table_id;
	  ^~~~~~~~~~~~~~

2016-10-20 17:40:59 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use a default interleave of 250ms for all codecs
	  https://bugzilla.gnome.org/show_bug.cgi?id=773217

2016-10-19 14:33:33 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use a default interleave when ProRes is used
	  The ProRes guidelines suggest an interleave of 0.5s is common, but
	  specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
	  be used per chunk.
	  It might also make sense to use similar numbers in general.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773217

2016-10-19 14:25:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Allow configuring the interleave size in bytes/time
	  Previously we were switching from one chunk to another on every single
	  buffer. This wastes some space in the headers and, depending on the
	  software, might depend in more reads (e.g. if the software is reading
	  multiple samples in one go if they're in the same chunk).
	  The ProRes guidelines suggest an interleave of 0.5s is common, but
	  specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
	  be used per chunk. This will be handled in a follow-up commit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773217

2016-09-30 18:22:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Set compressor name, horizontal/vertical resolution and depth for ProRes
	  This is also required by some software to handle ProRes files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769048

2016-09-30 18:05:38 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	  qt: Add support for ProRes 4444 XQ
	  And also 4444 in the muxer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769048

2016-09-30 17:58:37 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtmux: Write 'clap' atom for ProRes
	  It's required for ProRes to work with other software.
	  It is also in the MP4 standard, but inventing values here seems a bit
	  tricky for the general case and it does not really give any extra
	  information.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769048

2016-09-30 09:55:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Read colorimetry information from colr atom if available
	  https://bugzilla.gnome.org/show_bug.cgi?id=772181

2016-09-29 21:56:18 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Always write colr atom with the colorimetry information
	  https://bugzilla.gnome.org/show_bug.cgi?id=772181

2016-09-29 18:16:18 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix writing of the 'fiel' extension atom
	  This was also wrong for JPEG2000. Also write it for all MOV files and
	  JPEG2000, not only for ProRes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769048

2016-09-29 17:40:23 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: Write 4 bytes of zeroes at the end of the sample description extensions
	  This is working around some broken software.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769048

2016-09-28 20:55:24 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	  atoms: 'pasp' atom is also part of MP4, write it always
	  https://bugzilla.gnome.org/show_bug.cgi?id=769048

2016-07-11 19:30:12 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Write additional atoms for prores video
	  These required atoms are: colorimetry, field information, spatial/temporal
	  quality, and vendor.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769048

2014-06-16 17:20:32 +0200  Stian Selnes <stian.selnes@gmail.com>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: Don't drop mode b packets with picture start code
	  Some buggy payloaders, e.g. rtph263pay, may use mode B for packets
	  that starts with a picture (or GOB) start code although it's not
	  allowed. Let's be nice and not drop these packets/frames.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773516

2016-06-22 13:59:35 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtp/gstrtph263ppay.c:
	* tests/check/elements/rtph263.c:
	  rtph263ppay: Fix caps leak
	  Fix leaking caps when downstream has not-fixed caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773515

2016-10-26 16:42:19 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: Fix indentation
	  https://bugzilla.gnome.org/show_bug.cgi?id=773514

2016-10-18 11:35:58 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: Use GST_TRACE_OBJECT for logging bitstream parsing
	  Bump the bitstream parsing to TRACE log level so it doesn't flood the
	  output when trying to read the more useful DEBUG and LOG messages.
	  Also use GST_DEBUG_OBJECT instead of GST_DEBUG in various places
	  https://bugzilla.gnome.org/show_bug.cgi?id=773514

2016-10-18 11:09:10 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: Fix leak for B-fragments
	  Altough commits 6a16be7, 64f9d08 and 0c7e3a8 fixed some issues they
	  introduced others. This patch fixes the leak of one macroblock for every
	  B fragment.
	  Macroblock structures must not be freed immediately after finding the
	  boundaries as they are stored and used later. However the inital dummy
	  structure (used for finding the first boundary) must be freed.
	  CID #1212156
	  https://bugzilla.gnome.org/show_bug.cgi?id=773512

2016-10-20 13:14:13 +0200  Alejandro G. Castro <alex@igalia.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpbin: avoid generating errors when rtcp messages are empty and check the queue is not empty
	  Add a check to verify all the output buffers were empty for the
	  session in a timout and log an error.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773269

2016-10-26 13:21:29 +0200  Alejandro G. Castro <alex@igalia.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpbin: pipeline gets an EOS when any rtpsources byes
	  Instead of sending EOS when a source byes we have to wait for
	  all the sources to be gone, which means they already sent BYE and
	  were removed from the session. We now handle the EOS in the rtcp
	  loop checking the amount of sources in the session.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773218

2016-10-21 17:31:00 +0000  Matt Staples <staples255@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Also handle redirect on PLAY
	  https://bugzilla.gnome.org/show_bug.cgi?id=772610

2016-08-30 10:24:43 +0200  Petr Kulhavy <brain@jikos.cz>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: allow missing control attribute in case of a single stream
	  Improve RFC2326 - chapter C.3 compatibility:
	  In case just a single stream is specified in SDP and the control attribute
	  is missing do not drop the stream but rather assume "a=control:*"
	  https://bugzilla.gnome.org/show_bug.cgi?id=770568

2016-10-08 18:11:17 +0200  William Manley <will@williammanley.net>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2: Warn, don't assert if v4l gives us a buffer with a too large size
	  I've seen problems where the `bytesused` field of `v4l2_buffer` would be
	  a silly number causing the later call to:
	  gst_memory_resize (group->mem[i], 0, group->planes[i].bytesused);
	  to result in this error to be printed:
	  (pulsevideo:11): GStreamer-CRITICAL **: gst_memory_resize: assertion 'size + mem->offset + offset <= mem->maxsize' failed
	  besides causing who-knows what other problems.
	  We make the assumption that this buffer has still been dequeued correctly
	  so just clamp to a valid size so downstream elements won't end up in
	  undefined behaviour.
	  The invalid `v4l2_buffer` I saw from my capture device was:
	  buffer = {
	  index = 0,
	  type = 1,
	  bytesused = 534748928, // <- Invalid
	  flags = 8260, // V4L2_BUF_FLAG_TIMESTAMP_MONOTONIC | V4L2_BUF_FLAG_ERROR | V4L2_BUF_FLAG_DONE
	  field = 01330, // <- Invalid
	  timestamp = {
	  tv_sec = 0,
	  tv_usec = 0
	  },
	  timecode = {
	  type = 0,
	  flags = 0,
	  frames = 0 '\000',
	  seconds = 0 '\000',
	  minutes = 0 '\000',
	  hours = 0 '\000',
	  userbits = "\000\000\000"
	  },
	  sequence = 0,
	  memory = 2,
	  m = {
	  offset = 3537219584,
	  userptr = 140706665836544, // Could be nonsense, not sure
	  planes = 0x7ff8d2d5b000,
	  fd = -757747712
	  },
	  length = 2764800,
	  reserved2 = 0,
	  reserved = 0
	  }
	  This is from gdb with my own annotations added.
	  This was with gst-plugins-good 1.8.1, a Magewell XI100DUSB-HDMI video
	  capture device and kernel 3.13 using a dodgy HDMI cable which is great at
	  breaking HDMI capture devices.  I'm using io-mode=userptr and have built
	  gst-plugins-good without libv4l.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769765

2016-10-20 20:41:07 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use a better default value for the movie header timescale
	  Take the maximum video timescale, or if no video track is present the
	  previous value of 1800.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769041

2016-10-20 20:07:19 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Be more clever with the default video track timescale
	  Use the number of milliframes per second for integral and drop-frame
	  framerates, as suggested by the QT file format specification and other
	  places. We already did that for integral framerates before, but not for
	  drop-frame framerates. This now keeps precision better.
	  For all other framerates, check if it's close to a well-known framerate
	  and use that instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769041

2016-10-10 13:00:01 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: extract interlaced information from jpeg video
	  This information is hidden in a small chunk of data.
	  Format found at https://developer.apple.com/standards/qtff-2001.pdf,
	  page 92, "Video Sample Description", under table 3.1.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767771

2016-10-26 12:46:28 +0530  Jagadish <jagadishkamathk@gmail.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: Fixing x and y offset computation
	  While computing the x and y offsets, it's the video resolution and
	  resized overlay resolution to be used instead of actual overlay image
	  resoltuion. Due to this, the overlay image used to get wrongly overlayed
	  in undesired location
	  https://bugzilla.gnome.org/show_bug.cgi?id=757292

2016-11-01 18:09:00 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: update version

2016-10-24 16:56:31 +0000  Enrique Ocaña González <eocanha@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use the tfdt decode time on byte streams when it's significantly different than the time in the last sample
	  We consider there's a sifnificant difference when it's larger than on second
	  or than half the duration of the last processed fragment in case the latter is
	  larger.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754230

=== release 1.11.0 ===

2016-11-01 18:53:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.10.0 ===

2016-11-01 17:57:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.10.0

2016-11-01 17:47:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2016-11-01 17:41:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	  po: Update translations

2016-10-27 12:01:55 +0200  Tobias Schneider <tobias.schneider@voiceinterconnect.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: fix extra-controls leak
	  Gst struct v4l2object->extra_controls is created if user sets appropriate
	  option but it is not freed on destruction of v4l2object.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773580

2016-10-31 18:00:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  Revert "souphttpsrc: reduce reading latency by using non-blocking read"
	  This reverts commit 8816764112408766889c8b680a3af51115df4bf5.
	  It causes issues with the timeouts, and causes connections to be closed
	  without actual reason. Needs further investigation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773509

2016-10-31 09:00:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Don't try to add srcpad if we don't know valid caps yet
	  Otherwise we'll run into an assertion on specially crafted files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773643

2016-10-27 09:11:26 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/audiofx/gststereo.c:
	  Explicitly define float constants as float
	  With MSVC, this gives the following warning:
	  warning C4305: 'function': truncation from 'double' to 'gfloat'
	  Apparently, MSVC does not figure out what type to use for constants
	  based on the assignment. This warning is very spammy, so let's try to
	  fix it.

2016-10-27 11:23:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	  meson: Remove uselessly duplicated dep checks
	  These checks are done inside the meson.build files for each plugin.

2016-10-27 11:22:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/meson.build:
	  meson: dv plugin now works on MSVC
	  Needs a Meson patch to filter out the useless -lpthread
	  https://github.com/mesonbuild/meson/pull/962

2016-10-27 14:03:48 +0200  Branko Subasic <branko@axis.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: allow resolutions above 4096
	  Modify the caps string to allow width and height greater than 4096.
	  There is no need to restrict it since the matroska format allows the
	  width and height values to be up to eight bytes long.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773582

2016-10-23 17:23:10 -0700  Scott D Phillips <scott.d.phillips@intel.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Check for G_PLATFORM_WIN32 for presence of ipi_spec_dest
	  G_OS_WIN32 is only set when not building with cygwin, but
	  ipi_spec_dest is missing both with and without cygwin.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773114

2016-10-26 08:51:40 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: reset read_position when reading fails
	  souphttpsrc maintains two variables for the position:
	  * 'request_position' is where we want to be
	  * 'read_position' is where we are
	  During Normal operations both are updated in sync when data arrives. A seek
	  changes 'request_position' but not 'read_position'.
	  When the two positions get out of sync, then a new request is send and the
	  'Range' header is adjusted to the current 'request_position'.
	  Without this patch, if reading fails, then the source is destroyed. This
	  triggers a new request, but the range remains unchanged. As a result, the
	  old range is used and old data will be read.
	  Changing the 'read_position' to -1 makes it explicitly different from
	  'request_position' and as a result the 'Range' header is updated correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773509

2016-10-25 08:54:34 -0700  Scott D Phillips <scott.d.phillips@intel.com>

	* meson.build:
	  meson: Don't depend on gstreamer-check-1.0 on windows
	  https://bugzilla.gnome.org/show_bug.cgi?id=773114

2016-10-25 15:24:20 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: reset connection info to non-flushing when closing
	  This solves a hanging mainloop in following scenario:
	  * connect to source
	  * network/server drops
	  * pipeline set to NULL (and connection to flushing as part)
	  * pipeline set to PAUSED/PLAYING (connection to non-flushing, but not recorded)
	  * [connecting still not possible]
	  * pipeline set to NULL => mainloop hangs (since no actual flushing is done)

2016-10-26 14:32:48 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Only allow one video request pad
	  The pacing of the overall muxing is controlled
	  by the video GOPs arriving, so we can only handle
	  1 video stream, and the request pad is named accordingly.
	  Ignore a request for a 2nd video pad if there's already
	  an active one.

2016-10-26 11:59:32 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Take ownership of floating refs
	  sink the floating ref when handed a muxer or sink to use so
	  we clearly take ownership.

2016-10-25 14:51:52 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Set child elements to NULL when removing.
	  Make sure that elements are in the NULL state when removing.
	  Fixes critical warnings when errors occur early on in starting up.

2016-10-25 14:50:53 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Set pad template on request sink pads
	  Ensure that the ghost pad returned as a request pad
	  has the template that was requested

2016-10-25 10:50:47 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	* tests/check/meson.build:
	  Revert "meson: move gstreamer-check-1.0 dependency to tests/check"
	  This reverts commit 46632694662b96fddb848a1f2091a215b28a2d35.
	  Does not actually work. See:
	  https://bugzilla.gnome.org/show_bug.cgi?id=773114#c31

2016-06-08 11:24:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Assume PTS is DTS when PTS is missing
	  This fixes issue for encoders that only sets the DTS. We assume that
	  there was no re-ordering when that happens.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762207

2016-10-24 00:34:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/meson.build:
	  meson: fix build outside of gst-all

2016-10-21 00:42:54 -0700  Scott D Phillips <scott.d.phillips@intel.com>

	* sys/directsound/meson.build:
	  meson: directsound: Add ole32 library dependency
	  https://bugzilla.gnome.org/show_bug.cgi?id=773114

2016-10-21 00:42:18 -0700  Scott D Phillips <scott.d.phillips@intel.com>

	* meson.build:
	* tests/check/meson.build:
	  meson: move gstreamer-check-1.0 dependency to tests/check
	  https://bugzilla.gnome.org/show_bug.cgi?id=773114

2016-10-20 22:08:14 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/videomixer.c:
	  tests: videomixer: disable racy flush_start_flush_stop test
	  It's been broken for years, and it's unlikely it will ever
	  be fixed for collectpads/videomixer now that there's compositor
	  which works fine. So let's disable it, since all it does
	  is that it creates noise that distracts from other failures.
	  Also see the corresponding adder bug as it failed in the same way:
	  https://bugzilla.gnome.org/show_bug.cgi?id=708891

2016-10-09 16:56:10 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: Fix souphttpsrc tests without CK_FORK=no
	  It seems that the forked processes all attempt to handle the listening
	  socket from the server, and only one has to shutdown the socket to break
	  the server completely.
	  Create a new server inside each test to avoid this.
	  https://bugzilla.gnome.org/show_bug.cgi?id=772656

2016-10-09 15:23:51 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* tests/check/elements/level.c:
	  tests: Fix level test in CK_FORK=no mode
	  The tests accumulate buffers in GstCheck's buffers list, and the list is
	  not (consistently) reset between tests. Do that and remove the now
	  conflicting unrefs for outbuffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=772644

2016-10-07 13:04:27 +0530  Gaurav Gupta <g.gupta@samsung.com>

	* sys/waveform/gstwaveformsink.c:
	  waveformsink: Fix Memory leak using GST_PTR_FORMAT
	  https://bugzilla.gnome.org/show_bug.cgi?id=772497

2016-10-18 12:23:42 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/monoscope/meson.build:
	  meson: Add missing gstaudio dep to monoscope
	  In file included from ../subprojects/gst-plugins-good/gst/monoscope/gstmonoscope.c:42:0:
	  ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory
	  #include <gst/audio/audio-enumtypes.h>
	  ^
	  compilation terminated.
	  https://ci.gstreamer.net/job/GStreamer-master-meson/271/console

2016-10-16 12:40:22 +0200  Sergey Borovkov <sergey.borovkov@wireload.net>

	* ext/qt/qtwindow.cc:
	  qt: Fix failing build on RPI
	  https://bugzilla.gnome.org/show_bug.cgi?id=773026

2016-10-16 02:18:22 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/multifile/meson.build:
	  meson: Add missing pbutils dependency to multifile
	  Found via the Jenkins CI:
	  FAILED: subprojects/gst-plugins-good/gst/multifile/gstmultifile@sha/gstsplitmuxsink.c.o
	  [...]
	  In file included from ../subprojects/gst-plugins-good/gst/multifile/gstsplitmuxsink.h:24:0,
	  from ../subprojects/gst-plugins-good/gst/multifile/gstsplitmuxsink.c:59:
	  ../subprojects/gst-plugins-base/gst-libs/gst/pbutils/pbutils.h:30:43: fatal error: gst/pbutils/pbutils-enumtypes.h: No such file or directory
	  #include <gst/pbutils/pbutils-enumtypes.h>
	  ^
	  compilation terminated.
	  https://ci.gstreamer.net/job/GStreamer-master-meson/263/console

2016-10-15 22:11:08 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	  meson: Don't set c_std to gnu99
	  Use the default for each compiler on every platform instead. This
	  improves our compatibility with compilers that don't have gnu99 as
	  a c_std.

2016-10-04 18:04:11 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* meson.build:
	* tests/check/getpluginsdir:
	* tests/check/meson.build:
	  meson: Make use of new environment object and set plugin path to builddir
	  Workaround source_root being the root directory of all projects in the subproject
	  case and remove now unneeded getpluginsdir
	  Bump meson requirement to 0.35

2016-10-06 11:15:54 +0530  Gaurav Gupta <g.gupta@samsung.com>

	* tests/examples/rtp/client-rtpaux.c:
	  tests: Fix memory leak in test rtpaux test
	  https://bugzilla.gnome.org/show_bug.cgi?id=772496

2016-10-03 11:27:54 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Forward latency queries to upstream
	  Without this, latency queries to imagefreeze will fail.

2016-09-30 11:35:39 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* hooks/pre-commit.hook:
	* meson.build:
	* tests/check/getpluginsdir:
	  meson: Setup pre commit hook and fix getpluginsdir for standalone case

2016-09-29 04:55:14 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Handle stop point from segment
	  If the seek stop point (or start, during reverse play)
	  was within the segment we just finished, go EOS immediately
	  instead of proceeding through all other parts and sending
	  0 length seeks to them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=772138

2016-09-29 03:21:26 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Drop lock shutting down pads
	  Avoid a sporadic deadlock on shutdown by dropping
	  the splitmux lock around pad shutdown
	  https://bugzilla.gnome.org/show_bug.cgi?id=772138

2016-09-29 02:47:36 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxsrc: Fix extra unref handling queries
	  https://bugzilla.gnome.org/show_bug.cgi?id=772138

2016-09-29 04:50:25 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Avoid stall when parts get out of sync
	  When one part moves ahead of the others - due to excessive
	  downstream queueing, or really small input files - then
	  we can end up activating parts more than once. That can lead to
	  effects like shutting down pad tasks prematurely.
	  https://bugzilla.gnome.org/show_bug.cgi?id=772138

2016-09-30 11:41:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: update version

=== release 1.9.90 ===

2016-09-30 13:02:19 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.9.90

2016-09-30 12:17:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2016-09-30 11:43:54 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	  po: Update translations

2016-09-30 13:22:32 +0530  Arun Raghavan <arun@osg.samsung.com>

	* tests/check/pipelines/tagschecking.c:
	  tests: Fix tagschecking failure due to missing PTS
	  qtmux now needs the PTS (commit a993883b7), so let's make sure we
	  produce one with our buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=772228

2016-09-28 23:03:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't calculate PTS offset and DTS with GST_CLOCK_TIME_NONE
	  Just error out if there is no valid PTS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=772143

2016-09-29 17:37:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add JPEG2000 ihdr atom to the list of known ones
	  Otherwise qtdemux is always going to complain about it being unknown.

2016-09-29 10:19:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Always write the default frame duration for VP8/9 too
	  The WebM spec allows this now, and it allows us to guess a framerate.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=772141 and
	  also https://bugzilla.gnome.org/show_bug.cgi?id=654379

2016-09-27 15:26:19 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph265depay.c:
	  rtph26[45]depay: Don't handle NALs inside STAP units twice
	  They've already been handled before pushing them into the adapter.

2016-09-27 12:39:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/meson.build:
	  meson: tests: fix vp8 availability checks
	  Those variables are not defined if vp8 was not found.

2016-09-27 10:23:38 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	  Revert "multifilesink: streamline the file-switch code a bit"
	  This reverts commit f1ceaab02f3f557e23b77b14771a575788f92bb4.
	  This broke atomic file writes in "buffer" mode. It did make
	  sure that any streamheaders are prepended to each file in
	  buffer mode as well, but that's not really needed in practice,
	  whereas atomic file writes are, so let's restore the status
	  quo ante for now since this was primarily a code cleanup anyway,
	  and if anyone needs to streamheaders in buffer mode too they
	  can make a patch to implement that differently. Re-implementing
	  the atomic writes in the element also seems way too much work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766990

2016-09-27 10:22:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	  Revert "multifilesink: close file on write error with next-file mode is set to buffer"
	  This reverts commit 84e441d2685cf223d348a95be0c5ba693bbf6624.
	  This will no longer be needed once we revert f1ceaab02.

2016-09-26 13:22:29 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/check/meson.build:
	  meson: Add gst-plugins-base plugins directories to be used by tests

2016-09-26 14:30:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/meson.build:
	* meson.build:
	* tests/check/getpluginsdir:
	* tests/check/meson.build:
	  meson: add unit tests
	  Only works properly in an installed setup currently, most
	  likely won't work with a subprojects setup yet.

2016-09-24 09:36:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* po/meson.build:
	  meson: hook up translations

2016-09-08 17:30:41 +0530  Arun Raghavan <arun@arunraghavan.net>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Don't negotiate to less than two segments
	  GstAudioRingBuffer doesn't needs us to have at least 2 segments. We make
	  sure that if our buffer parameters are such that the maxlength is not at
	  least 2x fragsize, we still request the ringbuffer to keep that much
	  space so it continues to work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=770446

2016-09-24 23:22:01 +0530  Arun Raghavan <arun@arunraghavan.net>

	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: Fix timestamping
	  We were just picking the timestamp of the last buffer pushed into our
	  adapter before we had enough data to push out.
	  This fixes things to figure out how large each frame is and what
	  duration it covers, so we can set both the timestamp and duration
	  correctly.
	  Also adds some DISCONT handling.

2016-07-12 18:14:52 +0200  Georg Lippitsch <glippitsch@toolsonair.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix fourcc for ProRes Proxy
	  This is apco, according to
	  https://wiki.multimedia.cx/index.php?title=Apple_ProRes
	  https://bugzilla.gnome.org/show_bug.cgi?id=769048

2016-09-18 20:55:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/meson.build:
	  meson: fix build with vpx 1.3.x
	  vpx >= 1.4.0 is optional

2016-09-15 18:19:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use new bin suppressed flags API for managing the element flags

2016-09-15 09:52:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/jack/gstjackaudioclient.c:
	* gst/rtp/dboolhuff.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/videofilter/gstvideoflip.c:
	  ext, gst: fix indentation

2016-09-15 09:52:17 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/flvmux.c:
	* tests/check/elements/rtph263.c:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/rtpsession.c:
	* tests/check/elements/rtpvp9.c:
	  tests: fix indentation

2016-08-11 11:04:22 -0600  Thomas Bluemel <tbluemel@control4.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix calculating next_seqnum when dropping old buffers from a full queue.
	  Fixes calculating the next sequence number when a ITEM_TYPE_LOST with more than one
	  definitely lost packets is encountered.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769757

2016-08-11 23:07:44 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: improved rtx-rtt averaging
	  The basic idea is this:
	  1. For *larger* rtx-rtt, weigh a new measurement as before
	  2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less
	  3. For very large measurements, consider them "outliers"
	  and count them a lot less
	  The idea being that reducing the rtx-rtt is much more harmful then
	  increasing it, since we don't want to be underestimating the rtt of the
	  network, and when using this number to estimate the latency you need for
	  you jitterbuffer, you would rather want it to be a bit larger then a bit
	  smaller, potentially losing rtx-packets. The "outlier-detector" is there
	  to prevent a single skewed measurement to affect the outcome too much.
	  On wireless networks, these are surprisingly common.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-08-05 12:51:59 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Detect whether to assume equidistant spacing when loss
	  Assuming equidistant packet spacing when that's not true leads to more
	  loss than necessary in the case of reordering and jitter. Typically this
	  is true for video where one frame often consists of multiple packets
	  with the same rtp timestamp. In this case it's better to assume that the
	  missing packets have the same timestamp as the last received packet, so
	  that the scheduled lost timer does not time out too early causing the
	  packets to be considered lost even though they may arrive in time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-27 10:39:50 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Don't request rtx if 'now' is past retry period
	  There is no need to schedule another EXPECTED timer if we're already
	  past the retry period. Under normal operation this won't happen, but if
	  there are more timers than the jitterbuffer is able to process in
	  real-time, scheduling more timers will just make the situation worse.
	  Instead, consider this packet as lost and move on. This scenario can
	  occur with high loss rate, low rtt and high configured latency.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-26 18:01:48 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix lost duration when gap after lost timer
	  This patch fixes an issue with the estimated gap duration when there is
	  a gap immediately after a lost timer has been processed. Previously
	  there was a discrepancy beteen the gap in seqnum and gap in dts which
	  would cause wrong calculated duration. The issue would only be seen with
	  retranmission enabled since when it's disabled lost timers are only
	  created when a packet is received and the actual gap length and last dts
	  is known.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-19 01:11:58 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Expose rtx-deadline as a property
	  The default -1 gives the old behavior.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-08-11 12:02:19 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Improved expected-timer handling when gap > 0
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-08-11 11:51:50 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Major improvements for RTX stats
	  Stats should also be collected for unsuccessful packets.
	  rtx-rtt is very important for determining the necessary configured
	  latency on the jitterbuffer. It's especially important to be able to
	  increase the latency when retransmitted packets arrive too late and are
	  considered lost. This patch includes these late packets in the
	  calculation of the various rtx stats, making them more correct and
	  useful.
	  Also in the case where the original packet arrives after a NACK is sent,
	  the received RTX packet should update the stats since it provides useful
	  information about RTT.
	  The RTT is only updated if and only if all requested retranmissions are
	  received. That way the RTT is guaranteed to make sense. If not we don't
	  know which request the packet is a response to and the RTT may be bogus.
	  A consequence of this patch is that RTT is not updated for a request
	  when one of the RTX packets for that seqnum is lost, but that since
	  measured RTT will be more accurate.
	  The implementation store the RTX information from the timed out timers
	  and use this when the retransmitted packet arrives. For performance
	  these timers are stored separately from the "normal" timers in order to
	  not impact performance (see attached performance test).
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-08-11 11:02:44 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Add and expose more stats and increase testing of it
	  Add num-pushed and num-lost.
	  Expose num-late, num-duplicates and avg-jitter.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-07 10:20:02 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtxreceive: Set buffer flag for retransmitted packets
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-09 23:47:41 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Option to disable rtx-delay-reorder
	  When disabled we can save some iterations over timers.
	  There is probably an argument for rtx-delay-reorder to exist, but
	  for normal operations, handling jitter (reordering) is something a
	  jitterbuffer should do, and this variable feels like functionality that
	  is not "in-sync" with what the jitterbuffer is trying to achieve.
	  Example: You have 50ms jitter on your network, and are receiving
	  audio packets with 10ms durations. An audio packet should not be
	  considered late until its rtx-timeout has expired (and hence a rtx-event
	  is sent), but with rtx-delay-reorder, events will be sent pretty much
	  all the time due to the jitter on the network.
	  Point being: The jitterbuffer should adapt its size to the measured network
	  jitter, and then rtx-delay-reorder needs to adapt as well, or simply
	  get out of the way and let the other (better) rtx-mechanisms do their job.
	  Also change find_timer to only use seqnum as an argument, since there
	  will only ever be one timer per seqnum at any given time. In the
	  one case where the type matters, the caller simply checks the type.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-09-14 09:58:41 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: Fix double free from coverity
	  CID #1372887

2016-09-14 09:58:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: Indent as per gst-indent

2016-09-14 11:30:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  configure: Depend on gstreamer 1.9.2.1

2016-09-14 10:17:02 +0900  Wonchul Lee <wonchul.lee@collabora.com>

	* gst/autodetect/gstautodetect.c:
	  autodetect: Use gst_bin_set_suppressed_flags() API
	  https://bugzilla.gnome.org/show_bug.cgi?id=771395

2016-09-09 15:36:12 +0200  Thomas Scheuermann <Thomas.Scheuermann@barco.com>

	* ext/jack/gstjackaudioclient.c:
	  jack: Fix pipeline hang when jack changes sample rate or buffer size
	  If jackd changes the buffer size or sample rate, jackaudiosink hangs
	  and can't be stopped. This also happens if jack is configured as slave
	  and a gstreamer pipeline is started on the slave machine while the jack
	  master isn't running yet. If the the jack master is started it changes
	  the buffer size / sample rate and jackaudiosink can't be stopped.
	  This fix calls jack_shutdown_cb when jack_sample_rate_cb or
	  jack_buffer_size_cb is called.
	  https://bugzilla.gnome.org/show_bug.cgi?id=771272

2016-09-12 20:08:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix field ordering for reverse playback
	  And actually calculate the field duration instead of a frame duration so
	  that we can properly timestamp output frames in fields=all mode.
	  This is probably still broken for reverse playback in telecine mode.

2016-09-12 09:02:00 +0000  Thomas Klausner <tk@giga.or.at>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix compilation on NetBSD
	  https://bugzilla.gnome.org/show_bug.cgi?id=771278

2016-09-10 20:51:10 +1000  Jan Schmidt <jan@centricular.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From b18d820 to f980fd9

2016-09-09 14:02:25 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: offset is irrelevant when no crypto info
	  Cause later it will try to use the crypto info array to get an index and
	  attach on of the positions as buffer's crypto info.
	  https://bugzilla.gnome.org/show_bug.cgi?id=770951

2016-09-10 09:53:57 +1000  Jan Schmidt <jan@centricular.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From f49c55e to b18d820

2016-09-09 16:36:03 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/meson.build:
	  meson: add build files for the gtk plugin

2016-09-07 15:33:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/osxaudio/Makefile.am:
	  osxaudio: Distribute device provider files
	  Those where missing the the dev release tarballs for 1.9.2 which
	  prevented building from tarball on OSX platform

2016-09-06 09:49:39 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix crash with no cenc aux offset
	  https://bugzilla.gnome.org/show_bug.cgi?id=770951

2016-09-06 13:13:39 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/qt/gstqsgtexture.cc:
	  qmlglsink: check qt_context_ first in GstQSGTexture::bind()
	  When start qmlglsink app, it will set NULL buffer to GstQSGTexture
	  in which case that qt_context_ will be a random value and cause
	  gst_gl_context_activate() fail.
	  https://bugzilla.gnome.org/show_bug.cgi?id=770925

2016-09-05 09:39:33 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: parse a bit more of the humongous LOAS data
	  https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:39:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: make it clear when a potential LOAS frame is not one
	  https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:38:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: add a few comments to anchor parsing to the spec
	  https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:37:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	  aacparse: improve channel/rate handling
	  Keep track of the last parsed channels/rate fields so they can be
	  used even if the element was not yet configured.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:35:53 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix varlength number reading as per spec
	  https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:35:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: strip uneeded static arrays slack
	  https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-07-18 19:18:58 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4adepay.h:
	  rtpmp4adepay: Only declare a stream to be framed once a marker bit has been seen
	  This may cause a few packets to be processed by the parser, but it's
	  better than never pushing out buffers from a slightly broken stream
	  where no marker bits are set.

2016-09-06 14:25:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Fix timestamping in reverse playback mode
	  This is only supported right now if after a demuxer that supports reverse
	  playback, e.g. with DV container inside AVI container.

2016-09-05 12:23:54 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* meson.build:
	  meson: Bump version to 1.9.2

2015-06-26 20:13:17 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/isomp4/GstQTMux.prs:
	* gst/isomp4/Makefile.am:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Implement the preset interface.
	  + And provide a "youtube" preset, which based on
	  https://support.google.com/youtube/answer/1722171 sets
	  faststart to True.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751559

2016-09-01 12:27:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.9.2 ===

2016-09-01 12:27:15 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.9.2

2016-09-01 11:23:33 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: Update translations

2016-09-01 10:59:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/equalizer/demo.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  tests/examples: #define GDK_DISABLE_DEPRECATION_WARNINGS
	  We use gdk_cairo_create() which is deprecated since 3.22.

2016-08-31 05:50:44 +1000  Jan Schmidt <jan@centricular.com>

	* sys/osxvideo/Makefile.am:
	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/osxvideosink.h:
	  osxvideo: Remove QuickTime references.
	  QuickTime.h is no longer available on OS X 10.12 (Sierra),
	  and both the header and the framework seem unnecessary
	  for compilation - at least as of 10.11 (El Capitan).
	  https://bugzilla.gnome.org/show_bug.cgi?id=770526

2016-08-19 11:11:03 -0700  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* ext/dv/gstdvdemux.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/flv/gstflvdemux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/multifile/gstsplitmuxsrc.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/wavparse/gstwavparse.c:
	  Use the new API to post flow ERROR messages on the bus
	  https://bugzilla.gnome.org/show_bug.cgi?id=770158

2016-08-26 21:32:07 +0200  Josep Torra <n770galaxy@gmail.com>

	* tests/check/elements/.gitignore:
	  gitignore: ignore qtdemux, rtph261 and rtpvp9 tests

2016-08-26 21:22:16 +0200  Josep Torra <n770galaxy@gmail.com>

	* tests/check/Makefile.am:
	  tests: use GST_NET_LIBS instead of hardcoded -lgstnet
	  Fixes build in OSX when running 'make check' in gst-uninstalled.

2016-08-26 21:14:47 +0200  Josep Torra <n770galaxy@gmail.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: remove a wrong 'const' specifier
	  Fixes "error: duplicate 'const' declaration specifier"

2016-08-26 21:11:59 +0200  Josep Torra <n770galaxy@gmail.com>

	* configure.ac:
	* tests/check/Makefile.am:
	  build: silence error about pthread for 'make check' in osx
	  Fixes "clang: error: argument unused during compilation: '-pthread'"

2016-08-26 20:31:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/Makefile.am:
	  vp9enc: Fix build of unit test by letting it link to libgstvideo

2016-08-26 12:06:35 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  Revert "rtpmux: fix PROP_TIMESTAMP_OFFSET range problems"
	  This broke API, so we need a better solution!
	  This reverts commit c7579d31a6e9d788e94b83258309063d0aae481e.

2016-06-08 15:06:28 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp9depay.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpvp9.c:
	  rtpvp9depay: Support flexible mode

2016-06-06 17:03:36 +0200  Stian Selnes <stian@pexip.com>

	* ext/vpx/gstvp9enc.c:
	* tests/check/Makefile.am:
	* tests/check/elements/vp9enc.c:
	  vp9enc: Fix leak of vpx_image_t

2016-05-06 13:33:22 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph263pdepay.c:
	* tests/check/elements/rtph263.c:
	  rtph263pdepay: Don't try to push empty frame
	  If the result of depayloading is an empty frame, just drop it. This is
	  likely the result of a buggy payloader.

2016-05-06 16:06:53 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: fix PROP_TIMESTAMP_OFFSET range problems
	  It could not set the offset for the full guint32 range.

2016-05-06 09:44:42 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: introduce max-streams property
	  To be able to cap the number of allowed streams for one session.
	  This is useful for preventing DoS attacks, where a sender can change
	  SSRC for every buffer, effectively bringing rtpbin to a halt.
	  https://bugzilla.gnome.org/show_bug.cgi?id=770292

2016-03-31 00:10:49 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: reordered packets are very normal, and should not be a warning

2016-02-05 14:19:25 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: degrade g_warning to GST_ERROR
	  So we don't blow up while investigating

2016-02-04 14:16:40 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph263pdepay.c:
	* tests/check/elements/rtph263.c:
	  rtph263pdepay: Fix picture header for non-writable payload
	  Under certain conditions gst_rtp_buffer_get_payload() returns a copy of
	  the payload. In this case the payload modifications will not affect the
	  rtp buffer. So instead of modifying the payload buffer directly we
	  should modify the buffer that actually gets pushed on the adapter.

2015-11-19 11:50:47 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph261depay.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtph261.c:
	  rtph261depay: Fix check of valid payload length
	  Packets with no H.261 payload should be dropped to avoid invalid
	  write/reads.

2015-11-09 10:06:21 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph263pay.c:
	* tests/check/elements/rtph263.c:
	  rtph263pay: Fix double free, invalid reads and leak

2014-06-30 15:43:58 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: sanity check RTT before ignoring PLI/FIR

2014-06-30 15:07:45 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: handle sdes messages with non-utf8 more gracefully

2014-06-17 08:52:50 +0200  Stian Selnes <stian.selnes@gmail.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: change log level on bitstream parsing messages

2016-07-07 11:13:18 +0200  Mikhail Fludkov <misha@pexip.com>

	* tests/check/elements/rtprtx.c:
	  tests/rtprtx: refactor the tests to use gstharness
	  The functionality of all the tests was kept exactly the same. Some tests
	  were renamed:
	  test_push_forward_seq -> test_rtxsend_rtxreceive
	  test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
	  test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss
	  test_rtxreceive_data_reconstruction was testing that retransmitted
	  buffer produced by rtxsend was correctly transformed to the original
	  buffer by rtxreceive. Now we are checking for this in all the tests
	  where both rtxsend & rtxreceive are involved. That's why the test was
	  removed.

2016-08-25 15:52:36 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Set RTP marker bit
	  Set the RTP marker bit on the last RTP packet of an H.265 access unit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=770394

2016-07-26 19:39:58 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	  videoflip: added GstVideoDirection interface
	  It implements now this interface with its video-direction
	  property. Values are changed to GstVideoOrientationMethod but they have
	  the same value than the originals.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768687

2015-11-06 10:39:16 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: refactor duplicate code into a function
	  Less code, easier to read, more consistent.
	  https://bugzilla.gnome.org/show_bug.cgi?id=770293

2016-08-23 17:06:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix typo in max-misorder-time property name

2016-08-22 00:05:52 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: fix printf format compiler warning in debug message
	  On 32-bit x86: gstsplitmuxsink.c:966:31: warning: format ‘%u’ expects
	  argument of type ‘unsigned int’, but argument 9 has type
	  ‘guint64 {aka long long unsigned int}’

2016-08-12 21:25:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/twolame/meson.build:
	  Add support for Meson as alternative/parallel build system
	  https://github.com/mesonbuild/meson
	  With contributions from:
	  Tim-Philipp Müller <tim@centricular.com>
	  Jussi Pakkanen <jpakkane@gmail.com> (original port)
	  Highlights of the features provided are:
	  * Faster builds on Linux (~40-50% faster)
	  * The ability to build with MSVC on Windows
	  * Generate Visual Studio project files
	  * Generate XCode project files
	  * Much faster builds on Windows (on-par with Linux)
	  * Seriously fast configure and building on embedded
	  ... and many more. For more details see:
	  http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
	  http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
	  Building with Meson should work on both Linux and Windows, but may
	  need a few more tweaks on other operating systems.

2016-08-12 21:25:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/lame/meson.build:
	  Add support for Meson as alternative/parallel build system
	  https://github.com/mesonbuild/meson
	  With contributions from:
	  Tim-Philipp Müller <tim@centricular.com>
	  Jussi Pakkanen <jpakkane@gmail.com> (original port)
	  Highlights of the features provided are:
	  * Faster builds on Linux (~40-50% faster)
	  * The ability to build with MSVC on Windows
	  * Generate Visual Studio project files
	  * Generate XCode project files
	  * Much faster builds on Windows (on-par with Linux)
	  * Seriously fast configure and building on embedded
	  ... and many more. For more details see:
	  http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
	  http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
	  Building with Meson should work on both Linux and Windows, but may
	  need a few more tweaks on other operating systems.

2016-08-12 21:25:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/mpg123/meson.build:
	  Add support for Meson as alternative/parallel build system
	  https://github.com/mesonbuild/meson
	  With contributions from:
	  Tim-Philipp Müller <tim@centricular.com>
	  Jussi Pakkanen <jpakkane@gmail.com> (original port)
	  Highlights of the features provided are:
	  * Faster builds on Linux (~40-50% faster)
	  * The ability to build with MSVC on Windows
	  * Generate Visual Studio project files
	  * Generate XCode project files
	  * Much faster builds on Windows (on-par with Linux)
	  * Seriously fast configure and building on embedded
	  ... and many more. For more details see:
	  http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
	  http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
	  Building with Meson should work on both Linux and Windows, but may
	  need a few more tweaks on other operating systems.

2016-08-12 21:12:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* .gitignore:
	* config.h.meson:
	* ext/cairo/meson.build:
	* ext/dv/meson.build:
	* ext/flac/meson.build:
	* ext/gdk_pixbuf/meson.build:
	* ext/jack/meson.build:
	* ext/jpeg/meson.build:
	* ext/libpng/meson.build:
	* ext/meson.build:
	* ext/pulse/meson.build:
	* ext/shout2/meson.build:
	* ext/soup/meson.build:
	* ext/speex/meson.build:
	* ext/taglib/meson.build:
	* ext/vpx/meson.build:
	* ext/wavpack/meson.build:
	* gst/alpha/meson.build:
	* gst/apetag/meson.build:
	* gst/audiofx/meson.build:
	* gst/audioparsers/meson.build:
	* gst/auparse/meson.build:
	* gst/autodetect/meson.build:
	* gst/avi/meson.build:
	* gst/cutter/meson.build:
	* gst/debugutils/meson.build:
	* gst/deinterlace/meson.build:
	* gst/dtmf/meson.build:
	* gst/effectv/meson.build:
	* gst/equalizer/meson.build:
	* gst/flv/meson.build:
	* gst/flx/meson.build:
	* gst/goom/meson.build:
	* gst/goom2k1/meson.build:
	* gst/icydemux/meson.build:
	* gst/id3demux/meson.build:
	* gst/imagefreeze/meson.build:
	* gst/interleave/meson.build:
	* gst/isomp4/meson.build:
	* gst/law/meson.build:
	* gst/level/meson.build:
	* gst/matroska/meson.build:
	* gst/meson.build:
	* gst/monoscope/meson.build:
	* gst/multifile/meson.build:
	* gst/multipart/meson.build:
	* gst/replaygain/meson.build:
	* gst/rtp/meson.build:
	* gst/rtpmanager/meson.build:
	* gst/rtsp/meson.build:
	* gst/shapewipe/meson.build:
	* gst/smpte/meson.build:
	* gst/spectrum/meson.build:
	* gst/udp/meson.build:
	* gst/videobox/meson.build:
	* gst/videocrop/meson.build:
	* gst/videofilter/meson.build:
	* gst/videomixer/meson.build:
	* gst/wavenc/meson.build:
	* gst/wavparse/meson.build:
	* gst/y4m/meson.build:
	* meson.build:
	* meson_options.txt:
	* sys/directsound/meson.build:
	* sys/meson.build:
	* sys/v4l2/meson.build:
	* sys/ximage/meson.build:
	* tests/check/meson.build:
	* tests/meson.build:
	  Add support for Meson as alternative/parallel build system
	  https://github.com/mesonbuild/meson
	  With contributions from:
	  Tim-Philipp Müller <tim@centricular.com>
	  Jussi Pakkanen <jpakkane@gmail.com> (original port)
	  Highlights of the features provided are:
	  * Faster builds on Linux (~40-50% faster)
	  * The ability to build with MSVC on Windows
	  * Generate Visual Studio project files
	  * Generate XCode project files
	  * Much faster builds on Windows (on-par with Linux)
	  * Seriously fast configure and building on embedded
	  ... and many more. For more details see:
	  http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
	  http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
	  Building with Meson should work on both Linux and Windows, but may
	  need a few more tweaks on other operating systems.

2016-08-20 16:59:30 +0800  Jie Jiang <jiangjie@nudt.edu.cn>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  Fixed splitmuxsink 32-bit overflow bug
	  Extend the byte tracking counters to 64-bit on
	  all platforms, instead of using gsize, which overflows
	  after 4GB.
	  https://bugzilla.gnome.org/show_bug.cgi?id=770019

2016-08-19 17:18:16 +0300  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/isomp4/atoms.c:
	  isomp4: Fix coverity warning
	  If atom_copy_data fails to write anything, return 0
	  CID #1371458

2016-04-09 07:51:03 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/v4l2-utils.c:
	  v4l2: consistently check #ifdef HAVE_GUDEV instead of #if
	  Both work with autotools but they definitely don't mean the same thing, cause
	  problems with other build systems, and are bad form. Existence should always be
	  checked with #ifdef or #if defined.

2016-04-19 10:53:05 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  directsound: port away from old DirectX API
	  D3DX has been deprecated for the last 4 years and latest versions of
	  Windows no longer ship headers for it. This is fine as long as you're
	  building with Cerbero's Wine-based DirectX headers, but sucks if you
	  want to build against the actual Windows SDK.
	  We were just using it to get error strings anyway, so just use the
	  generic error string API.

2016-08-18 12:02:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  Revert "flacparse: Add maximum bitrate tag"
	  This reverts commit c703ab69f526092bb26cce41ca691a896c8383d8.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769392

2016-08-18 09:57:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix unit test by disabling adaptive misorder/dropout calculations
	  Need to set max-misorder-time and max-dropout-time to 0 so the
	  jitterbuffer does not base them on packet rate calculations.
	  If it does, out gap is big enough to be considered a new stream and
	  we wait for a few consecutive packets just to be sure
	  https://bugzilla.gnome.org/show_bug.cgi?id=751311

2016-08-09 12:55:59 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Add option to split at exactly max-size-time
	  Will try to request a keyframe from the encoder to be sent at the target
	  running time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769664

2016-08-09 20:16:16 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Allow time and bytes to reach their respective thresholds
	  https://bugzilla.gnome.org/show_bug.cgi?id=769664

2016-08-17 09:49:04 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Allow mimetypes with properties as long as they're application/sdp
	  Some servers add properties like charset, e.g.
	  application/sdp; charset=utf8
	  Ideally we should also parse the charset and do conversion of all messages,
	  but that's for a later time.

2016-06-24 16:32:37 +0300  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Added support for writing timecode track
	  https://bugzilla.gnome.org/show_bug.cgi?id=767950

2016-08-16 00:40:53 +1000  Jan Schmidt <jan@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qt: Use wglShareLists() workaround unconditionally.
	  Sometimes wglCreateContextAttribsARB() exists, but
	  isn't functional (some Intel drivers), so it's
	  easiest to do the workaround unconditionally.

2016-08-08 13:41:14 +1000  Jan Schmidt <jan@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qt: Move debug statement to after the category init
	  Don't output debug to an uninitialised debug category.

2016-08-11 16:32:21 -0600  Thomas Bluemel <tbluemel@control4.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Initialize bytes_sent field.
	  This fixes endpoints not receiving any data intermittently.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769773

2016-08-10 11:45:13 -0600  Thomas Bluemel <tbluemel@control4.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpstats.c:
	  rtpjitterbuffer: Actually calculate the packet rate for max-dropout and max-misorder calculations.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751311

2016-08-10 11:26:17 -0600  Thomas Bluemel <tbluemel@control4.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Don't warn for duplicate packets
	  This is a normal scenario and should not be a warning.  This can
	  happen frequently when re-transmits of lost packets are enabled.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762208

2016-08-08 13:49:19 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: Fix typo converting to running time.
	  Use the correct collected timestamp.

2016-08-08 02:53:48 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  Revert "splitmuxsink: Use GstBin async-handling instead of our own."
	  This reverts commit fa008f271a52f82dededc28bd81b020ca7939b47.
	  async-handling in GstBin causes the pipeline to spin at 100%
	  CPU as the top-level pipeline tries to change that state
	  to PLAYING constantly. This is a workaround for a core
	  problem, essentially, but an improvement in this case for now.

2016-08-08 00:56:38 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: Recheck state after unlocking mutex.
	  After dropping the splitmux lock, re-check the state,
	  don't just fall through and sleep unconditionally,
	  as we may have already missed the wakeup.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769514

2016-08-03 03:32:07 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Don't stop and error on EOS flow return
	  Don't immediately halt on EOS flow return from downstream
	  due to out of segment. Let the demuxer handle it and send
	  EOS.

2016-08-04 00:36:28 -0300  Thiago Santos <thiagossantos@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: avoid unref of null buffer
	  The current 'l' pointer will be NULL when the loop
	  is interrupted with a 'break' statement. Need to have
	  it advance to the next list item before interrupting.

2016-07-27 09:28:23 +0800  Haihua Hu <jared.hu@nxp.com>

	* tests/examples/qt/qmlsink/.gitignore:
	* tests/examples/qt/qmlsink/main.cpp:
	* tests/examples/qt/qmlsink/main.qml:
	* tests/examples/qt/qmlsink/play.pro:
	* tests/examples/qt/qmlsink/qml.qrc:
	* tests/examples/qt/qmlsrc/.gitignore:
	* tests/examples/qt/qmlsrc/grabqml.pro:
	* tests/examples/qt/qmlsrc/main.cpp:
	* tests/examples/qt/qmlsrc/main.qml:
	* tests/examples/qt/qmlsrc/qml.qrc:
	  qmlglsrc: Add qmlglsrc unit test example
	  https://bugzilla.gnome.org/show_bug.cgi?id=768160

2016-07-27 08:16:47 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/qt/Makefile.am:
	* ext/qt/gstplugin.cc:
	* ext/qt/gstqtglutility.cc:
	* ext/qt/gstqtglutility.h:
	* ext/qt/gstqtsrc.cc:
	* ext/qt/gstqtsrc.h:
	* ext/qt/qtitem.cc:
	* ext/qt/qtwindow.cc:
	* ext/qt/qtwindow.h:
	  qt: implement qmlglsrc for qml view grab
	  [Matthew Waters]: gst-indent sources
	  https://bugzilla.gnome.org/show_bug.cgi?id=768160

2016-08-02 14:01:14 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* gst/wavparse/Makefile.am:
	* gst/wavparse/gstwavparse.c:
	  wavparse: Add tags for container format and bitrate for uncompressed PCM
	  The PCM bitrate is added to help downstream elements (like uridecodebin)
	  figure out a proper network buffer size
	  https://bugzilla.gnome.org/show_bug.cgi?id=769390

2016-08-01 18:52:26 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Add maximum bitrate tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=769392

2016-07-28 17:58:16 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: When receiving a DISCONT buffer that does not point to a sample, remember the offset
	  And don't just reset everything. This makes sure that we can continue to
	  handle data in the following scenario:
	  moov: discont
	  moof: discont
	  mdat: continuous
	  Previously this would fail because the offset would be the accumulated offset
	  from moov and moof at the mdat position, while the buffer offset might be
	  something completely different.

2016-07-25 13:34:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpilbcpay.c:
	  rtp: Filter with the filter caps in the payloader's getcaps

2016-03-03 11:35:06 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: include http-status-code in error message details
	  https://bugzilla.gnome.org/show_bug.cgi?id=763038

2016-07-25 18:20:03 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix debug statement signedness.
	  The ts variable is a GstClockTime, don't print it
	  as a GstClockTimeDiff.

2016-07-22 17:00:14 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/qt/qml/main.cpp:
	  qml: Don't forget to unref the actual sink element after setting it on glsinkbin

2016-07-22 16:57:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/qt/qml/main.cpp:
	  qml: Use glsinkbin instead of glupload directly

2016-07-17 22:41:02 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Handle negative running time
	  Use signed clock times for running time everywhere
	  so that we handle negative running times without
	  going haywire, similar to what queue and multiqueue
	  do these days.

2016-07-18 00:12:55 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Drop lock when sending dummy event
	  When pushing the dummy event into the multiqueue,
	  drop the splitmux lock or else we might deadlock.

2016-06-30 01:56:41 +1000  Jan Schmidt <thaytan@noraisin.net>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Intersect with filter caps in getcaps function.
	  Always intersect with the filter caps in the getcaps function
	  to make sure we return a subset of what was requested.
	  Other payloaders also have this problem and need fixing
	  in future commits.

2016-07-12 17:30:56 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/qtdemux.c:
	  tests: qtdemux: fix element and pad leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=768739

2016-07-12 16:45:36 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/audiofirfilter.c:
	* tests/check/elements/audioiirfilter.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/videobox.c:
	* tests/check/pipelines/effectv.c:
	  tests: fix bus leaks
	  gst_bus_add_signal_watch() takes a ref on the bus which should be
	  released using gst_bus_remove_signal_watch().
	  https://bugzilla.gnome.org/show_bug.cgi?id=768739

2016-07-14 03:07:11 +0800  Ting-Wei Lan <lantw@src.gnome.org>

	* configure.ac:
	  configure: Call AG_GST_PKG_CONFIG_PATH to set GST_PKG_CONFIG_PATH
	  GST_PKG_CONFIG_PATH is used in docs/plugins directory, so
	  AG_GST_PKG_CONFIG_PATH must be called to set it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768787

2016-07-12 07:39:58 +0200  Edward Hervey <edward@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't drop final bytes of a range request
	  At the end of a range request, we don't want to return GST_FLOW_EOS otherwise
	  the last bytes we just read will be dropped by basesrc.
	  Instead just return GST_FLOW_OK (which was set just before) and let basesrc
	  handle the fact we are at the end of the segment.

2016-07-11 18:30:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2provider: Fix device type detection
	  The type detection would lead to assertion as it would try
	  to create a device without having found any type for it. It
	  also didn't detect MPLANE devices properly.

2016-07-11 18:29:01 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't assert when used by the monitor
	  The monitor sets the object->element object as a GstObject. This
	  works for debug traces, but will assert for ELEMENT_ERROR. This
	  was the only case where that could happen. Add a check for that.

2016-07-11 17:38:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Indent very long line

2016-07-12 00:42:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: At the end of a range request, read another time to finalize the request
	  If we're at the end of a range request, read again to let libsoup
	  finalize the request. This allows to reuse the connection again later,
	  otherwise we would have to cancel the message and close the connection.

2016-07-11 21:13:47 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From f363b32 to f49c55e

2016-07-11 19:57:18 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Fix keep-alive handling
	  We have to get rid of the message on EOS when the complete stream is read to
	  remember that we successfully finished handling this specific message.
	  Otherwise we will cancel it later and close the connection instead of reusing
	  it at a later time.
	  It might also make sense to reuse connections if a non-200 response is
	  received. As long as there was no connection error, the HTTP connection should
	  be re-usable.

2016-07-11 12:05:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	  Also enable V4L2 probe on aarch64 (aka ARM 64bit)

2016-07-11 11:59:19 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/examples/rtp/client-PCMA.c:
	  rtp example: Fix leak
	  Also stop fetching the internal source as this
	  functionality has been broken.

2016-07-08 14:58:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	  Enable v4l2 probe on Linux/ARM
	  Most of those have V4L2 drivers these days enabling it make sure that it
	  this code is enabled in major distribution, hence that HW accelerated
	  decoder/encoder can be used on platforms that support it. The probes are
	  slightly increasing the first init of gstreamer library, though the
	  result is cached in the registry for later use.

2016-07-11 09:46:49 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph265pay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtph265pay: Accept array_completeness=1
	  When parsing NAL unit type in codec_data, check the 6bits of
	  NAL_unit_type only and do not require the array_completeness bit to be
	  0, since the default and mandatory value of array_completeness is 1 for
	  hvc1.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768653

2016-07-10 21:35:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Also copy device_caps in gst_v4l2_dup
	  This fixes regression where M2M error out saying they have no output
	  format (the V4L2 CAPTURE side).
	  https://bugzilla.gnome.org/show_bug.cgi?id=768195

2016-07-10 21:30:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Use correct in6_pktinfo struct instead of in_pktinfo
	  Fixes the build on FreeBSD, which does not have the latter.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768623

2016-07-08 17:28:19 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: fix multiplanar capture
	  After switching to using V4L2_CAP_DEVICE_CAPS we lost support for
	  multiplanar device types. After some research, it looks like
	  vcap.capabilities treated the multiplanar flag of output and capture
	  devices equally, but not the new device_caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768195

2016-07-08 14:56:30 +0200  Mats Lindestam <matslm@axis.com>

	* gst/multipart/multipartmux.c:
	* gst/multipart/multipartmux.h:
	  multipartmux: Use PTS and DTS instead of timestamp
	  And pass-through both of them.
	  Based on a patch by Göran Jönsson <goranjn@axis.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=767900

2016-06-30 14:40:40 +0200  Thomas Scheuermann <Thomas.Scheuermann@barco.com>

	* ext/jack/gstjackaudioclient.c:
	  jack: don't wait for callbacks if the jack server shut down
	  Otherwise we'll wait forever.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747275

2016-06-23 15:30:19 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Let upstream events go through upstream
	  There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
	  Some elements might want to have that information.

2016-06-23 15:22:56 +0200  Edward Hervey <edward@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Let upstream events go through upstream
	  There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
	  Some elements might want to have that information.

2016-06-23 15:17:36 +0200  Edward Hervey <edward@centricular.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Let upstream events go through upstream
	  There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
	  Some elements might want to have that information.
	  Also remove downstream-only CAPS event handling and minimize code

2016-07-07 23:53:54 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* sys/v4l2/gstv4l2.c:
	  v4l2: fix v4l2 probe build error
	  A typo in gst_v4l2_probe_and_register() caused a build error when building
	  with --enable-v4l2-probe. Fixing it.
	  gstv4l2.c: In function 'gst_v4l2_probe_and_register':
	  gstv4l2.c:150:25: error: 'struct v4l2_capability' has no member named 'capabilitites'
	  device_caps = vcap.capabilitites;

2016-07-01 22:53:33 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: use gst_caps_intersect_full in negotiate()
	  Instead of reimplementing the GST_CAPS_INTERSECT_FIRST
	  interection mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768195

2016-07-02 01:56:07 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: use opened device caps instead of physical device ones
	  The same physical device can export multiple devices. In
	  this case, the capabilities field now contains a union of
	  all caps available from all exported V4L2 devices alongside
	  a V4L2_CAP_DEVICE_CAPS flag that should be used to decide
	  what capabilities to consider. In our case, we need the
	  ones from the exported device we are using.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768195

2016-07-07 18:24:59 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Remove suspicious checks for pads being active and linked
	  We should add all pads, no matter if they are linked or active or not at this
	  point. Skipping some that are not will cause different behaviour than with
	  other muxers.

2016-07-07 18:23:07 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Error out if we start writing data with some pads not having a codec id yet
	  This can only happen if a) upstream somehow gets around the CAPS event failing
	  or b) there never being any CAPS event.
	  The following code assumes that all pads have a codec-id.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768509

2016-07-07 18:14:43 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Consistently use gst_matroska_mux_set_codec_id() for setting the codec id

2016-07-04 09:50:11 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtph265pay/depay: Sync against RFC 7798
	  Handle sprop-vps, sprop-sps and sprop-pps in caps instead of
	  sprop-parameter-sets.
	  rtph265pay works with byte-stream and hvc1 formats but not hev1 yet. It
	  handles profile-id, tier-flag and level-id in caps query.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753760

2016-07-06 09:25:00 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Push nominal bitrate tags
	  Add per-stream tag lists, which are used to send nominal
	  bitrate tags. When remuxing FLV => FLV, this now passes
	  through the upstream bitrate.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768440

2016-07-06 09:24:49 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Refactor metadata tag handling
	  The FLV header cannot be trusted to indicate video or
	  audio presence, as the comments already mention. Don't
	  delay pushing tags waiting for streams that might never
	  appear.
	  Tags are now pushed immediately after they change:
	  - After parsing an onMetaData script object
	  - After negotiating caps on a pad
	  https://bugzilla.gnome.org/show_bug.cgi?id=768440

2016-07-06 12:44:10 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix AAC codec_data values
	  As seen in the parent switch for object_type_id, the 4 possible values are
	  0x40, 0x66, 0x67 and 0x68. Fixing the nested switch to match these values.
	  Looks like it was a typo making them decimal instead of hexadecimal.
	  CID 1363328

2016-07-06 13:51:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.9.1 ===

2016-07-06 13:06:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.9.1

2016-07-06 11:46:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2016-07-06 11:22:53 +0300  Steven Hoving <sh@bigbrother.nl>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix error messages to first convert to doubles before division

2016-07-06 10:18:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/da.po:
	* po/hr.po:
	* po/pt_BR.po:
	* po/sk.po:
	  po: Update translations

2016-07-05 21:11:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Set to PLAYING after a seek again after setting up the segment and everything else
	  There's a small window for a race condition otherwise.

2016-07-04 17:45:40 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/qtmux.c:
	  qtmux: Use complete AAC caps with codec_data in the tests

2016-07-04 16:58:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Reject raw AAC if no codec_data is found in the caps
	  If necessary, a demuxer will have to invent something here but this is only a
	  problem with non-conformant files anyway.

2016-07-04 16:55:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Invent AAC codec_data if none is present
	  Without, raw AAC can't be handled and we have some information available in
	  the decoder that most likely allows us to decode the stream in one way or
	  another. This is the same code already used by matroskademux for the same
	  reasons, and ffmpeg/vlc play such files just fine too by guesswork.

2016-07-04 14:54:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Reject raw AAC caps without codec_data
	  The resulting file is not going to be playable without guesswork and raw caps
	  should always have codec_data.

2016-07-01 19:22:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/qt/Makefile.am:
	  qt: fix build some more when QPA is not available
	  Compiler would complain about include directory that didn't
	  exist because QPA_INCLUDE_PATH gets subst-ed regardless
	  (and if it didn't we'd have just an empty -I argument).
	  https://bugzilla.gnome.org/show_bug.cgi?id=767553

2016-05-10 15:48:49 +0200  Edward Hervey <edward@centricular.com>

	  qtdemux: Handle upstream GAP in push-mode/time segment
	  This is to handle cases where upstream handles the fragmented streaming in TIME
	  segments and sends us data with gaps within fragments. This would happen when dealing
	  with trick-modes.
	  When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
	  it must obey the following rules:
	  * The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
	  * The buffers containing the first sample after a gap:
	  * MUST start at the beginning of a sample,
	  * MUST have the DISCONT flag set,
	  * MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767354

2016-07-01 11:54:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/v4l2-utils.c:
	  v4l2: fix potential double-free of error debug string
	  gst_v4l2_clear_error() doesn't work like g_clear_error(), it
	  doesn't NULLify the pointer, so set freed debug string to NULL
	  so it doesn't get freed again if gst_v4l2_clear_error() is
	  called twice on the error.
	  CID 1362901

2016-07-01 10:05:00 +0000  Brad Lackey <blackey@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't disable UDP protocols on redirecting
	  https://bugzilla.gnome.org/show_bug.cgi?id=768232

2016-07-01 17:28:17 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Push caps only when it was updated
	  Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 caused new caps
	  event per moof without consideration of duplication.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768268

2016-06-30 15:01:46 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: fix invalid memory access
	  10 bytes was allocated for stream_format but size of "byte-stream" is
	  more. Use g_strdup() instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753760

2016-06-29 23:31:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/shout2/gstshout2.c:
	  shout2: Use a non-timer GstPoll
	  Otherwise set_flushing() will have undefined semantics and nowadays causes a
	  g_critical() to warn about that.

2016-06-19 02:08:25 -0300  Thiago Santos <thiagossantos@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: dynamically adjust blocksize
	  Update the blocksize depending on how much is obtained from a read
	  of the input stream. This avoids doing too many reads in small chunks
	  when larger amounts of data are available and also prevents using
	  a very large memory area to read a small chunk of data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767833

2016-06-28 16:44:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Windows has no ipi_spec_dst in struct in_pktinfo

2016-06-28 15:15:14 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: #define __APPLE_USE_RFC_3542 to be able to use IPV6_PKTINFO on OSX/iOS

2016-06-28 15:08:04 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Move #includes around to a) work around broken glibc header and b) Windows

2016-06-28 14:25:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix compilation on Windows and *BSD/OSX

2016-06-23 20:21:59 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Filter out multicast packets that are not for our multicast address
	  https://bugzilla.gnome.org/show_bug.cgi?id=767980

2016-06-28 10:57:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state
	  If we consider the RTSP state, what can happen is that it is PLAYING but the
	  element already asynchronously tried to PAUSE and it just did not happen yet.
	  We would then override this setting to PAUSED (while the element actually is
	  in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
	  to produce packets while the sinks are all PAUSED, piling up thousands of
	  packets in the rtpjitterbuffer and other elements and finally failing.

2016-06-27 18:15:08 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/qt/qtitem.cc:
	  qmlglsink: Fix build error when don't have QPA installed.
	  Check header file existance and wrap the header file include
	  in the necessary #ifdef to avoid build error.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767553

2016-06-27 09:20:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Add comment about H263/MPEG4P2 being non-standard for FLV
	  They are however supported by ffmpeg and apparently used out there.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768006

2016-06-24 14:48:53 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Add support for H263 and MPEG4 part2
	  https://bugzilla.gnome.org/show_bug.cgi?id=768006

2016-06-16 15:13:02 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	* ext/qt/qtplugin.pro:
	  qmlglsink: add win32 support
	  The current state of c++ ABI's on Window's and Gst's/Qt's conflicting
	  mingw builds means that we cannot use mingw for building the qt plugin.
	  Instead, a qmake .pro file is provided that is expected to be used with the
	  msvc binaries provided by Qt like so:
	  (with the PATH environment variable containing the path to the qt biniaries
	  and PKG_CONFIG_PATH containing the path to GStreamer modules)
	  cd /path/to/sources/gst-plugins-bad/ext/qt
	  qmake -tp vc
	  Then open the resulting VS project and build the library.  Then
	  cp debug/libgstqtsink.dll /path/to/prefix/lib/gstreamer-1.0/libgstqtsink.cll
	  https://bugzilla.gnome.org/show_bug.cgi?id=761260

2016-06-21 17:10:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	  Update plugins doc
	  This is partly automated using "make update" in docs/plugins, but also
	  required manual merge. Additionally, missing plugins and elements have
	  been added.

2016-06-21 17:51:38 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/splitmux.c:
	  tests: splitmux: skip tests if theora or ogg plugins are not available
	  https://bugzilla.gnome.org/show_bug.cgi?id=767861

2016-06-21 11:46:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* common:
	  Automatic update of common submodule
	  From ac2f647 to f363b32

2016-06-21 07:40:42 -0400  Aaron Boxer <boxerab@gmail.com>

	* gst/rtp/gstrtpj2kpay.c:
	  gstrtpj2kpay: use tile bit and tile number to determine if there are multiple tiles in packet
	  Now we don't have to rely on a special value for the tile number.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767817

2016-06-21 09:34:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: fix compiler warning on OS/X
	  gstrtpj2kpay.c:364:21: error: implicit truncation from 'int' to bitfield changes value from -1 to 65535
	  https://bugzilla.gnome.org/show_bug.cgi?id=767817

2016-06-21 09:34:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	  docs: update

2016-05-16 17:31:58 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/capssetter.c:
	* tests/check/elements/icydemux.c:
	* tests/check/elements/jpegenc.c:
	* tests/check/elements/level.c:
	* tests/check/elements/multifile.c:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rtprtx.c:
	* tests/check/elements/udpsrc.c:
	  fix buffer leaks in tests
	  Need to call gst_check_drop_buffers() to release the buffers exchanged
	  during the test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766561

2016-05-17 12:52:43 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/interleave.c:
	  interleave: fix message leaks in test
	  Flush the bus when cleaning up so pending messages are destroyed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766561

2016-05-17 12:58:06 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/videomixer.c:
	  videomixer: fix event leaks in test
	  https://bugzilla.gnome.org/show_bug.cgi?id=766561

2016-05-13 15:12:22 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/deinterleave.c:
	  deinterleave: fix leaks
	  - Flush the bus so messages aren't leaked
	  - Fix pad leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=766561

2016-06-17 15:29:16 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Deprecated sprop-parameter-set property
	  This is supposed to be either in the codec_data (avc stream format) or inside
	  the stream, and we extract it from there. It should not be set from a
	  property as it's stream specific.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767789

2016-06-17 12:16:32 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: make all srtp encoder properties explicit
	  The Session Data Protocol doesn't allow specifying a cipher for the
	  SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher
	  "aes-128-icm" is the default for SRTP and SRTCP, but if we want to have
	  an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP
	  cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767799

2016-06-17 19:59:13 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/soup/gstsoup.c:
	  soup: work around frequent deadlocks in GLib type initialisation
	  .. by registering the types from the plugin init function. This
	  seems to help, but we'll see if it's enough (might need similar
	  things elsewhere).
	  https://bugzilla.gnome.org/show_bug.cgi?id=693911
	  https://bugzilla.gnome.org/show_bug.cgi?id=674885

2016-06-17 16:08:08 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: The prores variant is stored in the variant field, not format
	  And the caps in the sink pad template already used variant (only).

2016-06-17 13:00:48 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	  rtph265pay: Remove sprop-parameter-sets property
	  There is no valid use case when this property is needed since the values
	  must be in either codec_data or buffer data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753760

2016-06-10 16:17:26 +0200  Jonas Holmberg <jonashg@axis.com>

	* docs/plugins/scanobj-build.stamp:
	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Read NALU type the same way everywhere
	  Cosmetic change to read NALU type in gst_rtp_h265_pay_decode_nal() the
	  same way as in other places.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753760

2016-06-17 13:58:33 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: fix RTPJitterBufferMode documentation
	  Documentation lacks '@' before each enum values and there was an extra
	  line after symbol section which confuses GTK-Doc parser.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767788

2016-05-23 10:18:48 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: take the lock when changing stats
	  https://bugzilla.gnome.org/show_bug.cgi?id=766025

2016-04-14 18:14:32 +0300  Sergey Borovkov <sergey.borovkov@wireload.net>

	* ext/qt/qtitem.cc:
	  qml: Enable qmlglsink for eglfs
	  https://bugzilla.gnome.org/show_bug.cgi?id=763044

2016-06-16 00:44:48 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  qmlglsink: propagate GL context creation failure upwards
	  Otherwise an application cannot know if the qmlglsink will be displaying frames
	  incorrectly/at all.

2016-06-16 00:44:16 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  qmlglsink: also allow wayland-egl as a platform name

2016-06-12 15:35:28 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/qt/Makefile.am:
	* ext/qt/qtitem.cc:
	  qmlglsink: Add Wayland support
	  Don't use gstgldisplay to get wayland display. Should use QPA on wayland
	  to get wayland display for QT.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767553

2016-06-15 11:19:43 +0200  Jürgen Slowack <jurgen.slowack@barco.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265: fix NAL unit type parsing and SPS/PPS/VPS detection
	  Fixes sps/pps/vps insertion via the config-interval property.
	  https://bugzilla.gnome.org//show_bug.cgi?id=767680

2016-06-11 12:16:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/pipelines/simple-launch-lines.c:
	  simple-launch-lines: Use correct JPEG2000 caps

2016-06-10 13:43:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix indentation

2016-06-10 13:42:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix date parsing when there are trailing spaces
	  Fixes parsing of "Thu May 11 15:57:46 2006 ".
	  https://bugzilla.gnome.org/show_bug.cgi?id=767496

2016-05-13 15:08:24 -0400  Aaron Boxer <boxerab@gmail.com>

	* gst/rtp/gstrtpj2kcommon.h:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	  gstrtpj2k: set sampling field required by RFC
	  This field is now required in the sink caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766236

2016-06-09 09:30:48 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Fix unref assertion failure
	  Fix unref assertion failure
	  https://bugzilla.gnome.org/show_bug.cgi?id=767424

2016-05-14 14:46:17 +0200  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Work with non-TIME segments
	  With non-time segments, it now assumes that the arrival time of packets
	  is not relevant and that only the RTP timestamp matter and it produces
	  an output segment start at running time 0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766438

2016-06-07 20:53:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/libpng/gstpngdec.c:
	  pngdec: Wait for segment event before checking it
	  The heuristic to choose between packetise or not was changed to use the
	  segment format. The problem is that this change is reading the segment
	  during the caps event handling. The segment event will only be sent
	  after. That prevented the decoder to go in packetize mode, and avoid
	  useless parsing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736252

2016-06-06 17:00:22 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Wait for segment event before checking it
	  The heuristic to choose between packetise or not was change to use the
	  segment format. The problem is that this change is reading the segment
	  during the caps event handling. The segment event will only be sent
	  after. That prevented the decoder to go in packetize mode, and avoid
	  useless parsing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736252

2016-06-07 16:42:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Keep part of the input buffer
	  Instead of completely getting rid of the input buffer, copy
	  the metadata, the flags and the timestamp into an empty buffer.
	  This way the decoder base class can copy that information again
	  to the output buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758424

2016-06-07 16:41:58 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Coding style fixes

2016-06-07 16:09:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Coding style fixes

2016-06-07 16:04:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Add an error return to _try/_set_format
	  This way one can easily ignore errors. Previously, error were always
	  posted ont he bus.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766172

2016-06-07 16:01:55 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/v4l2-utils.c:
	* sys/v4l2/v4l2-utils.h:
	  v4l2-util: Introduce GstV4l2Error
	  This is to allow returning an error that can easily be sent as
	  message to the application if the element needs it. Using this
	  also allow ignoring errors.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766172

2016-06-07 12:41:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Avoid decide allocation on active pool
	  v4l2src will renegotiate only if the format have changed. As of now,
	  it's not possible to change the allocationw without resetting the
	  camera. To avoid unwanted side effect, simply keep the old allocation
	  if no renegotiation is taking place. This fixes assertion and possible
	  failures in USERPTR or DMABUF import mode (when using downstream pools).
	  https://bugzilla.gnome.org/show_bug.cgi?id=754042

2016-04-28 13:44:49 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Show state name in debugging
	  Makes it easier to trace what's going on

2016-05-10 15:45:42 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Remove useless variable
	  That variable is only needed for a debug statement, move it there

2016-05-10 15:10:36 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Add/Fix comments on the various structure variables
	  No variables were added/removed. This was just a good excuse to:
	  * Comment what most variables are used for (and when)
	  * Order them in such a way as to show first the common variables used
	  in all cases, followed by those only used in push-mode

2016-05-10 15:07:40 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Remove unused structure
	  Let's just remove it, been commented for 7+ years :)

2015-09-02 11:48:29 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: use decoder stop command instead of queueing empty buffers
	  Only if the decoder stop command fails, keep queueing empty buffers to
	  signal end of stream as before.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733864

2014-12-12 14:31:36 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: add gst_v4l2_decoder_cmd helper
	  https://bugzilla.gnome.org/show_bug.cgi?id=733864

2016-06-01 20:28:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Forward segments directly if we are operating in PUSH mode on fragmented streams
	  We shouldn't go through segment activation as we will only have a limited
	  understanding of how the whole stream timeline looks like from the moof. We
	  only know about the current fragment, while upstream knows about the whole
	  stream.
	  This fixes seeking in DASH streams, both for seeks after the current moof and
	  for seeks into the current moof. The former would fail because the moof ends
	  and we can't activate any segment, the latter would cause a segment that stops
	  at the moof end, and no further fragments would be played because we end up
	  being EOS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767071

2016-06-06 17:54:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Use looser caps for upstream
	  When we fixate for upstream, try to not introduce new fields when not
	  needed. This was imported from videoconvert element.

2015-01-28 12:07:58 +0100  Enrico Jorns <ejo@pengutronix.de>

	* sys/v4l2/gstv4l2transform.c:
	  gstv4l2transform: format fixation for preferring passthrough
	  * If outgoing format is unfixated, try to set it to input format.
	  * Call gst_caps_fixate () at end of fixation routine
	  https://bugzilla.gnome.org/show_bug.cgi?id=766719

2016-05-20 12:49:53 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: allow to change pixel aspect ratio
	  Scalers may change width and height independently,
	  allow to change pixel aspect ratio.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766712

2016-05-20 12:32:25 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: fix scaling in case of fixed pixel aspect ratio
	  To change pixel aspect ratio from DAR to PAR, the necessary scaling factor
	  is DAR/PAR, not DAR*PAR.
	  For good measure, add debug output similar to the fixed-width and
	  fixed-height cases.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766711

2016-05-13 16:39:25 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: fill colorimetry in gst_v4l2_object_acquire_format
	  Instead of relying on the default colorimetry chosen by
	  gst_video_info_set_format(), set info.colorimetry from the
	  values returned by G_FMT. This allows decoders to propagate
	  their input colorimetry downstream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766383

2016-05-18 10:17:12 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: refactor gst_v4l2_object_get_colorspace to take a v4l2_format parameter
	  Move the extraction of colorimetry parameters from struct v4l2_format and the
	  setting of the identity matrix for RGB formats into the function to avoid code
	  duplication.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766383

2016-05-13 14:58:41 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: use visible size, not coded size, for downstream negotiation filter
	  gst_v4l2_probe_caps() returns the coded size, not the visible size. Subtract
	  the known padding from probed caps with the coded size before using them as
	  filter for caps negotiation with downstream elements.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766382

2016-05-13 14:45:02 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: use G_SELECTION instead of G_CROP in gst_v4l2_object_acquire_format
	  The gst_v4l2_object_acquire_format() function is used by v4l2videodec to obtain
	  the currently set capture format. Since G_FMT returns the coded size, the
	  visible size needs to be obtained from the compose rectangle in order to
	  negotiate it with downstream elements. The G_CROP call hasn't worked on mem2mem
	  capture queues for a long time. Instead use the G_SELECTION call to obtain the
	  compose rectangle and only fall back to G_CROP for ancient kernels.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766381

2016-01-27 09:57:38 +0100  Andreas Naumann <anaumann@ultratronik.de>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Use V4L2_BUF_TYPE_VIDEO_OUTPUT_OVERLAY if driver advertises it.
	  On modern kernels, the G/S_FMT ioctls will always fail using
	  V4L2_BUF_TYPE_VIDEO_OVERLAY with VFL_DIR_TX (e.g. real overlay out drivers)
	  since this is not the intented use (rather rx, according to v4l2 API doc).
	  Probably this is why the Video Output Overlay interface was created, so if
	  the driver advertises it we might as well use.
	  For old kernels (pre 2012) the old way might still work so keeping this for
	  compatibility.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761165

2016-06-06 18:52:01 +0100  Kieran Bingham <kieran@bingham.xyz>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Use non-deprecated V4L2 type for RGB15
	  Support for the updated V4L2_PIX_FMT_XRGB555 was added in commit
	  2538fee2fd8fdb74b05f0a511281bc4707e7cc44 however, when setting the format
	  for use in v4l2 ioctls, the old deprecated format is still used. Convert
	  this to the new accepted format type, as the preferred format.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767300

2016-05-04 14:50:32 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/matroska/matroska-demux.c:
	  matroskademux: preserve seek flags
	  Without this some flags get lost in streaming mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767194

2016-06-06 10:47:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/Makefile.am:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  Revert "WIP revert soup"
	  This reverts commit fdac3a7a231f3848665636cf8122f96103b46e3b.
	  Was not supposed to be pushed but a local workaround for
	  https://bugzilla.gnome.org/show_bug.cgi?id=693911#c13

2016-06-03 13:09:35 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: complete warn log with SSRC
	  https://bugzilla.gnome.org/show_bug.cgi?id=767195

2016-05-31 15:29:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/Makefile.am:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  WIP revert soup

2016-06-03 13:18:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Unref seek event in any case
	  It would be leaked if no seek handler was currently set.

2016-06-03 10:49:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: Properly set event/message sequence numbers based on the previous seek
	  See https://bugzilla.gnome.org/show_bug.cgi?id=765935
	  https://bugzilla.gnome.org/show_bug.cgi?id=767157

2016-06-03 10:36:32 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: Remember if upstream had a time segment and if not properly create time segments
	  Previously the segment.time was wrong, and the position was not updated
	  correctly, resulting in seeks in PUSH mode with upstream providing a BYTES
	  segment to not work at all.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767157

2016-06-03 09:54:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Implement SEEKING query so we can actually seek if upstream can't seek in TIME
	  https://bugzilla.gnome.org/show_bug.cgi?id=767157

2016-06-02 14:19:15 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Recalculate the frame offsets at the beginning of each BYTE segment and whenever upstream gives us a timestamp
	  This fixes seeking in DV streams where upstream operates in PUSH mode with a
	  TIME segment (e.g. avidemux). Without this, we would generate wrong durations
	  and timestamps after a seek.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767157

2016-06-02 13:53:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: Pass-through buffer DISCONT flags
	  https://bugzilla.gnome.org/show_bug.cgi?id=767157

2016-06-02 16:16:45 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpvp9depay.c:
	  rtpvp9depay: Don't assert on flexible mode packets
	  Instead just post a warning on the bus for now.

2016-06-02 15:03:17 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  tests: rtpbin: fix caps leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=767156

2016-06-02 15:00:01 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* tests/check/elements/amrparse.c:
	  tests: amrparse: clean up test
	  - use GST_CHECK_MAIN() to reduce boilerplate
	  - unref the input caps using a teardown function to prevent leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=767156

2016-05-20 15:22:35 +0200  Edward Hervey <edward@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Ensure DISCONT flag is properly propagated
	  The output of deinterlace at startup, or when receiving a new DISCONT
	  buffer, should have the DISCONT flag set on the first buffer.

2016-05-31 21:34:04 +0200  Josep Torra <adn770@gmail.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2src: check for valid size on raw video buffers
	  Discard buffers that doesn't contain enough data when dealing
	  with raw video inputs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767086

2016-05-31 17:10:36 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use the demuxer segment instead of a new one for MSS streams
	  Upstream might have told us something about the to be expected segment, so
	  let's use that information instead of coming up with a [0,-1] segment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767071

2016-05-31 17:04:32 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Only activate segments and send SEGMENT events if we have streams
	  But in that case also remove the pending newsegment event, otherwise we would
	  later send a possibly outdated event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767071

2016-05-31 16:53:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: In PULL mode, nothing is ever going to send us a SEGMENT event
	  https://bugzilla.gnome.org/show_bug.cgi?id=767071

2016-05-31 16:38:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't override TIME segments from upstream that we just saw
	  The point of d8fb7a9c96b108814beeaa0e63f818d4648c7fe9 was to not have any
	  spurious segments stored for later if we do BYTES->TIME conversion, but
	  overriding any TIME segments from upstream does not make any sense.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=763165
	  https://bugzilla.gnome.org/show_bug.cgi?id=767071

2015-07-16 09:48:46 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: set position as offset from start-index
	  query position in GST_FORMAT_BUFFER returns
	  offset from start-index rather than index.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752462

2016-05-27 12:49:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/pipelines/simple-launch-lines.c:
	* tests/files/Makefile.am:
	* tests/files/gradient.j2k:
	  tests: add unit test for JPEG-2000 rtp payloader leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=766870

2016-05-25 17:11:13 +0200  Pierre Lamot <pierre.lamot@openwide.fr>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: Fix buffer memory leak
	  Input buffer memory was not unmapped
	  https://bugzilla.gnome.org/show_bug.cgi?id=766870

2016-05-18 12:12:15 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: fix caps leak
	  gst_v4l2_object_probe_caps() was taking an extra ref on the returned
	  caps for no reason.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766610

2016-05-22 20:14:18 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/videocrop/gstvideocrop.c:
	  videocrop mark crop properties as mutable in playing state

2016-05-20 16:47:35 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix buffer leak when flushing
	  When early returning in gst_soup_http_src_read_buffer() because the
	  element is FLUSHING, we need to unmap and unref the buffer which was just created.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766718

2016-05-20 11:15:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Set seek event seqnum on all SEGMENT events
	  Some were forgotten.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=765935

2016-05-20 11:12:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Pass through seek event seqnums in all SEGMENT/EOS events and SEGMENT_DONE messages/events
	  See https://bugzilla.gnome.org/show_bug.cgi?id=765935

2016-05-20 10:56:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Set seek event seqnum in EOS and SEGMENT_DONE messages/events
	  Also actually store the seqnum in pull mode seeks.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=765935

2016-05-17 13:40:38 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix caps leak
	  The caps returned by gst_pad_get_current_caps() was never unreffed when
	  not early returning.
	  Fix a leak with the elements/deinterlace test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766558

2016-01-25 16:25:51 +0100  Mikhail Fludkov <misha@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpsession.c:
	  rtpsession: don't act on suspicious BYE RTCP
	  Some endpoints (like Tandberg E20) can send BYE packet containing our
	  internal SSRC. I this case we would detect SSRC collision and get rid
	  of the source at some point. But because we are still sending packets
	  with that SSRC the source will be recreated immediately.
	  This brand new internal source will not have some variables incorrectly
	  set in its state. For example 'seqnum-base` and `clock-rate` values will be
	  -1.
	  The fix is not to act on BYE RTCP if it contains internal or unknown
	  SSRC.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762219

2015-11-15 14:54:28 +0100  Mikhail Fludkov <misha@pexip.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: Add test for locking of the stats signal
	  Keeping the lock while emitting the stats signal introduces potential
	  deadlock in those situations when the signal callback wants the access
	  to rtpsession's properties which also requre the lock.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762216

2016-05-19 15:36:57 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: don't hold object lock whilst pushing out headers
	  matroskademux would take the GST_OBJECT_LOCK in
	  - gst_matroska_demux_push_codec_data_all()
	  - gst_matroska_demux_query()
	  Some parse element such as FLAC checks upstream seekability, and
	  there is some use cases that matroska-demux is linked to a parse element
	  (e.g.,FLAC format) without intermediate elements (e.g., queue).
	  In this case, matroska-demux never returns from _push_codec_data_all()
	  because the parser can return only after it receives the response to
	  the upstream query, but that's not going to happen because it's
	  deadlocked.
	  Elements must not hold the object lock whilst pushing out events
	  or data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766645

2016-05-19 12:43:01 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Set sent_buffers and streamheader_buffers to NULL after freeing
	  Otherwise we might use an already freed list later and crash or worse.

2016-05-18 18:32:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix Since version for new "loop" property

2016-05-16 16:18:37 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/rtsp/gstrtpdec.c:
	  rtpdec: fix clock leak
	  gst_system_clock_obtain() returns a new ref.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766521

2016-05-17 05:33:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: add doc blurb with since marker for new "loop" property

2015-11-13 15:52:35 +0100  Dimitrios Katsaros <patcherwork@gmail.com>

	* gst/avi/gstavimux.c:
	  avimux: add support for png
	  https://bugzilla.gnome.org/show_bug.cgi?id=758059

2016-05-15 22:07:14 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxsrc: Connect to demux signals before activating
	  Fix a race in splitmuxsrc by properly connecting to the
	  demuxer signals we're interested in *before* setting it running.

2016-05-15 13:31:37 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: Update for git master

2016-05-15 12:16:23 +0200  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4gpay.h:
	  rtpmp4gpay: Don't produce timestamps based on byte count
	  The GST_BUFFER_OFFSET of output buffers returned to GstRtpBasePayload
	  should reflect the number of "samples" in the unit of the RTP clock in this
	  buffer. If this is not true, then it shouldn't be set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761943

2016-05-15 12:24:03 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Fix strcmp usage
	  Just use g_strcmp0 which can handle NULL entries

2016-03-04 10:14:00 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Use audio/x-unaligned-raw instead of audio/x-raw for L16 data
	  Directly setting audio/x-raw caps leads to problems when the delivered
	  data blocks do not align properly at sample boundaries (for example, a
	  data block with 391 bytes). So, instead, set audio/x-unaligned-raw to
	  let a parser be autoplugged.
	  https://bugzilla.gnome.org/show_bug.cgi?id=689460

2016-05-12 11:52:09 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Parsing elst box based on version
	  segment_duration and media_time should be parsed based on version
	  of elst box. Specification defines that an elst box with version 1
	  has uint64 and int64 values for segment_duration and media_time,
	  respectively.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766301

2016-05-14 12:57:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: check if request was cancelled when sending message
	  It might be that the request was aborted by the application and
	  we can return immediatelly

2016-05-14 12:43:54 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: proxy resolver is on by default
	  Remove from the session creation parameters

2016-05-14 12:15:48 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/soup/Makefile.am:
	  soup: update build to warn about newer deprecated functions
	  We already depend on 2.48

2016-05-14 11:09:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: reduce reading latency by using non-blocking read
	  Non-blocking read will return the amount of data available without
	  blocking to wait for the full requested size.
	  The downside is that now it souphttpsrc needs to have a waiting
	  mechanism in case there is no data available yet to avoid busy
	  looping arond the inputstream.

2016-05-15 12:30:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Take the lock already when reading the other stats, not just for the hash table
	  https://bugzilla.gnome.org/show_bug.cgi?id=766025

2016-05-14 17:04:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/ebml-read.c:
	  matroska: use math-compat.h for NAN define

2016-05-14 23:39:22 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Use GstBin async-handling instead of our own.
	  Set the async-handling property on GstBin to let it manage
	  async-handling instead of the local handling from the previous
	  commit. Works because of #174a5e in core

2016-05-13 10:17:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: refactor to use Soup's sync API
	  Replace the async API with the sync API to remove all the extra mainloop
	  and context handling. Currently it blocks reading until 'blocksize'
	  bytes are available but that can be improved by using:
	  https://developer.gnome.org/gio/unstable/GPollableInputStream.html#g-pollable-input-stream-read-nonblocking
	  https://bugzilla.gnome.org/show_bug.cgi?id=693911

2016-05-14 04:50:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: replace deprecated API
	  Avoid using soup_server_run_async and old get_port() APIs,
	  replace with me soup_server_listen and get the port through the
	  URIs list returned from the server.

2016-05-14 12:34:10 +0200  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Upgrade debug message to error
	  It causes the entire pipeline to fail, it should be easier to find.

2016-05-14 18:32:52 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Hide internal async state changes.
	  When switching fragments, hide the async-start/async-done
	  messages from the parent bin, as otherwise we sometimes (very rarely)
	  hang in PAUSED instead of returning / continuing to PLAYING
	  state.

2016-05-13 21:20:28 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Remove stray carriage-return from debug

2016-05-13 16:43:21 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/Makefile.am:
	  rtp: Ship gstrtpj2kcommon.h file to fix distcheck

2015-04-30 14:43:04 +0200  Jesper Larsen <knorr.jesper@gmail.com>

	* gst/avi/gstavimux.c:
	  avimux: Do not write index and header if idx is NULL
	  Fixes criticals with e.g.
	  videotestsrc num-buffers=1 ! identity drop-probability=1.0 ! avimux ! fakesink
	  https://bugzilla.gnome.org/show_bug.cgi?id=748700

2016-05-12 08:43:39 -0400  Aaron Boxer <boxerab@gmail.com>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: manage T tile invalidation bit correctly, update tile id in header correctly.
	  1. according to RFC, T bit is only set when either the RTP packet only contains the J2K main header, or the packet contains tile parts from multiple tiles. This is now being managed correctly in the code. The second scenario cannot happen with our payloader, since tile headers are always placed in their own RTP packet, and so a packet cannot contain tile parts from multiple tiles.
	  However, I have added code to track if multiple tile parts are included in a single RTP packet, in case in the future we want to put header and data in same packet.
	  2. Old code would set the tile id to zero for all J2K packets. This is now set correctly to the appropriate tile id.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745187

2016-05-12 08:41:51 -0400  Aaron Boxer <boxerab@gmail.com>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: manage fragmented headers correctly
	  J2K main header framentation across multiple RTP packets is now handled correctly
	  https://bugzilla.gnome.org/show_bug.cgi?id=745187

2016-05-11 15:04:26 -0400  Aaron Boxer <boxerab@gmail.com>

	* gst/rtp/gstrtpj2kcommon.h:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kdepay.h:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpj2kpay.h:
	  rtpj2k: move common code to shared header, code clean up
	  https://bugzilla.gnome.org/show_bug.cgi?id=745187

2016-05-11 15:01:32 -0400  Aaron Boxer <boxerab@gmail.com>

	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2k: update documentation
	  https://bugzilla.gnome.org/show_bug.cgi?id=745187

2016-05-12 14:43:43 +0200  Patricia Muscalu <patricia@axis.com>

	* gst/auparse/gstauparse.c:
	* gst/auparse/gstauparse.h:
	  auparse: Fix sticky event misordering warning
	  Make sure that src pad has caps before sending segment event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766359

2016-05-11 09:28:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Don't notify about stats property changes while taking the session lock
	  The signal handlers might want to actually get the value of the stats
	  property, which would take the session lock again and deadlock.
	  This was introduced by 2e960e70750a0cb7e1117d0c09d08597866a29ee.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766025

2016-05-03 13:59:54 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: improve edts segment handling after seeks in push mode
	  Properly handle edts segments for push-based operation seeking.
	  We only support edts that a single segment that has media at the end,
	  being preceeded by any number of gap segments.
	  This also allows the qt segment rate to be respected after seeks
	  https://bugzilla.gnome.org/show_bug.cgi?id=765669

2016-05-03 10:41:06 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: properly activate segment with rate != 1.0
	  Also use the qt rate to identify the position within a qt segment
	  to properly translate playback time to qt media time
	  https://bugzilla.gnome.org/show_bug.cgi?id=765669

2016-05-03 11:45:01 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix stall when receiving already lost packet
	  When a packet arrives that has already been considered lost as part of a
	  large gap the "lost timer" for this will be cancelled. If the remaining
	  packets of this large gap never arrives, there will be missing entries
	  in the queue and the loop function will keep waiting for these packets
	  to arrive and never push another packet, effectively stalling the
	  pipeline.
	  The proposed fix conciders parts of a large gap definitely lost (since
	  they are calculated from latency) and ignores the late arrivals.
	  In practice the issue is rare since large gaps are scheduled immediately,
	  and for the stall to happen the late arrival needs to be processed
	  before this times out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765933

2016-05-05 14:18:21 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Take session lock when creating stats
	  The access to the session hash table must happen while the session lock is
	  taken, otherwise another thread might modify the hash table while we're
	  creating the stats.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766025

2016-05-03 21:17:01 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: update segment when new duration is found
	  Otherwise the old segment will have a shorter stop time and would
	  cause the stream to end too early.

2016-05-04 11:37:29 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: dismember activate_segment into 2 parts
	  One that updates and push a new segment, the other will move the
	  stream to the new segment starting position

2016-05-04 09:30:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	  dv: Use correct pixel-aspect-ratio values
	  The previous ones resulted in odd display aspect ratios and were different
	  from the ones used by e.g. ffmpeg. The new ones now result in display aspect
	  ratios of 4:3 and 16:9.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765946

2015-11-09 17:55:09 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/splitmux.c:
	  tests: add splitmuxsrc test for new "format-location" signal
	  https://bugzilla.gnome.org/show_bug.cgi?id=753625

2015-11-09 17:51:12 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: add a format-location signal that allows bypassing the location property
	  This signal allows a user to directly return a sorted list of
	  files to be joined, so that they don't have to follow the
	  filename pattern that the "location" property expects.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753625

2016-05-04 11:15:20 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix deadlock case when source reaches EOS
	  https://bugzilla.gnome.org/show_bug.cgi?id=765072

2016-05-03 22:59:27 -0700  Stefan Sauer <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  wavparse: simplify and correct header scanning
	  The wav spec tells that 'fmt' (and 'bext' if present) must come before 'data'.
	  There is no requirement for 'fmt' to be first. We already had a list of chunks
	  to skip, but it is easier to just skip any chunk while seeking for 'fmt'.
	  This fixes reading files generated by ProTools.

2016-04-30 22:15:13 +0900  Hyunjun Ko <zzoon@igalia.com>

	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxaudio/gstosxaudiodeviceprovider.c:
	* sys/osxaudio/gstosxaudiodeviceprovider.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxaudiosrc.h:
	  osxaudio: Support audio device provider on osx
	  https://bugzilla.gnome.org/show_bug.cgi?id=753265

2016-05-01 15:09:27 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavimux.c:
	  avimux: set audio header rate according to calculated bps in stop_file
	  ... now that set_fields is no longer called there by
	  e538608b3f90539003de21c1db238f3c9b946e30

2016-04-29 15:04:11 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Store the segment sequence number in the EOS events and SEGMENT_DONE events/message
	  Also instead of storing it per stream, store it globally in the demuxer. It's
	  the same for each stream anyway.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765806

2016-04-11 10:54:38 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Always bind to ANY when address is a multicast address and not only on Windows
	  For IPv6 addresses, binding to a multicast group does not work on Linux
	  either. Always bind to ANY and then later join the multicast group.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764679

2016-04-26 17:01:49 +0800  Song Bing <b06498@freescale.com>

	* sys/ximage/ximageutil.c:
	  ximageutil: shouldn't implement transform if don't support it
	  shouldn't implement transform if don't support it. Or gst_buffer_copy_into()
	  will print ERROR log.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765583

2016-04-28 16:24:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Allow MPEG-1 Layer 1 and 2 in addition to 3 in MP4
	  Via the MPEG-4 Part 3 spec we can support the other layers too.
	  Also correct the samples per frame calculation for MP3 if it's MPEG-2 or
	  MPEG-2.5.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765725

2016-04-27 20:46:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Update caps for TCP whenever they change
	  We only changed them for UDP so far, which caused the wrong seqnum-base and
	  other information to be passed to rtpjitterbuffer/etc when seeking. This
	  usually wasn't that much of a problem as the code there is robust enough, but
	  every now and then it causes us to drop up to 32756 packets before we
	  continue doing anything meaningful.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765689

2016-04-27 20:33:38 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Ensure to not take caps with the wrong pt for getting the clock-rate
	  Especially the caps on the pad might be out of date, and the new caps would be
	  provided for the current pt via the request-pt-map signal.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765689

2016-04-27 18:27:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't propagate spurious state change returns from internal elements further
	  We handle them inside rtspsrc and override them in all other cases anyway, so
	  do the same for "internal" state changes like PAUSED->PAUSED and
	  PLAYING->PLAYING.
	  This keeps unexpected NO_PREROLL to confuse state changes in GstBin.
	  See also https://bugzilla.gnome.org/show_bug.cgi?id=760532
	  https://bugzilla.gnome.org/show_bug.cgi?id=765689

2016-04-27 14:09:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavimux.c:
	  avimux: Don't override maximum audio chunk size with the scale again just before writing it
	  set_fields() should only be called in the beginning, otherwise we will never
	  remember the maximum audio chunk size and write a wrong block align... which
	  then causes wrong timestamps and other problems.

2016-04-27 13:53:00 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavimux.c:
	  avimux: Actually store the largest audio chunk size for the VBR case of MP2/MP3
	  3ea338ce271e1f6a96d2ed49d4472b091f6f8b7e changed avimux to do that, but it
	  never actually kept track of the max audio chunk for MP3 and MP2. These are
	  knowing the hdr.scale only after parsing the frames instead of at setcaps
	  time.

2016-04-25 15:03:14 +0200  Mats Lindestam <matslm@axis.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Allow setting "socket-v6" without setting "socket" too
	  https://bugzilla.gnome.org/show_bug.cgi?id=764897

2016-04-22 15:02:16 +0100  Mario Sanchez Prada <mario@endlessm.com>

	* ext/vpx/gstvpxenc.c:
	  vpxenc: Properly handle frames with too low duration
	  When a frame's duration is too low, calling gst_util_uint64_scale()
	  to scale its value can result into it being truncated to zero, which
	  will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error
	  when trying to encode.
	  To prevent this from happening, we simply ignore the duration when
	  encoding if it becomes zero after scaling, logging a warning message.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765391

2016-04-22 15:48:08 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix description of linear interlacing method

2016-04-21 14:08:19 -0300  Thibault Saunier <tsaunier@gnome.org>

	* gst/flv/gstflvmux.c:
	  flv: Handle the case where we do not get any CollectData in handle_buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=765320

2016-04-11 22:41:20 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Do not use unreliable framerate
	  timescale/1 is unreliable value for framerate. Due to downstream
	  element usually use framerate generated by qtdemux, let it be omitted
	  until the framerate can be reliably calculated.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764733

2016-04-21 12:53:33 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  Revert "qtdemux: expose streams with first moof for fragmented format"
	  This reverts commit d8bb6687ea251570c331038279a43d448167d6ad.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764733

2016-02-09 17:17:09 +0000  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: support seeking of CENC encrypted streams
	  When playing a stream that has been protected by DASH CENC, playback
	  will fail if a seek is performed. Qtdemux produces the error "stream
	  is protected using cenc, but no cenc protection system information
	  has been found" and playback stops.
	  The problem is that gst_qtdemux_reset() gets called as part of the
	  FLUSH during a seek. This function frees the protection_system_ids
	  array. When gst_qtdemux_configure_protected_caps() is called after the
	  seek has completed, the protection_system_ids array is empty and
	  qtdemux is unable to create the correct output caps for the protected
	  stream.
	  This commit changes it to only free the protection_system_ids on
	  hard resets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761787

2016-04-18 14:33:10 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: add "retrieve-sender-address" property
	  This allows disabling of sender address retrieval, which might
	  be useful in certain scenarios, like when the socket is connected,
	  or the sender address is not of interest (e.g. when receiving an
	  MPEG-TS stream). Disabling sender address retrieval in those
	  cases can have minor performance advantages.
	  https://bugzilla.gnome.org/show_bug.cgi?id=563323

2015-11-26 13:15:06 +0100  Dimitrios Katsaros <patcherwork@gmail.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Change warning handling to break infinite message loop
	  v4l2src can cause an "infinite message loop" when a base control exposed as a
	  property is not provided by the device. In these cases, if in the warning message
	  handling for the bus, the GST_DEBUG_BIN_TO_DOT_FILE* category of functions are used,
	  the src lookup causes a new warning to be posted on the bus, causing a loop.
	  This patch changes the warning for these controls so they are not posted on the bus.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758703

2016-04-15 10:44:02 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  spitmuxsink: Avoid creating small file at EOS
	  When EOS is reached, the current file get closed and the last
	  GOP in the mq was written in a new file.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765072

2016-04-15 19:55:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/mpg123/gstmpg123audiodec.h:
	  mpg123: fix build with msvc
	  Fix syntax errors when compiling against cerbero-provided libmpg123
	  headers. We do the same as the libmpg123 internal visual studio
	  build here.
	  mpg123.h(1378): error C2143: syntax error: missing ')' before '('
	  mpg123.h(1378): error C2081: 'ssize_t': name in formal parameter list illegal
	  mpg123.h(1378): error C2143: syntax error: missing ')' before '*'
	  mpg123.h(1378): error C2091: function returns function
	  mpg123.h(1378): error C2143: syntax error: missing '{' before '*'
	  mpg123.h(1378): error C2059: syntax error: ')'
	  mpg123.h(1379): error C2143: syntax error: missing ')' before '*'
	  mpg123.h(1379): error C2365: 'off_t': redefinition; previous definition was 'typedef'
	  ...

2016-04-15 19:59:15 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: S16 uses S32 temporary buffers, float/double their own type
	  Make sure to allocate not only a S16 buffer for S16 but a twice as big one to
	  hold S32.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765116

2016-04-16 02:17:26 +1000  Jan Schmidt <jan@centricular.com>

	* ext/pulse/pulsesink.c:
	  Revert "pulsesink: uncork if needed upon commit"
	  This reverts commit 0dd46accf6d282ff07065852bd91c85c78af3394.
	  With some audiosinks, starting the ringbuffer on the first commit
	  causes audio glitches at startup by starting to output segments
	  from the ringbuffer before it has been filled / fully prerolled. This
	  doesn't usually happen with pulsesink because we map the pulseaudio
	  ringbuffer directly, but we should keep things consistent with
	  other sinks with regards to startup latency, plus it gives more
	  headway to avoid glitching, should the initial 2nd segment take
	  more than 10ms to generate.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657076

2016-04-15 00:46:56 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add srtp rollover counters from mikey crypto sessions
	  The server can send multiple crypto sessions, one for each SSRC with its
	  own rollover counter. We parse this information and pass it to the SRTP
	  decoder via the "request-key" signal.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730540

2016-04-15 14:35:07 +0000  Jan Schmidt <jan@centricular.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix debug output when resyncing
	  Don't output the pointer value of the time() function as a timestamp
	  by using the correct variable.
	  Fixes build on Raspberry Pi 3.

2016-04-15 11:36:36 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: If no proxy is set by properties, use the default libsoup proxy resolver
	  That is, use whatever system settings there might exist. This is the same
	  behaviour we use in the HTTP source.

2016-04-14 10:01:28 +0100  Julien Isorce <j.isorce@samsung.com>

	* README:
	* common:
	  Automatic update of common submodule
	  From 6f2d209 to ac2f647

2016-04-13 18:45:07 +0100  Damian Ziobro <damian@xmementoit.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Add max_files_number property
	  https://bugzilla.gnome.org/show_bug.cgi?id=744612

2016-04-13 10:57:03 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: drop reference to videomixer 2
	  Fix a small grammar mistake on "overlayed" while at it.

2016-04-13 09:57:16 +0300  Sebastian Dröge <sebastian@centricular.com>

	* sys/ximage/ximageutil.c:
	  ximage: Initialize all fields in the meta explicitly
	  The meta is not allocated with all fields initialized to zeroes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764902

2016-04-12 09:41:00 +0000  Paolo Pettinato <ppettina@cisco.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Forward sticky events on buffer lists too, not only on buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=764933

2016-04-12 15:01:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Drain the field history if the caps are changing
	  Otherwise we will use fields from the old caps with everything set up for the
	  new caps, causing crashes and worse.
	  Also don't do anything if the same caps are set twice.

2016-04-12 15:00:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Instead of confusing crashes later, just error out immediately if mapping a video frame fails
	  This probably still crashes but at least we get some hint about what goes
	  wrong instead of random behaviour later.

2016-04-12 11:38:51 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check stream is available in PIFF parser
	  qtdemux->streams is an array, it will never evaluate to true when comparing
	  to NULL. Instead we want to check the number of streams to make sure the
	  stream is available.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753614
	  CID 1358389

2016-04-12 11:37:36 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  Revert "qtdemux: redundant check in PIFF parser"
	  This reverts commit 41e10524f3babdd92aac8c8c9d5b9cdf184c2d4e.

2016-04-12 11:05:50 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: redundant check in PIFF parser
	  qtdemux->streams is an array of size GST_QTDEMUX_MAX_STREAMS, it will never
	  evaluate to true when comparing to NULL.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753614
	  CID 1358389

2016-04-12 11:56:08 +0200  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: avoid leaking GValues
	  unset the GValue if we don't use it any more to avoid leaks.

2016-04-12 10:15:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix rtp_jitter_buffer_get_ts_diff() fill level calculation
	  The head of the queue is the oldest packet (as in lowest seqnum), the tail is
	  the newest packet. To calculate the fill level, we should calculate tail-head
	  while considering wraparounds. Not the other way around.
	  Other code is already doing this in the correct order.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764889

2016-04-11 10:44:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/Makefile.am:
	  rtpmanager: It's GST_LIBS, not GST_LIBS_LIBS

2016-04-11 08:33:17 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix parsing segment duration of empty edit list box
	  For empty edit list, segment-duration in edit list box should not be
	  used for segment event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764870

2016-04-08 13:05:57 +0200  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: make timecodescale configurable
	  In some use cases the default timecodescale will produce blocks with the same timestamp
	  https://bugzilla.gnome.org/show_bug.cgi?id=764769

2016-04-07 13:01:52 +0200  Edward Hervey <edward@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jiterbuffer: Move assertion to the right location
	  We shouldn't have "late" lost timers at that point

2016-03-02 14:25:24 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Speed up lost timeout handling
	  When downstream blocks, "lost" timers are created to notify the
	  outgoing thread that packets are lost.
	  The problem is that for high packet-rate streams, we might end up with
	  a big list of lost timeouts (had a use-case with ~1000...).
	  The problem isn't so much the amount of lost timeouts to handle, but
	  rather the way they were handled. All timers would first be iterated,
	  then the one selected would be handled ... to re-iterate the list again.
	  All of this is being done while the jbuf lock is taken, which in some use-cases
	  would return in holding that lock for 10s... blocking any buffers from
	  being accepted in input... which would then arrive late ... which would
	  create plenty of lost timers ... which would cause the same issue.
	  In order to avoid that situation, handle the lost timers immediately when
	  iterating the list of pending timers. This modifies the complexity from
	  a quadratic to a linear complexity.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762988

2016-03-02 14:23:01 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Don't create lost events if we don't need them
	  When "do-lost" is set to FALSE we don't use/send the lost events.
	  In that case, don't create them to start with :)
	  https://bugzilla.gnome.org/show_bug.cgi?id=762988

2016-03-02 13:57:07 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Add tracing of lock usage
	  Helps with debugging lock usage
	  https://bugzilla.gnome.org/show_bug.cgi?id=762988

2016-02-10 19:56:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2: Don't leak v4l2 objects and props on probe errors

2016-04-04 17:42:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: add unit test for jpeg depayloader packet loss handling
	  Make sure it always outputs something that looks like a valid
	  JPEG frame, ie. starts with an SOI marker and ends with an EOI
	  marker.

2016-03-15 03:25:26 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtp/gstrtpjpegdepay.c:
	  rtpjpegdepay: Don't send invalid frames downstream after packet loss or a DISCONT
	  After clearing the adapter due to a DISCONT, as might happen when some packet(s)
	  have been lost, the depayloader was pushing data into the adapter (which had no
	  header due to the clear), creating a headerless frame out of it, and sending it
	  downstream. The downstream decoder would then usually ignore it; unless there
	  were lots of DISCONTs from the jitterbuffer in which case the decoder would reach
	  its max_errors limit and throw an element error. Now we just discard that data.
	  It is probaby not worth trying to salvage this data because non-progressive
	  jpeg does not degrade gracefully and makes the video unwatchable even with
	  low packet loss such as 3-5%.

2016-01-05 16:15:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtpjitterbuffer: Add RFC7273 media clock handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=762259

2015-07-10 09:44:15 +0200  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: PIFF box detection and parsing support
	  The PIFF data is stored in a custom UUID box which is parsed and the
	  crypto_info of the element is updated accordingly. This allows
	  downstream decryptors to process and decrypt the protected content.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753614

2016-04-01 12:15:05 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbisdepay: remove dead code
	  payload_buffer hasn't been assigned a value before the jumps to
	  switch_failed or packet_short. So the value must be NULL. No need
	  to unmap and unref.
	  CID #1316476

2016-03-31 14:57:20 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: fix leak
	  Free memory of current macroblock once it isn't needed so it isn't leaked
	  by the call of the gst_rtp_h263_pay_B_mbfinder function.
	  if (!(mac = gst_rtp_h263_pay_B_mbfinder (context, gob, mac, mb))) {
	  CID 1212156

2016-03-31 02:15:04 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: Handle a hang draining out at EOS
	  Make sure that all data is drained out when the reference pad
	  goes EOS. Fixes a problem where data that arrives on other
	  pads after the reference pad finishes can stall forever and
	  never pass EOS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763711

2016-03-18 15:45:01 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix occasional deadlock when ending file with subtitle
	  Deadlock occurs when splitting files if one stream received no buffer during
	  the first GOP of the next file. That can happen in that scenario for example:
	  1) The first GOP of video is collected, it has a duration of 10s.
	  max_in_running_time is set to 10s.
	  2) Other streams catchup and we receive the first subtitle buffer at ts=0 and
	  has a duration of 1min.
	  3) We receive the 2nd subtitle buffer with a ts=1min. in_running_time is set to
	  1min. That buffer is blocked in handle_mq_input() because
	  max_in_running_time is still 10s.
	  4) Since all in_running_time are now > 10s, max_out_running_time is now set to
	  10s. That first GOP gets recorded into the file. The muxer pop buffers out
	  of the mq, when it tries to pop a 2nd subtitle buffer it blocks because the
	  GstDataQueue is empty.
	  5) A 2nd GOP of video is collected and has a duration of 10s as well.
	  max_in_running_time is now 20s. Since subtitle's in_running_time is already
	  1min, that GOP is already complete.
	  6) But let's say we overran the max file size, we thus set state to
	  SPLITMUX_STATE_ENDING_FILE now. As soon as a buffer with ts > 10s (end of
	  previous GOP) arrives in handle_mq_output(), EOS event is sent downstream
	  instead. But since the subtitle queue is empty, that's never going to
	  happen. Pipeline is now deadlocked.
	  To fix this situation we have to:
	  - Send a dummy event through the queue to wakeup output thread.
	  - Update out_running_time to at least max_out_running_time so it sends EOS.
	  - Respect time order, so we set out_running_tim=max_in_running_time because
	  that's bigger than previous buffer and smaller than next.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763711

2015-11-17 18:17:35 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Add new signal 'on-app-rtcp'
	  Similar to the 'on-feedback-rtcp' signal, but emitted for RTCP APP
	  packets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762217

2016-03-24 15:57:11 +0900  Minjae Kim <nate.kim@lge.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpmanager: Set to initial value for 'ntpns' in get_current_times()
	  Initialize "ntpns" variable to -1 as the OE compiler for some reason doesn't
	  realize that the variable is set in all code paths.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764119

2016-03-27 14:29:58 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtksink.c:
	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Fix logging in base widget and fix desc of GL sink
	  Set a default category for gtkgstbasewidget lest the logging go to the 'default'
	  category where it can't be found easily

2016-01-31 11:08:38 +1100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Allow different quantization tables for components 2 and 3
	  RFC 2435 mentions in section 4.1 that U/V use table number 1, but this seems
	  just like an example. Some encoders are not following that and there seems to
	  be no reason to reject their streams.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761345

2016-03-25 17:49:14 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk/gl: don't assert when gdk doesn't provide a GL context
	  Allows the application to check whether gtkglsink is supported by setting
	  the element to READY.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764148

2016-03-24 19:23:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: Use threads on multi-core systems
	  This is a redo of commit b848c1b6ffd1e508228820a013f94fb445e4777f. The
	  code was lost when the elements where ported to use a baseclass.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764169

2016-02-29 23:40:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsink.c:
	* tests/check/elements/splitmux.c:
	  splitmuxsink: only try to create internal sink if it doesn't exist
	  This allows splitmuxsink to be reused after being put to NULL.
	  Test included
	  https://bugzilla.gnome.org/show_bug.cgi?id=762893

2015-10-01 13:41:23 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: probe all colorspace supported by device
	  A device can support more than one colorspace for a given image
	  dimension and pixel format. So we have to probe all the supported
	  colorspace and not only rely on the default one. Otherwise we could end
	  up with negotiation failure if the caps colorimetry field don't match
	  the v4l2 device default one even if the v4l2 could support such
	  colorimetry.
	  This patch enable probing if colorspace for both capture and output
	  device. It really makes sense for output device since the colorspace
	  shall be set by the application and a little less for capture device
	  which, at the moment, shall provide the colorspace; ie: the v4l2
	  specification seems to not take into account the fact that a capture
	  device could do colorspace conversion.
	  As a side effet, probing takes some times and so sligthly delay v4l2
	  initialization. Note that this patch only probe colorspace and not all
	  colorspace, matrix, transfer and range combination to avoid taking too
	  much time, especially with low-speed devices as full probing do 1782
	  ioctl.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755937

2016-03-24 16:21:56 +0100  Edward Hervey <edward@centricular.com>

	* tests/check/elements/flvdemux.c:
	  check: Fix indentation

2016-03-24 16:20:39 +0100  Edward Hervey <edward@centricular.com>

	* tests/check/elements/flvdemux.c:
	  tests: Remove unused variables

2016-03-10 08:44:57 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/gtk/gstgtkbasesink.c:
	  gtkbasesink: post message to application for unhandled keyboard/mouse events
	  https://bugzilla.gnome.org/show_bug.cgi?id=763403

2016-03-04 15:50:26 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/qt/gstqtsink.cc:
	  bad: use new gst_element_class_add_static_pad_template()
	  https://bugzilla.gnome.org/show_bug.cgi?id=763081

2016-03-04 15:50:26 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtksink.c:
	  bad: use new gst_element_class_add_static_pad_template()
	  https://bugzilla.gnome.org/show_bug.cgi?id=763081

2016-03-16 20:26:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: Return the current caps on the srcpads on caps queries
	  It's not like we could accept any other caps here. The caps are decided by the
	  upstream caps event.
	  Also keep the filter order intact when filtering the results against the
	  filter caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763326

2016-03-04 16:14:44 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/twolame/gsttwolamemp2enc.c:
	  ugly: use new gst_element_class_add_static_pad_template()
	  https://bugzilla.gnome.org/show_bug.cgi?id=763082

2016-03-04 16:14:44 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/lame/gstlamemp3enc.c:
	  ugly: use new gst_element_class_add_static_pad_template()
	  https://bugzilla.gnome.org/show_bug.cgi?id=763082

2016-03-24 15:14:23 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix qtdemux memory leak in src_convert function
	  If we don't find the index of the sample correctly in src_convert function,
	  we have to unref about the qtdemux before returning value.
	  So, I have modify it about instead pass qtdemux as a parameter into
	  src_convert function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763973

2016-03-22 13:15:20 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add check condition for fail case in get_duration function
	  Currently, get_duration function always return the TRUE even though
	  it can't be set duration correctly. So, we need to add the else condition
	  about the fail case. Also, we already set the GST_CLOCK_TIME_NONE
	  in this function. So I have modify it which is related code in some
	  function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763968

2016-03-21 10:11:23 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Modify data type of duration in handle_src_query function
	  Data type of duration need to modify from guint64 to GstClockTime
	  for consistency in handle_src_query function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763965

2016-03-18 14:40:58 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Added unit tests for field=auto
	  https://bugzilla.gnome.org/show_bug.cgi?id=763869

2016-03-17 21:21:02 +0200  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Added "auto" fields mode
	  The "auto" fields mode will detect the upstream and downstream framerates and
	  will decide to deinterlace all or only top fields.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763869

2016-03-16 20:17:55 +0100  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvdemux.c:
	* tests/check/elements/flvdemux.c:
	  flvdemux: don't emit pad-added until caps are ready
	  In other words, gst_pad_get_current_caps should never return NULL
	  in a pad-added callback from the demuxer.
	  Added tests for the two special cases with AAC and H.264 where this
	  would happen every time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763780

2016-03-04 10:30:12 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/aalib/gstaasink.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/gstscaletempo.c:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/gstcapsdebug.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/interleave/deinterleave.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpL24pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpstreamdepay.c:
	* gst/rtp/gstrtpstreampay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp9depay.c:
	* gst/rtp/gstrtpvp9pay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideomedian.c:
	* gst/videomixer/videomixer2.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* gst/y4m/gsty4mencode.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	* tests/check/elements/autodetect.c:
	* tests/check/elements/qtmux.c:
	  good: use new gst_element_class_add_static_pad_template()
	  https://bugzilla.gnome.org/show_bug.cgi?id=763076

2016-03-04 09:42:44 +0100  David Buchmann <david.buchmann@gmail.com>

	* tests/check/elements/flvmux.c:
	  flvmux: Test to verify flvmux handles DTS with GST_CLOCK_TIME NONE
	  https://bugzilla.gnome.org/show_bug.cgi?id=762207

2015-11-04 14:51:19 +0900  Jihae Yi <jihae.yi@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid potentially overflowing expression
	  https://bugzilla.gnome.org/show_bug.cgi?id=757569

2016-03-22 10:43:45 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add the function to get channels and sample rate for AAC
	  Add aac_get_channels and sample_rate function to get these value for
	  AAC.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749110

2016-03-24 13:33:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.8.0 ===

2016-03-24 12:27:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.8.0

2016-03-24 12:02:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2016-03-16 20:18:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: Use GstIterator for iterating all pads instead of manually iterating them while holding the object lock all the time
	  Doing queries while holding the object lock is a bit dangerous, and in this
	  case causes deadlocks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763326

2016-03-17 20:53:27 +0200  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix typo to not change the input caps but our filtered caps
	  Changing the input caps and not using them anymore afterwards is useless, and
	  it breaks negotiation in pipelines like:
	  gst-launch-1.0 videotestsrc ! "video/x-raw,framerate=25/1,interlace-mode=interleaved" !
	  deinterlace fields=all ! "video/x-raw,framerate=50/1,interlace-mode=progressive" !
	  fakesink

=== release 1.7.91 ===

2016-03-15 12:04:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.7.91

2016-03-15 11:53:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2016-03-15 11:41:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/hu.po:
	* po/sr.po:
	  po: Update translations

2016-03-15 03:26:14 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: Some comment and documentation clarifications/fixes

2016-03-13 10:33:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  Revert "flacparse: push tags in pre_push_frame"
	  This reverts commit 4065fcb80a49924b70f0c8fc159dec0ff47943a1.
	  flacparse should not push tags by itself, the base class is going to do that
	  while properly merging in upstream tags. It just didn't because of a bug in
	  the base class, which was hidden by this commit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763553

2016-02-25 05:17:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtp/dboolhuff.c:
	* gst/rtp/dboolhuff.h:
	* gst/rtp/gstrtpsbcpay.c:
	  win32: Don't use __attribute__ on MSVC
	  Use MSVC-equivalents for alignment and packing compiler directives when building
	  on MSVC

2016-02-25 05:16:42 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/matroska/ebml-read.c:
	  win32: Don't try to include xmath.h on newer Visual Studio

2016-02-25 05:16:09 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/monoscope/gstmonoscope.c:
	  gst Factor out endian-order RGB formats
	  MSVC seems to ignore preprocessor conditionals inside static pad
	  template macros.

2016-03-08 17:37:17 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  dirctsoundsink: Setting volume should not unmute
	  https://bugzilla.gnome.org/show_bug.cgi?id=755106

2016-03-08 13:57:24 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  dirctsoundsink: Fix volume reset on unmute
	  https://bugzilla.gnome.org/show_bug.cgi?id=755106

2016-03-08 13:03:55 +0100  Alban Bedel <alban.bedel@avionic-design.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: fix capture with bayer formats other than bggr
	  gst_v4l2_object_get_caps_info() always return V4L2_PIX_FMT_SBGGR8
	  for all bayer formats. This is obviously broken if the device use
	  another ordering. Fix this by properly reading the format parameter.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763318

2016-03-07 10:28:06 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: reset pending segment if we are already pushing one
	  When upstream is running in bytes in push-mode, qtdemux will
	  convert seeks from time to bytes and send it upstream. Upstream
	  element will perform a byte seek and send a byte segment to qtdemux
	  that will convert it to time and push it downstream.
	  There is, however, the pending_segment variable that stores a new
	  segment event to be pushed before the next data. When handling seeks
	  as mentioned above this variable was being ignored and, if it contained
	  some segment event, it would override the one resulting from the seek.
	  This would restore a previous segment and would cause the seek segment
	  to be discarded downstream.
	  This patch fixes this issue by unrefing any pending segment as the
	  seek from upstream should contain the latest one that should be
	  used, as requested by the application.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763165

2016-03-07 10:27:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: run gst-indent
	  Otherwise commits will fail with our indent check hook

2016-03-04 15:09:45 +0100  Josep Torra <n770galaxy@gmail.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix colorimetry for NV12
	  Replicate V4L2_MAP_QUANTIZATION_DEFAULT macro behavior.
	  At #v4l it was described that documentation might be wrong and that
	  we should trust this macro instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762529

2016-03-05 11:38:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/gtk/Makefile.am:
	  gtk: examples: #define GST_USE_UNSTABLE_API and link with X11_LIBS
	  X11_LIBS is needed for XInitThreads() and without the #define we get
	  warnings about the GL API being still unstable.

2016-03-04 14:07:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix multicast group joining with provided sockets on Windows
	  On Windows the socket will be bound to ANY instead of the multicast group,
	  as binding to a multicast group does not work. Which would mean that we
	  override src->addr to become ANY and won't automatically join a multicast
	  group anymore on Windows.
	  On Linux we would automatically join a multicast group, keep it consistent.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763093

2016-03-01 18:22:37 +0300  Sergey Borovkov <sergey.borovkov@wireload.net>

	* ext/qt/qtitem.cc:
	  qml: Fix leak of the OpenGL contexts
	  [Matthew Waters]: add NULL checks before unreffing
	  https://bugzilla.gnome.org/show_bug.cgi?id=762999

2016-03-02 13:13:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Revert "rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases"
	  This reverts commit a7fb7b53592d87f7983544debb74d364fc3257ad.
	  The mutex is taken by the caller, we should keep it locked when returning so
	  the caller can unlock it again.

2016-03-01 15:01:22 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: push tags in pre_push_frame
	  Push a tag event before pre-roll if we have tags.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762660

=== release 1.7.90 ===

2016-03-01 18:15:43 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.7.90

2016-03-01 17:03:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/ca.po:
	* po/da.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/or.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/tr.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2016-03-01 16:53:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/bg.po:
	* po/cs.po:
	* po/de.po:
	* po/fr.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: Update translations

2016-03-01 14:14:02 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases

2016-02-29 10:10:24 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: remove impossible condition
	  It is impossible for a guint to have a negative value, no need to check for
	  this. Introduced in commit 6861d11c49ea0f30d2432cf4ebf6108bc89897f1
	  CID 1354509

2016-02-28 10:12:36 +0100  Petr Viktorin <encukou@gmail.com>

	* gst/alpha/gstalpha.c:
	  alpha: Fix sample pipeline
	  Use the zorder pad property to make sure the semitransparent
	  video is on top of the background.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762809

2016-02-28 13:42:28 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/replaygain/gstrgvolume.c:
	* tests/check/elements/rgvolume.c:
	  rgvolume: make tag list writable before modifying it
	  Making the event itself writable is not enough, it won't make
	  the actual taglist in the event writable as well. Instead, just
	  make a copy of the taglist and then create a new tag event from
	  that if required, replacing the old one. Before we would
	  inadvertently modify taglists upstream elements might still
	  be holding on to. Add unit test for this as well.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762793

2016-02-28 13:01:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Properly error out if binding the UDP sockets fails
	  udpsrc is not returning us a socket in that case.

2016-02-27 20:33:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/goom/gstgoom.c:
	  goom: Use goom_set_resolution() instead of recreating the goom instance when the resolution changes
	  https://bugzilla.gnome.org/show_bug.cgi?id=762765

2016-02-27 20:32:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/goom/gstgoom.c:
	  Revert "goom: Initialize the goom struct only once we know width/height and recreate it if those change"
	  This reverts commit cc6e102643c1bae928316dca9f34db028fb9a67e.

2016-02-27 20:31:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/goom/gstgoom.c:
	  goom: Initialize the goom struct only once we know width/height and recreate it if those change
	  Fixes crash when the width and/or height is changing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762765

2016-02-26 12:41:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* common:
	  Automatic update of common submodule
	  From b64f03f to 6f2d209

2016-02-25 22:54:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-rtp.xml:
	  docs: add rtpopusdepay and rtpopuspay to documentation

2016-02-17 15:15:11 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopusdepay.h:
	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtpopuspay.h:
	  rtp: opus: move Opus RTP payloader/depayloader from -bad to -good
	  https://bugzilla.gnome.org/show_bug.cgi?id=756282

2016-02-17 15:10:00 +0000  Tim-Philipp Müller <tim@centricular.com>

	  Merge branch 'plugin-move-rtp-opus'
	  Move Opus RTP depayloader/payloader from -bad to -good.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756282

2016-02-25 11:33:13 +0100  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: cenc aux info parsing from mdat support in PULL mode
	  This is already supported for PUSH mode but was failing in PULL mode.
	  The aux info is sometimes stored in the mdat before the first sample,
	  so the loop task needs to pull data stored at that location and
	  perform the aux info cenc parsing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761700
	  https://bugzilla.gnome.org/show_bug.cgi?id=762516

2016-02-24 11:28:09 +0100  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: prevent buffer flow if any stream failed to be exposed
	  In some cases the stream configuration can fail, for instance if the
	  stream is protected and no decryptor was found. For those situations
	  the demuxer shouldn't emit any data on the corresponding source pad of
	  the stream and bail out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762516

2016-02-24 09:12:03 +0100  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't push encrypted buffer without cenc metadata
	  When the cenc metadata is stored outside of the moof box and the
	  stream is exposed it is possible that the cenc metadata hasn't been
	  processed yet while the first buffer is being pushed. When this
	  happens the buffer can't possibly be decrypted downstream so don't
	  push it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762516

2016-02-23 23:10:20 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtsink.cc:
	* ext/qt/qtitem.cc:
	  qt: use a static_cast instead of dynamic one
	  The dynamic_cast is a little but of overkill as the app will still crash if it
	  fails in the later g_assert.
	  Allows compilation with -fno-rtti
	  https://bugzilla.gnome.org/show_bug.cgi?id=762526

2015-10-21 16:21:45 +0200  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: read saio aux_info_type as a FOURCC
	  https://bugzilla.gnome.org/show_bug.cgi?id=756897

2016-02-23 18:27:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstdvdec.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/smpte/gstsmpte.c:
	  gst: Handle gst_pad_get_current_caps() returning NULL gracefully

2016-02-23 18:12:54 +0200  Dave Craig <dcraig@brightsign.biz>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
	  Remove calls to gst_pad_has_current_caps() which then go on to call
	  gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
	  use gst_pad_get_current_caps() and check for NULL.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759539

2015-12-16 12:40:39 +0000  Dave Craig <dcraig@brightsign.biz>

	* ext/flac/gstflacenc.c:
	* gst/flv/gstflvmux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/videocrop/gstaspectratiocrop.c:
	  gst: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
	  Remove calls to gst_pad_has_current_caps() which then go on to call
	  gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
	  use gst_pad_get_current_caps() and check for NULL.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759539

2015-12-16 10:54:17 +0000  Dave Craig <dcraig@brightsign.biz>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Handle gst_pad_get_current_caps() returning NULL gracefully
	  This can happen when the pipeline is currently shutting down.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759539

2016-02-23 15:57:18 +0100  Linus Svensson <linussn@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Don't handle seek until ready
	  https://bugzilla.gnome.org/show_bug.cgi?id=762542

2016-02-23 15:55:13 +0100  Linus Svensson <linussn@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Unref seek event
	  https://bugzilla.gnome.org/show_bug.cgi?id=762542

2016-02-22 11:01:40 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: close file on write error with next-file mode is set to buffer
	  If we have an error during fwrite call, file stays open and thus next
	  incoming buffer will trigger an assert when trying to opening a new
	  file.
	  This happens if we do not restart element, file is closed at stop, and
	  if application handles the returned GST_FLOW_ERROR to keep bin alive.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762434

2016-02-19 23:44:42 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: don't output empty tags/tag elements
	  Such files will not play on Android, because of bug in libwebm matroska parsing, which is still present in 6.0.1
	  https://bugzilla.gnome.org/show_bug.cgi?id=762349

2016-02-04 15:59:04 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: make up an OpusHead block if possible when missing
	  https://bugzilla.gnome.org/show_bug.cgi?id=761489

2016-02-04 10:43:15 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: make up an OpusHead block if possible when missing
	  This block is needed in the Matroska file, but data coming from
	  RTP may not have one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761489

2016-02-22 13:53:21 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: make stream-id more readable and order-friendly
	  ... as streams are so ordered by id by e.g. decodebin
	  (and as typically already honoured by other demuxers).

2016-02-22 13:25:51 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroska: remove confusing duplicate track uid field

2016-02-22 14:03:02 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtpvp9pay.c:
	  rtpvp9pay: add missing break
	  VP9_PAY_PICTURE_ID_7BITS and VP9_PAY_PICTURE_ID_15BITS are mutually
	  exclusive options of the picture-id-mode. We can break after the
	  first case.
	  1 or 2 bytes need to be added to the header length depending on the
	  PictureID size.
	  https://tools.ietf.org/html/draft-uberti-payload-vp9-00#section-4.2
	  CID 1353479

2016-01-24 17:40:37 +0300  Sergey Borovkov <sergey.borovkov@wireload.net>

	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  qmlglsink: Schedule onSceneGrpahInitialized to execute on render thread
	  onSceneGraphInitialized() is called from non render thread currently when
	  scene graph is already initialized.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761003

2016-02-22 09:09:01 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix buffer memory leak
	  buffer being mapped is not being unmapped in some cases
	  https://bugzilla.gnome.org/show_bug.cgi?id=762420

2015-11-04 10:19:03 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpmanager: Don't warn for duplicate/reordered packets
	  This is a normal scenario and should not be a warning.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762208

2016-02-21 09:47:43 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/alpha/alpha.vcproj:
	* gst/auparse/auparse.vcproj:
	* gst/avi/avi.vcproj:
	* gst/cutter/cutter.vcproj:
	* gst/debugutils/debug.vcproj:
	* gst/debugutils/navigationtest.vcproj:
	* gst/effectv/effectv.vcproj:
	* gst/flx/flxdec.vcproj:
	* gst/goom/goom.vcproj:
	* gst/goom2k1/goom.vcproj:
	* gst/interleave/interleave.vcproj:
	* gst/isomp4/qtdemux.vcproj:
	* gst/law/alaw.vcproj:
	* gst/law/mulaw.vcproj:
	* gst/matroska/matroska.vcproj:
	* gst/multipart/multipart.vcproj:
	* gst/rtp/rtp.vcproj:
	* gst/smpte/smpte.vcproj:
	* gst/spectrum/spectrum.vcproj:
	* gst/udp/udp.vcproj:
	* gst/videobox/videobox.vcproj:
	* gst/videocrop/videocrop.vcproj:
	* gst/videofilter/gamma.vcproj:
	* gst/videofilter/videobalance.vcproj:
	* gst/videofilter/videofilter.vcproj:
	* gst/videofilter/videoflip.vcproj:
	* gst/videomixer/videomixer.vcproj:
	* gst/wavenc/wavenc.vcproj:
	* gst/wavparse/wavparse.vcproj:
	* gst/y4m/y4menc.vcproj:
	* win32/MANIFEST:
	* win32/vs6/autogen.dsp:
	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstalaw.dsp:
	* win32/vs6/libgstalpha.dsp:
	* win32/vs6/libgstalphacolor.dsp:
	* win32/vs6/libgstapetag.dsp:
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstauparse.dsp:
	* win32/vs6/libgstautodetect.dsp:
	* win32/vs6/libgstavi.dsp:
	* win32/vs6/libgstcutter.dsp:
	* win32/vs6/libgstdirectsound.dsp:
	* win32/vs6/libgsteffectv.dsp:
	* win32/vs6/libgstflx.dsp:
	* win32/vs6/libgstgoom.dsp:
	* win32/vs6/libgsticydemux.dsp:
	* win32/vs6/libgstid3demux.dsp:
	* win32/vs6/libgstinterleave.dsp:
	* win32/vs6/libgstjpeg.dsp:
	* win32/vs6/libgstlevel.dsp:
	* win32/vs6/libgstmatroska.dsp:
	* win32/vs6/libgstmedian.dsp:
	* win32/vs6/libgstmonoscope.dsp:
	* win32/vs6/libgstmulaw.dsp:
	* win32/vs6/libgstmultipart.dsp:
	* win32/vs6/libgstpng.dsp:
	* win32/vs6/libgstqtdemux.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstsmpte.dsp:
	* win32/vs6/libgstspeex.dsp:
	* win32/vs6/libgstudp.dsp:
	* win32/vs6/libgstvideobalance.dsp:
	* win32/vs6/libgstvideobox.dsp:
	* win32/vs6/libgstvideocrop.dsp:
	* win32/vs6/libgstvideoflip.dsp:
	* win32/vs6/libgstvideomixer.dsp:
	* win32/vs6/libgstwaveform.dsp:
	* win32/vs6/libgstwavenc.dsp:
	* win32/vs6/libgstwavparse.dsp:
	* win32/vs7/libgstdirectsound.vcproj:
	* win32/vs8/gst-plugins-good.sln:
	* win32/vs8/libgst1394.vcproj:
	* win32/vs8/libgstaasink.vcproj:
	* win32/vs8/libgstalaw.vcproj:
	* win32/vs8/libgstalpha.vcproj:
	* win32/vs8/libgstalphacolor.vcproj:
	* win32/vs8/libgstannodex.vcproj:
	* win32/vs8/libgstapetag.vcproj:
	* win32/vs8/libgstaudiofx.vcproj:
	* win32/vs8/libgstauparse.vcproj:
	* win32/vs8/libgstautodetect.vcproj:
	* win32/vs8/libgstavi.vcproj:
	* win32/vs8/libgstcacasink.vcproj:
	* win32/vs8/libgstcdio.vcproj:
	* win32/vs8/libgstcutter.vcproj:
	* win32/vs8/libgstdirectsound.vcproj:
	* win32/vs8/libgstdv.vcproj:
	* win32/vs8/libgsteffectv.vcproj:
	* win32/vs8/libgstflac.vcproj:
	* win32/vs8/libgstflxdec.vcproj:
	* win32/vs8/libgstgoom.vcproj:
	* win32/vs8/libgsticydemux.vcproj:
	* win32/vs8/libgstid3demux.vcproj:
	* win32/vs8/libgstjpeg.vcproj:
	* win32/vs8/libgstladspa.vcproj:
	* win32/vs8/libgstlevel.vcproj:
	* win32/vs8/libgstmatroska.vcproj:
	* win32/vs8/libgstmng.vcproj:
	* win32/vs8/libgstmonoscope.vcproj:
	* win32/vs8/libgstmulaw.vcproj:
	* win32/vs8/libgstmultipart.vcproj:
	* win32/vs8/libgstpng.vcproj:
	* win32/vs8/libgstrtp.vcproj:
	* win32/vs8/libgstrtsp.vcproj:
	* win32/vs8/libgstshout2.vcproj:
	* win32/vs8/libgstsmpte.vcproj:
	* win32/vs8/libgstspeex.vcproj:
	* win32/vs8/libgsttaglib.vcproj:
	* win32/vs8/libgstudp.vcproj:
	* win32/vs8/libgstvideobalance.vcproj:
	* win32/vs8/libgstvideobox.vcproj:
	* win32/vs8/libgstvideoflip.vcproj:
	* win32/vs8/libgstvideomixer.vcproj:
	* win32/vs8/libgstwavenc.vcproj:
	* win32/vs8/libgstwavparse.vcproj:
	  win32: remove outdated build cruft
	  This hasn't been touched for generations, doesn't work,
	  and is just causing confusion. We also don't want to
	  maintain these files manually.

2016-02-20 11:51:56 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: don't use undeclared core debug category symbols

2016-02-06 14:39:05 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: workaround for files with wrong color_table_id value
	  Instead of erroring out, just use the default color table.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761637

2016-02-19 15:02:04 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvmux.c:
	* gst/rtp/gstrtpvp9depay.c:
	  flvmux, rtpvp9depay: fix indentation

2016-02-19 15:03:04 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2src: fix indentation

2015-12-04 00:46:34 +1100  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvmux.c:
	  flvmux: plug leak(s) in error-scenario
	  https://bugzilla.gnome.org/show_bug.cgi?id=762210

2015-12-04 00:46:12 +1100  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix eos event leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=762209

2016-02-19 14:41:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/flvdemux.c:
	* tests/check/elements/flvmux.c:
	* tests/check/elements/rtph263.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  tests: fix indentation

2016-02-18 16:09:29 +0100  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: rtpjitterbuffer: port testharness to GstHarness and cleanup/improve
	  Probably found a bug as well, in that there are some timestamps in
	  there that are looking very wrong. (marked with FIXME)
	  https://bugzilla.gnome.org/show_bug.cgi?id=762267

2016-02-18 10:27:19 +0100  Havard Graff <havard.graff@gmail.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: rtpjitterbuffer: test cleanups/improvements
	  Use fail_unless and friends instead of g_assert
	  Factor seq-num checking out to separate function
	  Check more return-values from push and crank and others
	  https://bugzilla.gnome.org/show_bug.cgi?id=762254

2015-12-03 11:07:05 +0100  Stian Selnes <stian@pexip.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: rtpjitterbuffer: fix leaks in unit test
	  https://bugzilla.gnome.org/show_bug.cgi?id=762214

2016-02-19 12:38:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.7.2 ===

2016-02-19 11:49:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.7.2

2016-02-19 10:31:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: Update translations

2016-02-18 18:33:13 +0100  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: plug leaks in cenc aux info parsing

2016-02-18 13:43:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	  tests: fix spurious souphttpsrc test timouts
	  Set GSETTINGS_BACKEND=memory, apparently there's something
	  about fork() and the dconf backend (or whatever else that
	  drags in or activates) that messes up locking and causes
	  timeouts due to deadlocks in g_mutex_lock(), since
	  everything works fine with CK_FORK=no as well.

2016-02-18 11:10:14 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Unmap wavpack header buffer after creating it
	  Otherwise it will be mapped writable all the time and we can't read from it
	  anywhere.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762239

2015-12-08 18:49:40 +0100  Stian Selnes <stian@pexip.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Add test for big seqnum gap handling
	  Make sure that the packets queued when detecting a big gap are pushed
	  after reset (5 consective seqnums) and not dropped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762211

2016-02-17 15:03:13 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtputils.h:
	  rtp: sprinkle some G_GNUC_INTERNAL for internal utils functions

2016-02-09 13:17:00 +0000  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only transform protected caps once
	  Commit 7873bede3134b15e5066e8d14e54d1f5054d2063
	  (https://bugzilla.gnome.org/show_bug.cgi?id=760774) changed the
	  behaviour of qtdemux to call gst_qtdemux_configure_stream() for
	  every new moof.
	  When playing a protected stream, gst_qtdemux_configure_stream()
	  calls gst_qtdemux_configure_protected_caps(). The
	  gst_qtdemux_configure_protected_caps() function takes the original
	  media format, puts this in a field called "original-media-type"
	  and then changes the caps to "application/x-cenc".
	  The gst_qtdemux_configure_protected_caps() did not handle the case
	  of being called multiple times, causing it to incorrectly set the
	  caps. The second call was causing the caps to be set to:
	  application/x-cenc, original-media-type"application/x-cenc"
	  This commit makes gst_qtdemux_configure_protected_caps() check that
	  the caps have already been transformed, so that it only gets
	  changed once.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761769

2015-11-03 14:50:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  opus: Add proper support for multichannel audio
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-06-30 13:51:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  opus: Copy metadata in the (de)payloader, but only the relevant ones
	  The payloader didn't copy anything so far, the depayloader copied every
	  possible meta. Let's make it consistent and just copy all metas without tags or
	  with only the audio tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751774

2015-05-04 11:23:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpopusdepay.c:
	  opusdepay: Set multistream=FALSE on the Opus caps
	  The RTP Opus mapping only allows mono/stereo, and not multistream Opus
	  streams.

2015-03-24 13:57:54 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: Forward stereo preferences from caps upstream
	  https://bugzilla.gnome.org/show_bug.cgi?id=746617

2015-03-24 13:56:21 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: Set the number of channels to 2 as per RFC draft
	  https://bugzilla.gnome.org/show_bug.cgi?id=746617

2015-03-23 12:24:55 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  opus: Handle sprop-stereo and sprop-maxcapturerate RTP caps fields
	  https://bugzilla.gnome.org/show_bug.cgi?id=746617

2015-02-19 14:30:10 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: default encoding name to OPUS
	  https://bugzilla.gnome.org/show_bug.cgi?id=737810

2015-02-19 14:05:06 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: make caps writable before truncating them
	  https://bugzilla.gnome.org/show_bug.cgi?id=737810

2015-02-05 10:27:51 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: negotiate the encoding name
	  Chrome uses a different encoding name that gstreamer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737810

2014-11-01 10:10:27 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  rtpopus: Use OPUS encoding name
	  Both Firefox and Chrome uses OPUS as the encoding in their SDP.
	  Adding this now defacto standard name remove the need for special
	  case in SDP parsing code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737810

2013-01-31 12:30:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpopuspay.c:
	  opuspay: fix timestamps
	  Copy timestamps to payloaded buffer.
	  Avoid input buffer memory leak.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692929

2012-11-03 20:38:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopusdepay.h:
	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtpopuspay.h:
	  Fix FSF address
	  https://bugzilla.gnome.org/show_bug.cgi?id=687520

2012-10-22 12:08:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpopuspay.c:
	  opuspay: remove pointless caps serialization
	  Remove the caps serialization in the rtp caps. the spec nor the receiver
	  does anything with it.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686547

2012-10-17 17:34:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  Use gst_element_class_set_static_metadata()
	  where possible. Avoids some string copies. Also re-indent
	  some stuff. Also some indent fixes here and there.

2012-09-20 18:41:24 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: Allocate the rtp buffer correctly
	  Use the right functions to allocate the rtp buffer

2012-09-14 17:08:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  replace gst_element_class_set_details_simple with gst_element_class_set_metadata

2012-03-07 17:14:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpopuspay.c:
	  opus: port to updated 0.11

2011-12-30 11:41:17 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopusdepay.h:
	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtpopuspay.h:
	  Merge remote-tracking branch 'origin/master' into 0.11-premerge
	  Conflicts:
	  docs/libs/Makefile.am
	  ext/kate/gstkatetiger.c
	  ext/opus/gstopusdec.c
	  ext/xvid/gstxvidenc.c
	  gst-libs/gst/basecamerabinsrc/Makefile.am
	  gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
	  gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
	  gst-libs/gst/video/gstbasevideocodec.c
	  gst-libs/gst/video/gstbasevideocodec.h
	  gst-libs/gst/video/gstbasevideodecoder.c
	  gst-libs/gst/video/gstbasevideoencoder.c
	  gst/asfmux/gstasfmux.c
	  gst/audiovisualizers/gstwavescope.c
	  gst/camerabin2/gstcamerabin2.c
	  gst/debugutils/gstcompare.c
	  gst/frei0r/gstfrei0rmixer.c
	  gst/mpegpsmux/mpegpsmux.c
	  gst/mpegtsmux/mpegtsmux.c
	  gst/mxf/mxfmux.c
	  gst/videomeasure/gstvideomeasure_ssim.c
	  gst/videoparsers/gsth264parse.c
	  gst/videoparsers/gstmpeg4videoparse.c

2011-12-09 17:25:41 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpopuspay.c:
	  opusenc: add upstream negotiation for multistream ability
	  This will help elements that cannot deal with multistream,
	  such as the RTP payloader.
	  The caps now do not include a "streams" field anymore, but
	  a "multistream" boolean, since we have no real use for knowing
	  the exact amount of streams.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665078

2011-12-07 15:13:11 -0200  Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>

	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopusdepay.h:
	* gst/rtp/gstrtpopuspay.c:
	* gst/rtp/gstrtpopuspay.h:
	  Adding opus RTP payloader/depayloader element
	  Adding OPUS RTP module based on the current draft:
	  http://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.txt
	  https://bugzilla.gnome.org/show_bug.cgi?id=664817

2016-02-17 13:26:02 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtputils.c:
	* gst/rtp/gstrtputils.h:
	  rtp: h264/h265: avoid duplication of read_golomb()
	  There is no need to have two identical implementations of the read_golomb
	  function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761606

2016-02-17 14:37:44 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Simple implementation of TRICKMODE_KEY_UNITS
	  When the trickmode key-units flag is set on the segment, simply skip
	  any sample on a video stream that isn't a keyframe
	  https://bugzilla.gnome.org/show_bug.cgi?id=762185

2015-08-21 14:15:18 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: send GAP events for lagging audio and video streams too
	  Send GAP events for non-subtitle streams too if they lag too much
	  behind, but use a higher threshold than for subtitles.
	  This helps with fixing prerolling with a file where one of the
	  audio streams only has data starting from 19s onwards. It's not
	  a complete fix yet, it also requires changes elsewhere, such as
	  in baseparse, to make sure caps are propagated.
	  https://bugzilla.gnome.org/show_bug.cgi?id=614460
	  https://bugzilla.gnome.org/show_bug.cgi?id=753899

2015-12-23 19:54:13 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpvp9depay.c:
	* gst/rtp/gstrtpvp9depay.h:
	* gst/rtp/gstrtpvp9pay.c:
	* gst/rtp/gstrtpvp9pay.h:
	  rtpvp9pay: rtpvp9depay: Initial implementation of draft 01
	  Quick and dirty implementation of an RTP payloader and depayloader
	  for VP9. In particalur it assumes no spatial or temporal layering,
	  non-flexible mode, and some other bits and pieces.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754773

2016-02-16 09:02:30 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix string memory leak
	  codec_name is not being freed in all conditions leading to memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=762117

2015-12-10 12:15:52 +0100  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: add "get-session" signal
	  This gets the GstRTPSession element, as compared to the RTPSession object
	  that is returned by get-internal-session.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759293

2015-12-14 11:09:46 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  plugins-bad: Fix example pipelines
	  rename gst-launch --> gst-launch-1.0
	  replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
	  fix caps in examples
	  https://bugzilla.gnome.org/show_bug.cgi?id=759432

2015-08-17 11:50:28 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: still reset pending audio info on hard flush
	  Follow-up to previous commit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752431

2015-07-15 10:44:02 -0600  Jason Litzinger <jlitzinger@control4.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: fix handling of sample rate change during playback
	  If the sample rate of the media changes, the resulting flush will
	  clear the has_next_audioinfo flag, and the caps won't be sent
	  downstream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752431

2015-08-15 12:58:40 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  audiodecoders: use default pad accept-caps handling
	  Avoids useless check of downstream caps when handling an
	  accept-caps query
	  Elements: dtsdec, faad, gsmdec, mpg123audiodec, opusdec,
	  sbcdec, adpcmdec, sirendec

2015-04-26 18:04:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/mpg123/Makefile.am:
	  Remove obsolete Android build cruft
	  This is not needed any longer.

2015-01-11 01:08:08 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: fix compiler warning and simplify checks in set_caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=740195

2015-01-03 13:06:45 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: rework set_format code so mpg123audiodec works with decodebin/playbin
	  The old code was using gst_caps_normalize() and was generally overly
	  complex. Simplify by picking sample rate and number of channels from
	  upstream and the sample format from the allowed caps. If the format caps
	  is a list of strins, just pick the first one. And if the srcpad isn't
	  linked yet, use the default format (S16).
	  https://bugzilla.gnome.org/show_bug.cgi?id=740195

2014-09-10 17:24:39 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  Fix up one-element lists in template caps

2014-03-05 00:51:04 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/mpg123audiodec.c:
	  tests: fix mpg123audiodec test for big-endian architectures

2014-02-04 17:22:27 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: improved error report and checks
	  Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>

2013-12-05 12:04:39 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123audiodec: Require caps to be set before any data processing

2013-07-26 17:25:42 +0200  Edward Hervey <edward@collabora.com>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: Remove dead assignment
	  harder ? :)

2013-05-15 11:25:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/mpg123audiodec.c:
	  mpg123audiodec: Fix event handling in unit test

2012-10-24 12:16:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/mpg123/Makefile.am:
	  gst: Add better support for static plugins

2013-04-15 00:22:39 -0700  David Schleef <ds@schleef.org>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: Add conditional on API version for new enum

2016-02-16 19:59:13 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkbasesink.h:
	  gtk(gl)sink: remove the signal handlers on finalize
	  It's possible that the sink element will be freed before the widget is
	  destroyed.  When the widget was eventually destroyed, it was attempting to
	  access member variables of the freed sink struct which resulted in undefined
	  behaviour.
	  Fix by disconnecting our signal on finalize.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762098

2016-02-16 00:19:00 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	  rtp: h265: hook up move RTP H.265 payloader/depayloader to build
	  https://bugzilla.gnome.org/show_bug.cgi?id=761606

2016-02-16 00:14:27 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	* gst/rtp/gstrtph265pay.c:
	  rtp: h265: use common meta utility functions
	  https://bugzilla.gnome.org/show_bug.cgi?id=761606

2016-02-05 18:18:31 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.h:
	* gst/rtp/gstrtph265pay.h:
	* gst/rtp/gstrtph265types.h:
	  rtp: h265: remove codecparser dependency from h265 payloader/depayloader
	  Looks like it just uses the NAL enums and nothing else from
	  the codecparsers, and that's the only reason it had to be
	  moved from -good to -bad when it was originally added. We
	  can probably keep those NAL enums up to date enough, so let's
	  remove the codecparser dependency so it can be moved back into
	  -good.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761606

2016-02-16 00:24:58 +0000  Tim-Philipp Müller <tim@centricular.com>

	  Merge branch 'plugin-move-rtp-h265'
	  Move RTP H.265 payloader/depayloader from -bad to -good.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761606

2016-02-05 15:34:51 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	  gstrtph265depay: keep consistency with rtph264depay
	  Use gst_rtp_drop_meta() and the same function prototype for
	  gst_rtp_copy_meta() to keep consistency with the RTP elements in
	  gst-plugins-good

2016-02-05 13:56:34 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: fix termination of access unit
	  Only consider the access unit complete when the next-occurring VCL NAL unit
	  has the first bit after its NAL unit header equal to 1.

2016-01-15 16:10:02 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: fix unneeded sub-buffer creation
	  We create a sub-buffer just to copy over its metas and then throw it
	  away immediately, just use the original input buffer directly.

2016-01-15 15:56:59 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: add "send VPS/SPS/PPS with every key frame" mode
	  It's not enough to have timeout or event based VPS/SPS/PPS information
	  sent in RTP packets. There are some scenarios when key frames may appear
	  more frequently than once a second, in which case the minimum timeout
	  for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough.
	  It might also be desirable in general to make sure the VPS/SPS/PPS is
	  available with every keyframe (packet loss aside), so receivers can
	  actually pick up decoding immediately from the first keyframe if
	  VPS/SPS/PPS is not signaled out of band.
	  This commit adds the possibility to send VPS/SPS/PPS with every key frame.
	  This mode can be enabled by setting "config-interval" property to -1. In
	  this case the payloader will add VPS, SPS and PPS before every key (IDR)
	  frame.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757892

2016-01-15 15:19:41 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	  rtph265pay: change config-interval property type from uint to int
	  This way we can use -1 as special value, which is nicer than MAXUINT.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757892

2015-08-15 16:22:20 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: make sure we call handle_nal for each NAL
	  Call handle_nal for each NAL in the STAP-A RTP packet. This makes sure
	  we correctly extract the SPS and PPS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730999

2015-08-15 14:45:34 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Copy metadata in the payloader, but only the relevant ones
	  The payloader didn't copy anything so far, the depayloader copied every
	  possible meta. Let's make it consistent and just copy all metas without
	  tags or with only the video tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751774

2015-08-15 11:41:40 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()
	  https://bugzilla.gnome.org/show_bug.cgi?id=753228

2015-08-15 11:30:36 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: fix potential crash when shutting down
	  A race condition in the state change function may cause buffers to be
	  unreffed while they are still used by the streaming thread in
	  gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the
	  parent class first in the state change function to make sure streaming
	  has stopped and only then free those buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741381

2015-08-14 15:08:08 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265pay: fix buffer leak when using SPS/PPS
	  Fixes a buffer leak that would occur if the pipeline was shutdown while a
	  SPS/PPS header was being created.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741271

2015-08-14 11:49:51 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	  rtph265depay: copy metadata in the depayloader, but only the relevant ones
	  The payloader didn't copy anything so far, the depayloader copied every
	  possible meta. Let's make it consistent and just copy all metas without
	  tags or with only the video tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751774

2015-08-12 17:54:52 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: checking if depay has sps/pps nals before insertion
	  Related to: https://bugzilla.gnome.org/show_bug.cgi?id=753430
	  https://bugzilla.gnome.org/show_bug.cgi?id=753228

2015-08-12 17:22:42 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: only update the srcpad caps if something else than the codec_data changed
	  h264parse and gstrtph264depay do the same, let's keep the behaviour
	  consistent. As we now include the codec_data inside the stream, this causes
	  less caps renegotiation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753228

2015-08-12 16:43:48 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: PPS replaces old PPS if it has the same id
	  https://bugzilla.gnome.org/show_bug.cgi?id=753228

2015-08-12 16:11:00 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: Insert SPS/PPS NALs into the stream
	  rtph264depay does the same and this fixes decoding of some streams with 32
	  SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255),
	  but the field in the codec_data for the number of SPS or PPS is only 5
	  (or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
	  This looks like a mistake in the part of the spect about the codec_data.

2015-08-12 15:49:50 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: implement process_rtp_packet() vfunc
	  For more optimised RTP packet handling: means we don't need to map the
	  input buffer again but can just re-use the mapping the base class has
	  already done.
	  Based on: https://bugzilla.gnome.org/show_bug.cgi?id=750235
	  https://bugzilla.gnome.org/show_bug.cgi?id=753228

2015-08-12 15:14:50 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
	  Switching to GST_BUFFER_TIMESTAMP() to be consistent with other rtp code.

2015-08-12 14:59:53 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: prevent trying to get 0 bytes from adapter
	  This causes an assertion and would lead to getting a NULL instead
	  of a buffer. Without proper checking this would easily lead to a
	  segfault.
	  Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199

2015-07-29 17:29:28 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph265pay.c:
	  rtp: remove dead assignment
	  Value set to ret will be overwritten at least once at the end of the while
	  loop, removing assignment.

2015-04-24 16:48:23 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/rtp/gstrtph265pay.c:
	  remove unused enum items PROP_LAST
	  This were probably added to the enums due to cargo cult programming and are
	  unused.

2015-03-06 14:54:41 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/rtp/gstrtph265depay.c:
	  rtp: donl_present variable unused
	  donl_present is not implemented, yet the value is set and checked a few times.
	  Cleaning this.
	  CID #1249687

2015-01-08 15:36:04 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/rtp/gstrtph265pay.c:
	  rtp: value truncated too short creates dead code
	  type is truncated to 0-31 with "& 0x1f", but right after that it is checks if
	  the value is equivalent to GST_H265_NAL_VPS, GST_H265_NAL_SPS, and
	  GST_H265_NAL_PPS (which are 32, 33, and 34 respectively). Obviously, this will
	  never be True if the value is maximum 31 after the truncation.
	  The intention of the code was to truncate to 0-63.

2015-01-08 15:27:44 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/rtp/gstrtph265depay.c:
	  rtp: fix nal unit type check
	  After further investigation the previous commit is wrong. The code intended to
	  check if the type is 39 or the ranges 41-44 and 48-55. Just like gsth265parse.c
	  does. Type 40 would not be complete.

2015-01-08 13:47:09 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/rtp/gstrtph265depay.c:
	  rtp: fix dead code and check for impossible values
	  nal_type is the index for a GstH265NalUnitType enum. There are two types of dead
	  code here:
	  First, after checking if nal_type is >= 39 there are two OR conditionals that
	  check if the value is in ranges higher than that number, so if nal_type >= 39
	  falls in the True branch those other conditions aren't checked and if it falls
	  in the False branch and they are checked, they will always also be False. They
	  are redundant.
	  Second, the enum has a range of 0 to 40. So the checks for ranges higher than 41
	  should never be True.
	  Removing this redundant checks.
	  CID 1249684

2014-10-16 10:34:01 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	  rtp: add h265 RTP payloader + depayloader

2016-02-15 11:51:46 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* tests/check/elements/rtpmux.c:
	  tests: rtpmux: Fix element memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=762057

2016-02-12 20:57:29 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/monoscope/monoscope.c:
	  monoscope: rework the scaling code
	  The running average was wrong and the resulting scaling factor was only held in
	  place using the CLAMP. In addtion we are now convering quickly to volume
	  changes.
	  FInally now with this change, we can change the resolution defines and
	  everythign adjusts.

2016-01-28 17:00:55 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/monoscope/convolve.c:
	* gst/monoscope/monoscope.c:
	* gst/monoscope/monoscope.h:
	  monoscope: use constants in the drawing code
	  Make all the drawing ops be based on the constants. This way we can change
	  the fixed size at least at compile time.

2016-01-28 09:51:17 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/monoscope/gstmonoscope.c:
	  monoscope: replace hardcoded values by constants
	  This at least establishes the relationship.

2016-01-28 09:43:12 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/monoscope/convolve.c:
	* gst/monoscope/convolve.h:
	* gst/monoscope/monoscope.c:
	* gst/monoscope/monoscope.h:
	  monoscpe: make the convolver use dynamic memory
	  Replace all #defines with members and initialize the convolver with a parameter.

2016-01-28 08:56:44 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/monoscope/README:
	  monoscope: update README
	  We can already create multiple instances.

2016-01-28 08:53:35 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/monoscope/convolve.c:
	* gst/monoscope/monoscope.c:
	  monoscope: code cleanup
	  Use constants more often. Cleanup comments and add more to explain how things
	  work.

2016-02-09 12:14:04 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  glsyncmeta: separate out gpu/cpu waits.
	  CPU waits are more expensive and are only required if the CPU is ever going to
	  access the data. GPU waits perform inter-context synchronisation and are cheaper
	  as they don't require CPU intervention.

2016-02-08 23:41:32 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: remove check for impossible condition
	  Commit bd27a1f30b4458f2edee53c76dd07fb35904b61d added a few error handling
	  memory management checks. These check srccaps to see if it needs to be
	  unreferenced before returning, in the case of invalid_caps this goto jump
	  always happens before srccaps is set, so it will always be NULL in this
	  error label.
	  CID #1352035

2016-02-08 12:48:46 +0100  Piotr Drąg <piotrdrag@gmail.com>

	* po/POTFILES.in:
	  po: update POTFILES
	  https://bugzilla.gnome.org/show_bug.cgi?id=761705

2016-02-08 15:31:55 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix spelling of reenqueueing
	  To match commit 7d7074cef0272cd5155098bfc2bda6849dd89267. I love the idea
	  of aiming for the maximum number of consecutive vowels.

2016-02-08 10:17:49 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix spelling of queueing
	  Didn't know which one to choose between queuing and queueing, so I picked
	  the one with the biggest amount of vowels in a row ;-P (both are
	  acceptable apparently)

2016-02-07 15:02:35 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Don't pass the same data over and over
	  We already pass the entire frame to the decoder. If the decoder ask for
	  more data, don't pass the same data again as this leads to infinit loop.
	  Instead, simply fail the fill function to signal the problem with that
	  frame. It will then be skipped properly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761670

2016-02-08 00:10:33 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/lzo.c:
	  matroska: get rid of _stdint.h include

2016-02-05 20:00:57 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/Makefile.am:
	  tests: extend the AM_TESTS_ENVIRONMENT from check.mak
	  To get the CK_DEFAULT_TIMEOUT defined for all tests
	  https://bugzilla.gnome.org/show_bug.cgi?id=761472

2016-02-05 18:04:31 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From 86e4663 to b64f03f

2016-01-24 15:47:12 +0100  Holger Kaelberer <holger.k@elberer.de>

	* tests/examples/qt/qml/main.qml:
	  tests: fix warning in qml example
	  https://bugzilla.gnome.org/show_bug.cgi?id=756082

2016-01-30 18:43:30 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Skip APP and JPG markers and print warnings for unknown markers
	  For APP/JPG markers the size is following and we have to skip that. This is
	  not really a problem unless the marker contains e.g. a preview JPEG or
	  something else that we might interprete as another marker.

2016-01-26 22:37:30 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix framerate calculation for fragmented format
	  qtdemux calculates framerate using duration and the number of sample.
	  In case of fragmented mp4 format, however, the number of sample can
	  be figure out after parsing every moof box. Because qtdemux does not
	  parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect
	  framerate calculation.
	  This patch will triger gst_qtdemux_configure_stream() for every new moof.
	  Then, framerate will be calculated by using duration and n_samples of the moof.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760774

2016-01-28 22:36:23 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: handling zero segment-duration edit list
	  Based on document ISO_IEC_14496-12, edit list box can have
	  segment duration as zero. It does not imply that media_start equals to
	  media_stop. But, it just indicates a sample which should be presented
	  at the first. This patch derives segment duration using media_time
	  and duration of file. And set derived duration to segment-duration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760781

2016-01-28 21:36:54 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: expose streams with first moof for fragmented format
	  In case of push mode, qtdemux expose streams after got moov box.
	  We can not guarantee that a moov box has sample data such as sample duration
	  and the number of sample in stbl box for fragmented format case.
	  So, if a moov has no sample data, streams will not be exposed until get the first moof.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760779

2016-01-27 18:48:17 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Check for subset instead of non-empty intersection for ACCEPT_CAPS

2016-01-27 18:44:23 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Unset RECONFIGURE flag on srcpad whenever we configure new caps
	  Prevents double-negotiation during startup and in some other cases.

2016-01-27 16:43:22 +0100  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Add negotiation unit tests for all 4 modes
	  These now check the output caps based on the input caps and a following
	  capsfilter and make sure the caps are exactly as expected.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760995
	  https://bugzilla.gnome.org/show_bug.cgi?id=720388

2016-01-26 17:39:20 +0100  Vivia Nikolaidou <vivia@toolsonair.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Do passthrough in auto mode if downstream only supports interlaced
	  If the following conditions are met:
	  1) upstream and downstream caps are compatible
	  2) upstream is interlaced
	  3) downstream doesn't support progressive mode
	  then deinterlace will just do passthrough instead of failing to link.
	  This is done with the following scenario in mind:
	  videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
	  name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
	  queue ! deinterlace name=dein_desktop ! autovideosink
	  In this case, dein_src will do the deinterlacing. However,
	  videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
	  name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
	  queue ! deinterlace name=dein_desktop ! autovideosink t. ! queue !
	  "video/x-raw,interlace-mode=interleaved" ! fakesink
	  In this case, caps auto-negotiation will make dein_file and dein_desktop do
	  the deinterlacing, while dein_src will be passthrough.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760995

2016-01-26 18:05:51 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Add mode=auto-strict
	  In this mode we will passthrough all progressive caps but interlaced caps must be
	  caps where we actually support deinterlacing.
	  This is the only difference between auto and auto-strict, auto would
	  passthrough all unsupported interlaced caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720388

2016-01-26 17:50:30 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Implement reconfiguration a bit better
	  And e.g. consider reconfiguration caused by RECONFIGURE events too.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720388

2016-01-26 11:57:09 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Rewrite caps negotiation
	  Previously the result of the CAPS query and ACCEPT_CAPS depended on what kind
	  of caps were last set, and e.g. if we last had interlaced caps or not. That's
	  just broken.
	  Also previously the handling of non-sysmem caps features was rather random and
	  unusuable.
	  Now the behaviour is the following, depending on the mode property:
	  1) mode=disabled
	  Completely do passthrough of everything
	  2) mode=interlaced
	  Only accept formats we can actually deinterlace, and accept interlaced
	  and progressive content and always run the deinterlacer and output
	  progressive content
	  3) mode=auto (i.e. playbin)
	  Accept all progressive formats as passthrough, accept all formats that we
	  can deinterlace ourselves (which we do then), but also accept everything
	  else for which we then just passthrough. In auto mode, deinterlacing is best
	  effort: If we can, we deinterlace, if we can't we just output interlaced
	  content.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720388
	  https://bugzilla.gnome.org/show_bug.cgi?id=760553

2016-01-26 11:34:40 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Remove unused, obsolete bufferalloc code

2016-01-26 18:50:38 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: use A_AAC instead of A_AAC/MPEGx/y
	  Some GoogleCast compatible devices ignore A_AAC/MPEGx/y tracks; Also according to http://wiki.multimedia.cx/index.php?title=Matroska A_AAC/MPEGx/y is obsolete
	  https://bugzilla.gnome.org/show_bug.cgi?id=761144

2016-01-25 17:21:24 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/rtp/gstrtph261pay.c:
	  gst: Fix unintialized variable warnings
	  While cross-compiling with Linaro GCC 5.1-2015.08, it complained
	  about a couple unitialized variables.
	  This patch initializes them to zero.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761094

2016-01-25 16:29:46 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqtsink.cc:
	  qt: specify that we currently only take 2D textures
	  Fixes black screen video playback on android without a caps filter.

2016-01-25 15:03:23 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxsrc: print potentially negative offset with a sign

2016-01-21 17:41:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Re-add colorimetry field for RGB formats
	  This time, check if it's an RGB format and sets the transformation
	  matrix to identity. The rest of the colorimetry information is
	  meaningfull and shall be kept.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-22 10:03:50 +0100  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix sRGB colorspace definition
	  V4l2 can also use the sRGB colorspace for YUV formats and thus needs a
	  default matrix.

2016-01-21 15:29:46 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/debugutils/gsttaginject.c:
	  taginject: fix sample pipeline in docs
	  https://bugzilla.gnome.org/show_bug.cgi?id=679571

2016-01-21 10:49:44 +0100  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add adobe colorspace support
	  Use the new primaries and transfer function for Adobe RGB.
	  Explicitly list the colorimetry instead of using the default GStreamer
	  ones. The defaults for BT2020, for example, do not match.
	  Explicitly set the matrix of SRGB to RGB.

2016-01-20 13:41:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Ensure that we always have valid frame user data before using it
	  Otherwise we're going to dereference NULL pointers.

2016-01-20 10:02:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: Unref frame in all code paths of handle_frame()
	  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-19 22:49:20 +0100  Thibault Saunier <tsaunier@gnome.org>

	* ext/vpx/gstvpxenc.c:
	  vpxenc: Unref frame on ERROR
	  All code paths for handle_frame() must somehow take ownership of the frame, be
	  it by actually unreffing, forwarding the frame elsewhere or storing it for
	  later.
	  http://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-20 18:20:43 +1100  Jan Schmidt <jan@centricular.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2: Don't free props structure twice.
	  gst_v4l2_device_provider_probe_device() frees the passed props
	  structure, don't free it again in the caller.

2016-01-19 15:15:35 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Cleanup uneeded return statement

2016-01-19 15:14:59 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't set colorimetry for non YUV formats
	  Setting colormetry in caps for RGB have no meaning, but worst it
	  confuses the converters downstream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-19 13:01:17 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpchannels.c:
	* gst/rtp/gstrtpchannels.h:
	  rtp: fix compiler warnings with gcc-6
	  In file included from gstrtpL16depay.h:27:0,
	  from gstrtp.c:73:
	  gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable]
	  static const GstRTPChannelOrder channel_orders[] =

2016-01-19 14:57:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Don't play anything after the end of the data chunk even when seeking
	  Especially in push mode we would completely ignore the size of the data chunk
	  when not stop position is given for the seek. Instead make sure that the end
	  offset is at most the end of the data chunk if known.
	  Without this we would output anything after the data chunk, possibly causing
	  loud noises if the media file is followed by an INFO chunk or an ID3 tag.

2016-01-19 14:55:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Don't do calculations with -1 offsets when handling SEGMENT events
	  We use that to signal "infinity", taking the difference between that and some
	  other value is not going to give us any useful result for the end offsets of
	  segments.

2016-01-18 11:30:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling"
	  This reverts commit 271501f6576de4d141e7c2f618e28b9e3b1e5b38.
	  It wasn't meant to be pushed yet as the commit message indicates.

2016-01-12 14:01:21 -0800  Aleix Conchillo Flaqué <aconchillo@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
	  We check the stream profile and use the proper RTCP caps:
	  application/x-srtcp if we are using a secure profile and
	  application/x-rtcp otherwise.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760556

2016-01-05 16:15:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  WIP: rtpjitterbuffer: Add RFC7273 media clock handling

2016-01-15 11:36:35 +0000  Thibault Saunier <tsaunier@gnome.org>

	* ext/vpx/gstvpxenc.c:
	  vp8enc: Return FLOW_ERROR when an error accures
	  FALSE would mean FLOW_OK
	  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-08 22:19:06 +0300  Sergey Borovkov <serge.borovkov@gmail.com>

	* ext/qt/qtitem.cc:
	  qml: Mark material dirty when texture buffer is updated
	  Qt might not redraw the scene otherwise.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758286

2016-01-15 03:57:45 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: break as soon as the device is found
	  No need to loop further if there's no side-effects for it

2016-01-15 03:56:49 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: Fix error handling when selecting/opening devices
	  Post an element error when the CoreAudio device cannot be selected or opened.
	  Also ensure that we post a GST_ERROR with more detail.

2016-01-13 23:40:20 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: When flushing on EOS, don't process more data than the "data" size
	  Even if we have more data queued up when flushing than the size of the data
	  chunk, don't process and output it. If the data size is known, this likely
	  contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just
	  outputting them as if they were data is going to cause unexpected behaviour
	  and unpleasant audio noises.

2014-08-29 15:40:23 +0200  Antonio Ospite <ao2@ao2.it>

	* tests/check/pipelines/wavenc.c:
	  tests: fix a thinko in the wavenc example
	  The code is supposed to follow somehow what the comment above says, that
	  is to have one channel with a wave of freq 440 and the other channel
	  with a wave of freq 880, but an off by one error results in frequencies
	  of 0 and 440.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735673

2014-08-29 15:07:58 +0200  Antonio Ospite <ao2@ao2.it>

	* gst/interleave/interleave.c:
	  interleave: Fix the example by setting channel-masks in the sink pads
	  The current example does not work, it fails with:
	  ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error.
	  gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0:
	  streaming task paused, reason not-negotiated (-4)
	  This is because negotiation with wavenc gets messed up by the missing
	  channel positions configuration.
	  The proper way to define the channel layout when using the interleave
	  element in code would be to set the channel-positions property, but
	  gst-launch-1.0 does not know how to deal with arrays; so the example
	  pipeline works around the issue by setting the channel-masks in the sink
	  pads.
	  Also fix a repetition in the deinterleave example description
	  https://bugzilla.gnome.org/show_bug.cgi?id=735673

2016-01-11 16:29:55 +0000  Tim Sheridan <tim.sheridan@imgtec.com>

	* gst/audioparsers/gstsbcparse.c:
	  sbcparse: Fix frame length calculation
	  SBC frame length calculation wasn't being rounded up to the nearest byte
	  (as specified in the A2DP 1.0 specification, section 12.9). This could
	  cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly
	  calculated frame lengths.
	  Incorrect frame length calculation causes frame coalescing to fail, as
	  subsequent frames in the stream aren't found in the expected locations.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742446

2016-01-10 22:54:12 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: demote warning on wrong reserved value to fixme
	  We are likely just parsing a backward-compatible stream we
	  don't fully support.

2016-01-08 16:27:05 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: simplify caps selection
	  The downstream caps query with a filter alraedy gives us the possible
	  intersection so there is no need to check it again with downstream
	  if it is supported. Just try to set it directly.

2016-01-07 20:42:41 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: fix unnecessary sub-buffer creation
	  We create a sub-buffer just to copy over its metas and then
	  throw it away immediately, just use the original input buffer
	  directly.

2016-01-07 20:38:27 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpdvdepay.c:
	  rtpdvdepay: fix unnecessary sub-buffer creation
	  We create a sub-buffer just to copy over its metas and then
	  throw it away immediately, just use the original input buffer
	  directly.

2016-01-07 20:34:05 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpamrdepay.c:
	  rtpamrdepay: fix unnecessary sub-buffer creation
	  We create a sub-buffer just to copy over its metas and then
	  throw it away immediately, just use the original input buffer
	  directly.

2016-01-07 20:27:29 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtpvrawdepay: fix major memory leak and performance issue
	  We call gst_rtp_buffer_get_payload() which creates a sub-buffer
	  of each input buffer, just to copy over metas, and then leak it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760289

2016-01-08 15:32:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rganalysis.c:
	  rganalysis: Fix compiler warnings in the unit test
	  elements/rganalysis.c:919:66: error: shifting a negative signed value is undefined
	  [-Werror,-Wshift-negative-value]
	  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0));
	  ~~ ^
	  elements/rganalysis.c:929:69: error: shifting a negative signed value is undefined
	  [-Werror,-Wshift-negative-value]
	  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14));
	  ~~ ^
	  elements/rganalysis.c:939:64: error: shifting a negative signed value is undefined
	  [-Werror,-Wshift-negative-value]
	  push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14));
	  ~~ ^

2016-01-05 18:13:06 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: don't map buffer multiple times when parsing

2016-01-07 18:20:30 +0200  Steven Hoving <sh@bigbrother.nl>

	* gst/matroska/matroska-read-common.c:
	  matroska: Store subtitle stream count in the correct variable
	  And don't override the video stream count instead.

2016-01-05 18:59:06 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/equalizer/gstiirequalizernbands.c:
	  equalizer: The child-proxy API is GObject based in 1.x
	  Not GstObject anymore.

2015-05-21 17:41:12 +0200  Pablo Anton <pablo.anton@vodalys-labs.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2-*: Configuring output pool correctly for using drivers min_buffer if present.
	  Signed-off-by: Pablo Anton <pablo.anton@vodalys-labs.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=755736

2015-12-31 15:46:31 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: add debug msg on CRC mismatch while validating frame header

2015-12-31 16:00:49 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: drop unneeded braces at _parse_frame() exit
	  Additionally, drop redundant comment & line break

2015-12-31 15:55:18 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: minor grammar correction

2015-12-31 15:34:57 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: update URLs on pointers to online spec

2015-12-31 14:40:15 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: make buffer DTS setting explicitly unconditional
	  We are setting it to PTS regardless of block_strategy

2015-12-31 14:21:40 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: add actual invalid block type to warning
	  For someone that read the spec is clear the only *invalid*
	  data block type is 127. For the rest, its useful information.
	  Additionally. values 7-126 are currently reserved by the
	  spec so the situation might change in the future.

2015-12-31 14:12:36 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: use shift instead of mask & comp
	  We are only interested on the first bit of the first
	  byte of the metadata block header to figure out whether
	  is marked as the last one. The shift makes it quite
	  clearer.

2015-12-31 12:52:13 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: warn on wishful parsing of weird headers
	  If we get anything from 7 to 126 as type when parsing
	  a metadata block header, we are likely dealing with a
	  FLAC stream version we don't fully understand. Issue
	  a warning if so.
	  Document function assumptions regarding the passed-on
	  type while at this.

2015-12-31 11:33:45 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: show meaningful info on frame CRC check
	  As CRCs are calculated for the comparition already, we
	  might as well (cheaply) inform the user how the numbers
	  differ if a missmatched pair is found.
	  While at it:
	  Rephrase candidate-frame message to make more sense

2015-12-31 02:40:43 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: drop remaining trailing whitespace

2015-12-31 02:15:06 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: drop superflous else clauses

2015-12-31 01:09:51 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: factor out buffer time and offset resetting
	  Avoids multiple occurrences of the same resetting pattern

2015-12-31 00:54:48 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: move block handling by type out of _parse_frame()

2015-10-07 18:51:25 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: replace duplicated codes to call new base sdp apis
	  https://bugzilla.gnome.org/show_bug.cgi?id=745880

2015-12-30 12:16:56 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: drop redundant return statement on _header_is_valid()
	  Fix the rather vague error message while at it.

2015-12-30 01:56:26 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: rework gst_flac_parse_frame_is_valid()
	  drop unnecessary nesting looking for end of frame

2015-12-30 00:37:04 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: factor out context clearing routine

2015-12-29 18:05:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Guard against no codec data in prores caps creation
	  CID 1346532

2015-12-29 17:58:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: Initialize buffer variable to NULL
	  False positive but trivial to fix and possibly causing compiler warnings at
	  some point in the future too.
	  CID 1346535

2015-07-27 15:53:26 +0200  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2deviceprovider: add properties to the device
	  Add properties to the device with exactly the same keys and sematics
	  as what pulseaudio uses as property keys.
	  Also handle the case when a device is probed manually and not through gudev.
	  https://bugzilla.gnome.org//show_bug.cgi?id=759780

2015-12-25 11:41:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Free the various buffers in GstBaseTransform::stop()
	  Previously we leaked them completely, but as they're specific to the caps
	  freeing them in stop() instead of finalize() makes most sense.

2015-12-24 15:28:06 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.7.1 ===

2015-12-24 14:16:21 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.7.1

2015-12-24 13:19:24 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2015-12-24 12:22:32 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/cs.po:
	* po/de.po:
	* po/el.po:
	* po/hu.po:
	* po/nb.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: Update translations

2015-12-21 09:57:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: drop flushes from our own offset seek
	  Prevents downstream from receiving flushes for a seek only in
	  upstream. Those seeks are only to start reading from the right
	  offset when skipping or returning to qt atoms.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758928

2015-11-11 16:53:19 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Always set the channel mask for PCM streams
	  Just use the gst_audio_channel_get_fallback_mask function for now as
	  the specification is too complicated and nobody implements it.

2015-12-21 11:37:26 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix sleep for buffer-time lower than 200000
	  https://bugzilla.gnome.org/show_bug.cgi?id=748680

2015-12-21 12:31:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  configure: Use -Bsymbolic-functions if available
	  While this is more useful for libraries, some of our plugins with multiple
	  files and some internal API can also benefit from this.

2015-12-18 15:34:52 +0000  William Manley <will@williammanley.net>

	* gst/debugutils/progressreport.c:
	* gst/debugutils/progressreport.h:
	  progressreport: add support for using format=buffers with do-query=false
	  This is useful for investigating and debugging pipelines which are
	  producing buffers at a slower/faster rate than you would expect.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759635

2015-12-18 15:49:43 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Update formats table
	  This change add all the new RGB based format. Those format removes the
	  ambiguity with the ALPHA channel. Some other missing multiplanar format
	  has been added with some additional cleanup.

2015-12-18 05:17:15 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't write invalid edit list start time.
	  Avoid writing a negative number as a large positive
	  integer in an edit list when the first_ts is smaller
	  than the first_dts - which can happen when the first
	  packet received has a PTS but no DTS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759615

2015-12-04 23:16:45 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Only update running time when it increases.
	  Don't increment running time from every buffer. The correct
	  logic to only increment when running time advances is a
	  little further down, so delete this left-over line.

2015-11-18 11:01:20 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Implement prores support
	  https://bugzilla.gnome.org/show_bug.cgi?id=758258

2015-11-18 16:20:38 +1100  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroska-demux: Play ProRes video streams
	  Generate video/x-prores caps for ProRes video streams.
	  Every frame needs an 8 byte header prepended, as described in
	  http://wiki.multimedia.cx/index.php?title=Apple_ProRes#Frame_layout
	  so do that in a post-processing callback.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758258

2015-12-18 10:18:09 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* ext/dv/gstdvdec.h:
	  dvdec: Remove unused fields
	  Remove unused fields frame_len and space
	  https://bugzilla.gnome.org/show_bug.cgi?id=759614

2015-12-17 16:03:04 +0100  Vincent Dehors <vincent.dehors@openwide.fr>

	* gst/rtp/gstrtpj2kdepay.c:
	  rtpj2kdepay: Push one JPEG2000 frame per buffer, not a buffer list with multiple buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=758943

2015-12-16 11:43:58 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	  dv1394: log error if failed to set socket status flag
	  Log an error message if failed to set write or read socket as
	  non-blocking.
	  CID 1139608
	  CID 1139609

2015-12-15 17:10:00 +0000  Dave Craig <davecraig@unbalancedaudio.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: Check for NULL return value of gst_pad_get_current_caps()
	  https://bugzilla.gnome.org/show_bug.cgi?id=759503

2015-12-16 09:35:53 +0100  Sebastian Dröge <sebastian@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update to git

2015-12-15 19:28:05 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/qt/Makefile.am:
	  qtsink: Add configured GL cflags to the build
	  We don't directly link to GL in the element, though we use GL headers.
	  For this reason we need to include the proper GL headers path. This
	  prevent this element from using a different GL header then libgstgl.

2015-12-15 14:27:22 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/Makefile.am:
	  vpx: Add missing headers in Makefile.am
	  This fixes distcheck.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755510

2015-09-24 12:57:00 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* ext/vpx/Makefile.am:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	* ext/vpx/gstvpxenc.c:
	* ext/vpx/gstvpxenc.h:
	  vpx: created common baseclass GstVPXEnc
	  GstVP8Enc and GstVP9Enc has almost 80% code in common.
	  created common baseclass GstVPXEnc for GstVP8Enc and GstVP9Enc
	  https://bugzilla.gnome.org/show_bug.cgi?id=755510

2015-12-15 12:57:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxdec.h:
	  vpxdec: Remove unneeded add video_meta
	  This also remove copies for VP8, which was not correctly in place
	  in previous related patch.

2015-12-15 09:49:24 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* ext/vpx/Makefile.am:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9dec.h:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxdec.h:
	  vpx: created common base class GstVPXdec for vpx decoders
	  Base class for the vp8dec and vp9dec.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755510

2015-12-14 11:09:46 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audiofx/gststereo.c:
	  plugins-bad: Fix example pipelines
	  rename gst-launch --> gst-launch-1.0
	  replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
	  fix caps in examples
	  https://bugzilla.gnome.org/show_bug.cgi?id=759432

2015-06-10 09:17:08 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* configure.ac:
	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Add GTlsInteraction property
	  https://bugzilla.gnome.org/show_bug.cgi?id=750709

2015-12-14 09:05:06 -0500  Evan Callaway <evan.callaway@ipconfigure.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Retry connection if tunneling needs authentication
	  Leverage response from gst_rtsp_connection_connect_with_response to
	  determine if the connection should be retried using authentication.  If
	  so, add the appropriate authentication headers based upon the response
	  and retry the connection.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749596

2015-12-14 14:19:05 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: check port-range format
	  The string could exist but with a wrong format, in that case we still want
	  to reset the values of client_port_range.min and max like we do if there is
	  no string.
	  CID 1139593

2015-12-14 14:55:12 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Check device property and fail if device can't be found
	  Don't use default if a specific device is set but it can't be found.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759452

2015-12-14 14:15:00 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix handling of the mute property
	  - set mute value at startup
	  - correct set and get mute functions
	  https://bugzilla.gnome.org/show_bug.cgi?id=755106

2015-12-14 13:43:59 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqsgtexture.cc:
	  glmemory: base classify and add the pbo memory on top
	  The base class is useful for having multiple backing memory types other
	  than the default.  e.g. IOSurface, EGLImage, dmabuf?
	  The PBO transfer logic is now inside GstGLMemoryPBO which uses GstGLBuffer
	  to manage the PBO memory.
	  This also moves the format utility functions into their own file.

2015-12-11 11:23:13 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Check the return value of GetStatus() too to decide if there was an error
	  If GetStatus() fails, the status itself won't be very meaningful but we also
	  have to look at its return value. This fixes blocking pipelines when removing
	  sound devices or during other errors, where we wouldn't notice the error and
	  then wait forever.
	  https://bugzilla.gnome.org/show_bug.cgi?id=734098

2015-12-10 17:41:46 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  isomp4: remove unused parameters in build_*_extension
	  AtomTRAK parameter is not used by build_mov_alac_extension(),
	  build_jp2h_extension(), or build_mov_alac_extension()  and can be
	  removed.

2015-12-10 15:11:07 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: replace variable only used once
	  Replace has_shift variable with value since it is only use once.

2015-12-09 12:24:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix packet dropping after a big discont
	  We would queue 5 consective packets before considering a reset and a proper
	  discont here. Instead of expecting the next output packet to have the current
	  seqnum (i.e. the fifth), expect it to have the first seqnum. Otherwise we're
	  going to drop all queued up packets.

2015-12-09 11:49:02 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/interleave/interleave.h:
	  interleave: Remove unsed field
	  Remove unused field collect_event in interleave.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759226

2015-12-07 16:33:14 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Stop pushing data as soon as possible in push-mode
	  When working in push-mode, we attempt to push out everything currently
	  buffered in the adapter.
	  This has two pitfalls:
	  * We could stop earlier (the moment we get a non-ok or non-not-linked)
	  * We return the last combined flow return, which might be completely
	  different from the previous combined flow return

2015-12-07 09:08:09 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From b319909 to 86e4663

2015-12-07 14:41:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Add a warning if an empty RTCP packet is tried to be sent
	  https://bugzilla.gnome.org/show_bug.cgi?id=759119

2015-11-30 19:20:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9dec.h:
	  vpxdec: Use GstMemory to avoid copies
	  With the VPX decoders it's not simple to use downstream buffer pool,
	  because we don't know the image size and alignment when buffers get
	  allocated. We can though use GstAllocator (for downstream, or the system
	  allocator) to avoid a copy before pushing if downstream supports
	  GstVideoMeta. This would still cause a copy for sink that requires
	  specialized memory and does not have a GstAllocator for that, though
	  it will greatly improve performance for sink like glimagesink and
	  cluttersink. To avoid allocating for every buffer, we also use a
	  internal buffer pool.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745372

2015-11-30 08:42:35 +0100  Edward Hervey <edward@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Avoid over-skipping when checking LOAS config
	  There might be multiple LOAS config in a row in a full frame. The first
	  one might be a multi-layer config (which we can't properly parse yet)...
	  but then followed by a valid (single-layer) one.
	  The code was previously skipping whole frames (instead of just the LOAS
	  config we failed to read) resulting in multiple frames (seen up to 6s in
	  some situation) being dropped before finally getting the configuration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758826

2015-11-25 17:08:56 +0100  Edward Hervey <edward@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Properly set SPARSE stream flags for subpicture/subtitle
	  And while we're at it, also detect 'DXSA' as being a variant fourcc
	  of 'DXSB' for XSUB

2015-11-30 21:23:52 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: grammar fix

2015-11-30 21:01:17 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: switch shoutcast stream provider
	  Fixes failing ICY test. Previous provider has
	  streaming disabled outside UK.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758114

2015-11-18 16:10:11 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/avi/gstavimux.c:
	  avimux: don't crash if we never got audio caps before stopping
	  auds.blockalign is set once the first caps arrive. If
	  gst_avi_mux_stop_file() is called before this happens then auds.blockalign
	  is zero and gst_avi_mux_audsink_set_fields() cause a crash:
	  [...]
	  avipad->parent.hdr.rate = avipad->auds.av_bps / avipad->auds.blockalign;
	  [...]
	  https://bugzilla.gnome.org/show_bug.cgi?id=758912

2015-12-01 18:20:23 +0100  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: don't block when resurecting a buffer
	  When we are resurecting a buffer, don't block. instead let us copy a
	  buffer.

2015-12-01 00:30:08 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: remove extra variable to improve readability
	  Makes it easier to see that the event is being replaced/unrefed

2015-12-01 00:22:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: respect seqnum in seek events
	  Propagate the original seek seqnum to events originated from
	  seeking to make sure they have the same value

2015-12-01 00:03:21 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: flush upstream when seeking in pull mode
	  Makes sure upstream will unblock and return the thread so that
	  seeking can continue
	  https://bugzilla.gnome.org/show_bug.cgi?id=758861

2015-11-27 09:27:29 +0100  Anton Bondarenko <antonbo@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: add "send SPS/PPS with every key frame" mode
	  It's not enough to have timeout or event based SPS/PPS information sent
	  in RTP packets. There are some scenarios when key frames may appear
	  more frequently than once a second, in which case the minimum timeout
	  for "config-interval" of 1 second for sending SPS/PPS is not sufficient.
	  It might also be desirable in general to make sure the SPS/PPS is
	  available with every keyframe (packet loss aside), so receivers can
	  actually pick up decoding immediately from the first keyframe if
	  SPS/PPS is not signaled out of band.
	  This patch adds the possibility to send SPS/PPS with every key frame. This
	  mode can be enabled by setting "config-interval" property to -1. In this
	  case the payloader will add SPS and PPS before every key (IDR) frame.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757892

2015-11-27 09:03:51 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtph264pay: change config-interval property type from uint to int
	  This way we can use -1 as special value, which is nicer than MAXUINT.
	  This is backwards compatible even with the GValue API, as shown by
	  a unit test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757892

2015-11-26 21:46:11 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for Opus
	  Add support for demuxing Opus encapsulated in MP4 files, based on the
	  following spec: https://www.opus-codec.org/docs/opus_in_isobmff.html
	  https://bugzilla.gnome.org/show_bug.cgi?id=742643

2015-11-25 22:48:32 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: use macro for codec_name
	  Use _codec() macro instead of duplicating code.

2015-03-25 16:32:55 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: videodec: choose format from caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=733827

2015-03-27 15:02:33 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: add gst_v4l2_object_probe_caps
	  Add a variant of gst_v4l2_object_get_caps that bypasses the probed_caps cache.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733827

2015-11-19 17:20:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2.c:
	  v4l2-probe: Skip devices without supported formats

2015-11-13 12:35:59 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	* sys/v4l2/gstv4l2.c:
	  v4l2: Track /dev/video* to triggered required probe
	  If something in /dev/video* get added, removed or replaced, we need to
	  probe the devices again in order to ensure the dynamic devices are up to
	  date.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758085

2015-11-25 14:51:40 +1100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: rtpsession: don't send empty RTCP packets
	  generate_rtcp can produce empty packets when reduced size RTCP is turned on.
	  Skip them since it doesn't make sense to push them and they cause errors with
	  elements that expect RTCP packets to contain data (like srtpenc).

2015-11-24 10:57:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: restore the segment on case of soft reset
	  When seeking back to restore the mdat position a flush is pushed
	  through and it resets downstream segment information. Make sure
	  that after the flush (that does a soft reset) a segment will
	  be pushed again
	  Fixes regressions spotted at
	  https://ci.gstreamer.net/job/GStreamer-master-validate/2100/

2015-11-20 12:44:22 +0000  Graham Leggett <minfrin@sharp.fm>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: fix spelling of variable
	  https://bugzilla.gnome.org/show_bug.cgi?id=758390

2015-11-20 11:05:51 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: unite duplicate FourCC
	  Unite in fourcc.h the FourCCs that are used twice or more in qtdemux

2015-11-20 11:18:43 +1100  Roman Nowicki <rnowicki@sims.pl>

	* ext/qt/qtitem.cc:
	  qml: reuse existing GstQSGTexture
	  Fixes a memory leak leaking the texture objects.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758286

2015-11-20 11:08:37 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqsgtexture.cc:
	  qml: activate the wrapped context when binding
	  Mitigates the following critical
	  gst_gl_context_thread_add: assertion 'context->priv->active_thread == g_thread_self ()' failed

2015-11-19 11:55:19 +0100  Roman Nowicki <rnowicki@sims.pl>

	* ext/qt/qtitem.cc:
	  qml: proper initialization if scene is already initialized
	  The scene graph can be initialized when the we receive window handle change
	  notification and so we will not receive a scenegraph initialization
	  notification.  Initialize ourself in this case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758337

2015-11-19 15:33:45 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Fix capture/output-io-mode properties
	  There was some miss-match in the implementation. This makes it
	  concistent, though functionally it worked, except the video decoder
	  output-io-mode getter.

2015-11-19 19:48:06 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  atoms: remove unused argument of build_mov_wave_extension()
	  AtomTrak * trak argument of build_move_wave_extension() isn't used.
	  Removing it.

2015-11-19 19:28:20 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: remove duplicate FourCC
	  Use the available FourCCs in fourcc.h instead of duplicating them.

2015-11-19 18:36:39 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	  isomp4: centralize all FourCC
	  10 FourCCs generated with GST_MAKE_FOURCC() in gstqtmux.c and atoms.c
	  already exist in fourcc.h. Don't duplicate these and use them directly.
	  Plus moving 6 to fourcc.h, to centralize them all.

2015-11-19 17:32:12 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/matroska/webm-mux.c:
	  matroska/webmmux: fix outdated example launch lines
	  Update gst-launch-0.10 lines to gst-launch-1.0

2015-11-16 13:26:50 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  isomp4: add support for Opus in mp4mpux
	  Add support for muxing MP4 files containing Opus. Based on the spec
	  detailed here:
	  https://www.opus-codec.org/docs/opus_in_isobmff.html
	  https://bugzilla.gnome.org/show_bug.cgi?id=742643

2015-11-17 15:23:17 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* tests/examples/gtk/glliveshader.c:
	  Remove unnecessary NULL checks before g_free()
	  g_free() is NULL-safe

2015-11-18 19:10:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Replace tabs with spaces

2015-11-18 19:07:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Cast to signed integers to prevent unsigned compare between negative and positive numbers
	  This fixes seeking if the first entries in the samples table are negative. The
	  binary search would always fail on this as the array would not be sorted if
	  interpreting the negative numbers as huge positive numbers. This caused us to
	  always output buffers from the beginning after a seek instead of close to the
	  seek position.
	  Also add a case to the comparison function for equality.

2015-11-18 16:01:48 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: remove duplicate check
	  We want 1 or 2 streamheaders, the check  if (bufarr->len != 1 &&
	  bufarr->len != 2) is enough. Not need to check if bufarr->len is <= 0 or
	  > 255.

2015-11-18 14:48:36 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Fix error leak and handle error
	  g_thread_try_new allows for possiblity of failures. In case it fails,
	  error is not handled and leaked.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758260

2015-11-15 17:16:29 -0800  Josep Torra <n770galaxy@gmail.com>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: Properly handle backward compat for event deserialization
	  Actual code is checking for a NULL terminator and a ';' terminator,
	  for backward compat, in a chained way that cause all events being rejected.
	  The proper condition is to reject the events when terminator isn't
	  in ['\0', ';'] set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758151

2015-11-15 17:11:02 -0800  Josep Torra <n770galaxy@gmail.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: Test for handling of custom events in rtpgst
	  Add a simple test that checks proper serialization/deserialization
	  of custom events with rtpgstpay and rtpgstdepay.

2015-11-16 16:23:43 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  vpxdec: Use threads on multi-core systems
	  This adds an automatic mode to the threads property of vpxdec in order to
	  use as many threads as there is CPU on the platform. This brings back
	  GStreamer VPX decoding performance closer to what is achieved by other
	  players, including Chromium.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758195

2015-11-16 10:58:32 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only send initial gaps for non-fragmented streams
	  It would be unusual to have the header segment with an 'edts' atom
	  indicating gaps at the beginning when handling fragmented streams.
	  The header usually doesn't contain any timestamping information, this
	  should come from the playlist/manifest and the segments with media
	  in those scenarios.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758171

2015-11-17 09:41:34 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  Revert "Revert "qtdemux: respect qt segments in push-mode for empty starts""
	  This reverts commit d842ff288a9d01214a046becbfd9cbff3a4acea0.
	  This was reverted by accident

2015-11-17 12:39:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: Add "loop" property for enabling/disabling multicast loopback
	  On POSIX, IP_MULTICAST_LOOP is a setting for the sender socket. On Windows it
	  is a setting for the receiver socket. As such we will need it on udpsrc too to
	  allow filtering out our own multicast packets.

2015-11-16 13:52:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  Revert "qtdemux: respect qt segments in push-mode for empty starts"
	  This reverts commit 142d8e2d23e5602e7382977af1043d621625f8c8.

2015-11-16 16:56:04 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix string memory leak
	  The string got using g_strdup_printf will be allocated memory
	  and should be freed after use.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758161

2015-11-14 21:51:11 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2/object: remove unnecessary NULL check before g_free()

2015-11-14 21:45:29 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/oss/gstosssrc.c:
	  osssrc: remove unnecessary NULL check before g_free()

2015-11-14 21:43:24 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/sunaudio/gstsunaudiosrc.c:
	  sunaudiosrc: remove unnecessary NULL checks before g_free()

2015-11-14 21:36:30 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: remove unnecessary NULL checks before g_free()

2015-11-14 21:31:08 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: remove unnecessary NULL checks before g_free()

2015-11-14 21:26:21 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/matroska/matroska-read-common.c:
	  matroska/read-common: remove unnecessary NULL checks before g_free()

2015-11-14 20:43:10 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  isomp4/atoms: remove unnecessary NULL checks before g_free()

2015-11-14 20:35:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtp/gstrtptheorapay.c:
	  rtp/theorapay: remove unnecessary NULL checks before g_free()

2015-11-14 20:33:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtp/vorbispay: remove unnecessary NULL checks before g_free()

2015-11-14 20:31:34 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtp/jpegpay: remove unnecessary NULL checks before g_free()

2015-11-14 20:27:04 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: remove unnecessary NULL checks before g_free()

2015-11-14 20:22:09 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove unnecessary NULL checks before g_free()

2015-11-14 20:14:25 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/flx/gstflxdec.c:
	  flxdec: remove unnecessary NULL check before g_free()

2015-11-14 20:09:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gstop.c:
	  effectv/optv: remove unnecessary NULL checks before g_free()

2015-11-14 20:05:03 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gstshagadelic.c:
	  effectv/shagadelictv: remove unnecessary NULL checks before g_free()

2015-11-14 20:01:43 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gstripple.c:
	  effectv/ripple: remove unnecessary NULL checks before g_free()

2015-11-14 19:56:57 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gstradioac.c:
	  effectv/radioac: remove unnecessary NULL checks before g_free()

2015-11-14 19:52:12 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gststreak.c:
	  effectv/streak: remove unnecessary NULL check before g_free()

2015-11-14 17:04:55 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/shout2/gstshout2.c:
	  shout2: remove unnecessary NULL checks before g_free()

2015-11-14 16:57:13 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/vpx/gstvp9enc.c:
	  vp9enc: remove unnecessary NULL check before g_free()

2015-11-14 16:54:42 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: remove unnecessary NULL check before g_free()

2015-11-14 16:20:33 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: remove unnecessary NULL checks before g_free()

2015-11-13 13:34:02 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: add support of NV16, NV61 and NV24 formats
	  Mapped respectively to V4L2_PIX_FMT_NV16/V4L2_PIX_FMT_NV16M,
	  V4L2_PIX_FMT_NV61,V4L2_PIX_FMT_NV61M and V4L2_PIX_FMT_NV24 v4l2 formats.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758058

2015-11-11 14:10:53 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxpartreader: Fix GCond leak
	  inactive_cond is not being cleared resulting in memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757924

2015-08-06 12:44:20 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix output state memory leak
	  When jpeg_finish_decompress is called, output state reference is being created.
	  But if there is any failures in finishing decompress, it jumps to setjmp,
	  and at that point state was not referenced. Resulting in leak of output state.
	  Hence adding another setjmp after output state is referenced.
	  Similarly adding another setjmp to unmap the frame in case error happens before
	  finish_decompress
	  https://bugzilla.gnome.org/show_bug.cgi?id=753087

2015-11-10 12:32:39 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	  gtk: add the overlaycomposition feature to the template caps
	  There is a possibility that the _get_caps impl will be called with the
	  feature in the filter caps which when interecting with the template,
	  will return EMPTY and therefore fail negotiation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757854

2015-08-10 11:23:45 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: respect qt segments in push-mode for empty starts
	  In push-mode it is hard to support qt segments overall but it is
	  possible to support when the file isn't heavily edited but just contain
	  a segment to indicate a gap at the beginning. This also allows properly
	  timestamping data that has negative DTS in push-mode.
	  It is relevant to support those for 2 scenarios:
	  1) fragmented streaming
	  2) HTTP playback of 'regular' mp4
	  https://bugzilla.gnome.org/show_bug.cgi?id=753484

2015-11-05 18:39:33 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/pulse/pulsedeviceprovider.c:
	  pulse: Don't leak caps and structures in the device provider

2015-11-04 19:01:20 +0530  Arun Raghavan <arun@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: Document properties that are expressed in bits per second
	  This changed in 928cd110bcea5d143cab3ea747991851d52ecbad and
	  73c0c2920f9aca96982a4de0c20b3417aa148b81 but was not documented.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747863

2015-11-04 18:51:32 +0530  Arun Raghavan <arun@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: Trivial gst-indent fixes

2015-08-12 13:35:40 +0200  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: support for cenc auxiliary info parsing outside of moof box
	  When the cenc aux info index is out of moof boundaries, keep track of
	  it and parse the beginning of the mdat box, before the first sample.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755614

2015-11-03 20:33:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Use codecutils helpers for creating Opus caps
	  Also fix up codec data with values from the container.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-03 14:51:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: There is no multistream field for Opus anymore
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-03 12:42:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/webm-mux.c:
	  matroska/webmmux: Support Opus in webmmux and VP9 in matroskamux
	  https://bugzilla.gnome.org/show_bug.cgi?id=729950

2015-11-03 12:40:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Parse and handle CodecDelay, SeekPreroll and DiscardPadding
	  https://bugzilla.gnome.org/show_bug.cgi?id=727305

2015-11-03 12:18:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: Write CodecDelay, DiscardPadding and SeekPreroll for Opus
	  And also adjust timestamps and durations according to the codec delay, both
	  should include it for whatever reason.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727305

2015-11-03 11:49:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Opus headers are not in-band
	  https://bugzilla.gnome.org/show_bug.cgi?id=727305

2015-11-03 22:01:07 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/v4l2/gstv4l2.c:
	  v4l2: Set O_CLOEXEC on the device fd
	  This is needed to make sure that child processes don't inherit the video
	  device fd which can cause problems with some drivers.

2015-11-03 14:46:30 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpmanager: switch G_GINT64_FORMAT for GST_STIME_ARGS
	  No need to use G_GINT64_FORMAT for potentially negative values of
	  GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
	  Plus it creates more readable values in the logs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-03 14:26:29 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: use GST_STIME_ARGS for GstClockTimeDiff
	  No need to manually handle negative values of diff, GST_STIME_ARGS does
	  exactly this.

2015-11-02 16:53:15 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: use GST_STIME_ARGS for GstClockTimeDiff
	  No need to manually handle negative values of diff, GST_STIME_ARGS does
	  exactly this.

2015-11-02 16:43:46 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: use GST_STIME_ARGS for GstClockTimeDiff
	  No need to manually handle negative values of diff, GST_STIME_ARGS is
	  available for this.

2015-10-30 10:05:37 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audiochebband.c:
	  audiochebband: Fix typo in example pipeline
	  Fix typo in example pipeline.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757340

2015-10-28 23:47:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2: fix double-unref in the v4l2 device provider

2015-10-27 10:48:00 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-ids.c:
	  matroskamux: don't drop JPEG frames that only have PTS but no DTS set
	  For the MS/VfW codec ids, we want to write DTS timestamps instead
	  of PTS because that's what everyone else seems to do (and it's also
	  how it is in AVI). So for those input formats we use the buffer DTS
	  instead of the PTS. However, if there's no DTS set but only the PTS
	  then just take the PTS instead of dropping the input buffer. This
	  is useful especially for I-frame only codecs like JPEG and huffyuv,
	  but should also be fine as fallback in general.
	  Fixes regression with input JPEG frames that only have PTS set on them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756967

2015-10-24 23:57:38 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/splitmux.c:
	  tests/check/splitmux: test that the release_pad vfunc of splitmuxsink actually releases pads
	  https://bugzilla.gnome.org/show_bug.cgi?id=753622

2015-10-24 23:57:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: do not destroy the multiqueue & muxer when going to NULL
	  Instead, delay it until all request pads have been released. This is
	  because the release_pad() vfunc requires the multiqueue and muxer to
	  be there in order to release their request pads as well. If those
	  elements are destroyed earlier, release_pad() does not work, no
	  pads are released and some resources are leaked.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753622

2015-10-20 15:28:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Read buffer timestamp *after* actually setting it
	  https://bugzilla.gnome.org/show_bug.cgi?id=756809

2015-10-24 17:14:07 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempo.h:
	  scaletempo: Fix handling of rate < 0
	  We have to reverse all samples in a buffer before processing them to properly
	  have continuous data from one buffer to another. As a result we will have a
	  negative applied rate and a rate of 1.0.
	  Also make sure that input buffers are correctly clipped to the segment,
	  otherwise our calculations are going to go wrong.
	  Also copy over the segment event's sequence number to the output segment while
	  we're at it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757033

2015-10-19 18:04:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: break as soon as non-interlaced if found
	  It looks for a non-interlaced entry on the filter caps, break
	  as soon as one is found to avoid wasting cpu

2015-10-19 17:50:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: implement accept-caps
	  Implement accept-caps handler to avoid doing a full caps query
	  downstream to handle it.
	  This commit implements accept-caps as a simplification of the _getcaps
	  function, so it exposes the same limitations that getcaps would.
	  For example, not accepting renegotiation to caps with capsfeatures when
	  it was last configured to a caps that it has to deinterlace.

2015-10-19 17:06:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/deinterlace.c:
	  tests: deinterlace: fix small typo in comment

2015-10-26 00:41:28 +1100  Jan Schmidt <jan@centricular.com>

	* tests/files/Makefile.am:
	  check: Dist splitvideo0[012].ogg test files.

2015-10-23 20:16:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempo.h:
	  scaletempo: Add support for F64

2015-10-22 17:40:38 -0700  Mischa Spiegelmock <mspiegelmock@gmail.com>

	* docs/plugins/inspect/plugin-rtp.xml:
	* gst/multipart/multipartdemux.c:
	* gst/rtp/README:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/udp/gstudpsrc.c:
	  docs: Minor fixes in various places
	  https://bugzilla.gnome.org/show_bug.cgi?id=756996

2015-10-21 17:43:31 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/goom/plugin_info.c:
	  goom: remove compiler trick
	  After commit 2cb6cfed22166b262ae50cb58f3ff11dd8ba91f9 there is no need to
	  trick the compiler anymore about the usage of variable cpuFlavour.

2015-10-21 14:35:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* common:
	  Automatic update of common submodule
	  From b99800a to b319909

2015-10-21 17:41:38 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audiofxbaseiirfilter.h:
	  audiofx: remove unused variable
	  Remove unsued variable have_coeffs in audiofxbaseiirfilter
	  https://bugzilla.gnome.org/show_bug.cgi?id=756905

2015-10-20 17:29:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Use new GST_ENABLE_EXTRA_CHECKS #define
	  https://bugzilla.gnome.org/show_bug.cgi?id=756870

2015-10-21 14:25:55 +0300  Sebastian Dröge <sebastian@centricular.com>

	* README:
	* common:
	  Automatic update of common submodule
	  From 9aed1d7 to b99800a

2015-10-21 11:53:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: relax creation time parsing
	  Parse wrong timestamps like we used to write as well,
	  e.g. 10:9:42, and the hour might be without a leading
	  zero in any case.

2015-10-21 11:45:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix indentation

2015-10-21 11:44:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: extract both creation date and time
	  Before we only extracted the date part.

2015-10-21 11:16:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: fix writing of creation time
	  Don't write time as e.g. 11:9:42

2015-10-13 12:42:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: update fragment offset
	  It was always being set to 0, making the resulting stream broken
	  for the receiver
	  https://bugzilla.gnome.org/show_bug.cgi?id=756422

2015-10-19 15:36:37 +0300  Ryan Hendrickson <ryan.hendrickson@alum.mit.edu>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't unconditionally use strnlen()
	  It's not available on older OSX and we can as well use memchr() here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756154

2015-10-19 17:38:32 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/auparse/gstauparse.c:
	  auparse: Fix event memory leak
	  Free the event after being handled to prevent memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756799

2015-10-19 09:14:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: unify raw audio caps into a single caps structure

2015-10-19 15:15:30 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  gl: be consistent in gobject boilerpate
	  GST_GL_IS_* vs GST_IS_GL_*
	  git grep -l 'GST_GL_IS_' | xargs sed -i 's/GST_GL_IS_/GST_IS_GL_/g'

2015-10-19 15:15:30 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gl: be consistent in gobject boilerpate
	  GST_GL_IS_* vs GST_IS_GL_*
	  git grep -l 'GST_GL_IS_' | xargs sed -i 's/GST_GL_IS_/GST_IS_GL_/g'

2015-10-17 15:26:46 +1100  Matthew Waters <matthew@centricular.com>

	* tests/examples/gtk/glliveshader.c:
	  glshaderelement: implement on-demand create-shader signalling
	  One may not have an GstGLContext available or current in the thread where one
	  would need to update the shader.  Support this by signalling create-shader
	  whenever the one-shot 'update-shader' is set to TRUE.

2015-10-17 02:40:50 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkbasesink.c:
	  gtk: separate out the widget/window destroy callbacks
	  Fixes assertion due to the sink_finalize() being run before the widget destroy
	  callback.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755969

2015-10-17 01:08:29 +1100  Matthew Waters <matthew@centricular.com>

	* tests/examples/gtk/Makefile.am:
	* tests/examples/gtk/glliveshader.c:
	  gl/examples: add a live shader demo using the new GstGLSLStage
	  Implemented with videotestsrc ! glshader ! glupload ! gtkglsink
	  Errors on an invalid shader compilation are ignored however any error
	  provided by the glsl compiler is printed to stdout.

2015-10-14 15:42:50 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for FFV1 coded streams in mov
	  https://bugzilla.gnome.org/show_bug.cgi?id=752495

2015-09-04 16:02:32 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  glshader: port to using GstGLSLStage objects for string management
	  A GstGLShader is now simply a collection of stages that are
	  compiled and linked together into a program.  The uniform/attribute
	  interface has remained the same.

2015-10-14 15:53:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: EOS immediately if we have an empty seek segment
	  https://bugzilla.gnome.org/show_bug.cgi?id=748316

2015-10-14 10:43:19 +0300  Stavros Vagionitis <stavrosv@digisoft.tv>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Make non-inclusive segment boundaries inclusive
	  The problem is that the filesrc and souphttpsrc are behaving
	  differently regarding the calculation of the segment boundaries. The
	  filesrc is using a non-inclusive boundaries, while the souphttpsrc
	  uses inclusive. Currently the hlsdemux calculates the boundaries as
	  inclusive, so for this reason there is no problem with the souphttpsrc,
	  but there is an issue in the filesrc.
	  The GstSegment is non-inclusive, so the proposed solution is to use
	  non-inclusive boundaries in the hlsdemux in order to be consistent.
	  Make the change in the hlsdemux, will break the souphttpsrc, which
	  will expect inclusive boundaries, but the hlsdemux will offer
	  non-inclusive. This change makes sure that the non-inclusive
	  boundaries are converted to inclusive.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748316

2015-10-11 22:07:54 +0000  Graham Leggett <minfrin@sharp.fm>

	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpclientsink.h:
	  souphttpclientsink: Add the retry and retry-delay properties
	  These allow a failed request to be retried after the given number of seconds
	  instead of failing the pipeline. Take account of the Retry-After header if
	  present. Add retries parameter that controls the number of times an HTTP
	  request will be retried before failing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756318

2015-10-14 12:03:15 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix caps leak
	  If the QtDemuxStream are re-used they may already have caps which used
	  to be leaked.
	  Reproduced using the
	  validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
	  scenario.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756561

2015-10-14 09:29:50 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix taglist memory leak
	  Free the stream and its sub items instead of just the stream
	  https://bugzilla.gnome.org/show_bug.cgi?id=756544

2015-10-11 12:06:26 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Allow negotiating to S8 as a raw format but stop making it best choice
	  Negotiation to audio/x-raw,format=S8 was not possible because S8 does
	  not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
	  https://bugzilla.gnome.org/show_bug.cgi?id=756387

2015-10-11 09:18:40 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Add prores support
	  https://bugzilla.gnome.org/show_bug.cgi?id=756388

2015-10-12 18:56:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	  tests: add GST_PLUGINS_BASE_LIBS for flvdemux check
	  So it pulls in the right libgsttag-1.0.

2015-10-11 22:27:47 +0100  Julien Isorce <j.isorce@samsung.com>

	* gst/goom/Makefile.am:
	* gst/goom/gstaudiovisualizer.c:
	* gst/goom/gstaudiovisualizer.h:
	* gst/goom/gstgoom.h:
	* gst/goom2k1/Makefile.am:
	* gst/goom2k1/gstaudiovisualizer.c:
	* gst/goom2k1/gstaudiovisualizer.h:
	* gst/goom2k1/gstgoom.h:
	  goom/goom2k1: remove obsolete left over files
	  They now use the new GstAudioVisualizer base class
	  from gst-plugins-base/gst-libs/gst/pbutils
	  Also fixed undefined reference to gst_audio_visualizer_get_type
	  Added GST_PLUGINS_BASE_LIBS to Makefile.am and re-order LIBADD.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-10-12 10:48:23 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: Fix buffer memory leak during failures
	  mapped buffer is not being unmapped during failures
	  https://bugzilla.gnome.org/show_bug.cgi?id=756231

2015-10-12 11:18:51 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Check if soup message is created
	  If soup message is not created then the same should not be passed
	  on, which is resulting in segfault. Hence throwing a warning message
	  and returning
	  https://bugzilla.gnome.org/show_bug.cgi?id=755326

2015-10-12 11:15:15 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Check if location being set is valid
	  Adding a check in set_property to find if the location uri is valid
	  and printing warning if not valid.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755326

2015-10-12 11:09:30 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Fix memory leaks during failures
	  freeing streamheader_buffers and sent_buffers during failure cases.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755326

2015-10-12 11:03:17 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Replace redundant free_buffer_list function
	  Removing free_buffer_list and replacing it with already available function
	  g_list_free_full
	  https://bugzilla.gnome.org/show_bug.cgi?id=755326

2015-10-11 16:40:01 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/Makefile.am:
	  check: Don't forget base CFLAGS for flvdemux check
	  elements/flvdemux.c:25:25: fatal error: gst/tag/tag.h: No such file or directory

2015-10-11 11:37:51 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: Create a TIME segment when creating streamable output
	  Related to https://bugzilla.gnome.org/show_bug.cgi?id=754435 which
	  does the same for flvmux.

2015-09-23 13:50:52 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/Makefile.am:
	* gst/flv/gstflvdemux.c:
	* tests/check/Makefile.am:
	* tests/check/elements/flvdemux.c:
	  flvdemux: output speex vorbiscomment as a GstTagList
	  This is what speexdec expects.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755478

2015-09-22 22:59:16 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvmux.c:
	* tests/check/elements/flvmux.c:
	  flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE
	  Or else flvdemux don't understand it
	  https://bugzilla.gnome.org/show_bug.cgi?id=754435

2015-09-02 10:44:59 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvmux.c:
	* tests/check/elements/flvmux.c:
	  flvmux: use time segment and copy timestamps when streamable
	  Add a basic test using speex data to verify timestamping.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754435

2015-09-23 13:14:03 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: speex is also always 16KHz
	  This is just a cosmetic change for the logs, since the right caps
	  for Speex is being set elsewhere.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755479

2015-07-14 15:19:44 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: Add 'source-stats' to stats and notify
	  Add statitics from each rtp source to the rtp session property.
	  'source-stats' is a GValueArray where each element is a GstStructure of
	  stats for one rtp source.
	  The availability of new stats is signaled via g_object_notify.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752669

2015-06-05 17:20:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Implement sending of reduced size RTCP packets
	  https://bugzilla.gnome.org/show_bug.cgi?id=750456

2015-10-08 15:01:13 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audiodynamic.h:
	  audiofx: Remove unused variable
	  Remove unused variable 'degree' in audiodynamic
	  https://bugzilla.gnome.org/show_bug.cgi?id=756234

2015-10-08 14:44:07 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix memory leak for corrupted file
	  Free brands before overriding them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756226

2015-10-08 11:44:04 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	  gdkpixbufdec: Fix pixbuf_loader leak during failures
	  https://bugzilla.gnome.org/show_bug.cgi?id=756219

2015-10-07 23:23:45 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Add missing break

2015-10-07 13:03:02 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations
	  https://bugzilla.gnome.org/show_bug.cgi?id=751311

2015-10-07 13:02:12 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpmanager: add "max-dropout-time" and "max-misorder-time" props
	  https://bugzilla.gnome.org/show_bug.cgi?id=751311

2015-10-07 17:14:57 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix date memory leak
	  When getting date from taglist, the memory should be freed after
	  using it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756171

2015-10-05 11:03:38 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix sample memory leak
	  When getting sample from taglist, the memory should be freed after
	  using it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756068

2015-10-05 13:10:56 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/cutter/gstcutter.c:
	  cutter: Fix buffer leak
	  Buffer is added to the internal cache, and pushed only when accumulated
	  buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
	  is not freed. Freeing the cache when the state changes from PAUSED to READY.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754212

2015-08-31 21:10:16 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Use default upstream event handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=752694

2015-08-31 21:05:03 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: As 0xFFFFFFFF is a valid ssrc, check if it has been set
	  https://bugzilla.gnome.org/show_bug.cgi?id=752694

2015-07-22 09:47:22 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* tests/check/elements/rtpmux.c:
	  gstrtpmux: allow the ssrc-property to decide ssrc on outgoing buffers
	  By not doing this, the muxer is not effectively a rtpmuxer, rather a
	  funnel, since it should be a single stream that exists the muxer.
	  If not specified, take the first ssrc seen on a sinkpad, allowing upstream
	  to decide ssrc in "passthrough" with only one sinkpad.
	  Also, let downstream ssrc overrule internal configured one
	  We hence has the following order for determining the ssrc used by
	  rtpmux:
	  0. Suggestion from GstRTPCollision event
	  1. Downstream caps
	  2. ssrc-Property
	  3. (First) upstream caps containing ssrc
	  4. Randomly generated
	  https://bugzilla.gnome.org/show_bug.cgi?id=752694

2015-10-02 22:42:20 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fixup last commit

2015-10-02 22:21:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	* gst/udp/gstudpsrc.c:
	  Update GLib dependency to 2.40.0

2015-06-30 16:56:19 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpstats: add utility for calculating RTP packet rate

2015-08-10 18:14:39 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: handle empty segments in seeking adjust
	  If seeking targets an empty segment skip it as there is no media
	  offset to get from it. Instead look for the next one.
	  This doesn't make seeking in push-mode work if you seek to an
	  empty segment but at least won't get you to wrong offsets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753484

2015-04-17 14:25:43 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: post messages when fragments are being opened and closed
	  This can be useful for applications that need to track the created fragments
	  (to log them in a recording database, for example)
	  https://bugzilla.gnome.org/show_bug.cgi?id=750108

2015-04-29 18:23:28 +0100  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: allow non-video streams to serve as reference
	  In the absence of a video stream, the first stream will be used as
	  reference.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753617

2015-07-22 17:45:12 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: initialize mux_start_time properly
	  mux_start_time refers to the running_time of the buffer
	  that goes first in the output file. Normally this time is
	  0, so this variable is initialized to 0 during the state
	  change to PAUSED.
	  However, when dealing with dynamic pipelines and starting
	  a recording while the pipeline has already run for a while,
	  the running_time of the first buffer is > 0 and this causes
	  a problem with detecting the end of the first file(s) when
	  splitting by duration, because the code will later compare
	  the threshold_time with (last buffer running_time - mux_start_time)
	  and will get it wrong until mux_start_time advances enough
	  to make this difference < threshold_time, creating empty files
	  in the meantime.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753624

2015-09-16 16:03:02 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Reverse playback does not consider segment.start
	  During reverse playback, the media should stop playing at segment.start
	  This does not happen, and avidemux continues to process data even when
	  current timestamp is less that segment.start.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755094

2015-09-23 12:39:35 +0900  Manasa Athreya <manasa.athreya@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check multi trex to find track id in mp4 mpeg-dash stream
	  If stream has more than one trex box which is not matched to actual
	  track id, it makes qtdemux crashed.
	  Author : Manasa Athreya (manasa.athreya@lge.com)
	  https://bugzilla.gnome.org/show_bug.cgi?id=754864

2015-09-04 14:24:45 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/smpte/gstsmpte.c:
	  smpte: get size, stride info using VideoInfo
	  Use VideoInfo data to get size stride and
	  offset, instead of hard coded macros.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754558

2015-09-04 14:18:50 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/smpte/gstsmpte.c:
	  smpte: free mask
	  Free the memory allocated to 'mask' to avoid
	  memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754555

2015-08-20 11:02:58 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* tests/examples/equalizer/demo.c:
	* tests/icles/equalizer-test.c:
	* tests/icles/gdkpixbufoverlay-test.c:
	* tests/icles/gdkpixbufsink-test.c:
	* tests/icles/test-oss4.c:
	* tests/icles/videocrop-test.c:
	  gstreamer: good: tests: Fix memory leaks when context parse fails.
	  When g_option_context_parse fails, context and error variables are not getting free'd
	  which results in memory leaks. Free'ing the same.
	  And replacing g_error_free with g_clear_error, which checks if the error being passed
	  https://bugzilla.gnome.org/show_bug.cgi?id=753853

2015-10-02 16:18:15 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: doesn't handle probation and rtp gap in case of sender
	  https://bugzilla.gnome.org/show_bug.cgi?id=754548

2015-10-02 16:16:32 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* docs/plugins/gst-plugins-good-plugins.signals:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpsession.h:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpmanager: add new on-new-sender-ssrc, on-sender-ssrc-active signals
	  Allows for applications to get internal source's RTP statistics.
	  (eg. sender sources for a server/client)
	  https://bugzilla.gnome.org/show_bug.cgi?id=746747

2015-09-15 03:14:37 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstplugin.cc:
	* ext/qt/gstqsgtexture.h:
	* ext/qt/gstqtsink.cc:
	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  qt: add support for building on osx/ios
	  Including:
	  - Necessary configure checks
	  - Necessary compile time platform checks
	  - Necessary runtime qt iOS/OSX platform detection
	  https://bugzilla.gnome.org/show_bug.cgi?id=755100

2015-10-02 14:17:48 +1000  Jan Schmidt <jan@centricular.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Gather and coalesce all damaged areas before retrieving.
	  These days the xserver seems to give us the same damage regions
	  over and over for entire windows, and we retrieve them multiple
	  times, which gives time for more damage to appear. Instead, just
	  quickly gather all damaged areas into a region list and copy
	  out once.

2015-10-01 16:24:32 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/goom2k1/Makefile.am:
	* gst/goom2k1/gstgoom.h:
	  goom2k1: use the new audiovisualizer base class
	  Rebase to have goom using the GstAudioVisualizer base class in
	  gst-plugins-base/gst-libs/gst/pbutils
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-10-01 16:16:08 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/goom/Makefile.am:
	* gst/goom/gstgoom.h:
	  goom: use the new audiovisualizer base class
	  Rebase to have goom using the GstAudioVisualizer base class in
	  gst-plugins-base/gst-libs/gst/pbutils
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-09-30 17:35:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/interleave/deinterleave.c:
	* tests/check/elements/deinterleave.c:
	  deinterleave: implement accept-caps
	  Avoid using default accept-caps handler that will query downstream
	  and is more expensive. Just check if the caps is compatible with
	  the template and check if the channels are the same.

2015-09-30 09:35:39 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/deinterleave.c:
	  tests: deinterleave: also check for caps query results

2015-09-30 12:30:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: use the caps query filter
	  It was being ignored and would lead to wrong results if the
	  element doing the query would rely on the intersection being made.

2015-09-30 10:00:31 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: implement a caps query handler for the sinkpad
	  It was missing and apparently code relied on having it there
	  for not allowing a change in the number of channels

2015-09-30 09:05:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: fix caps leak
	  Caps from the pad template are being leaked. In any case it is
	  from a static pad template and will 'leak' in the end, just doing
	  the cleanup for the good practice.

2015-09-29 22:57:52 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: add some GL debug statements to show up in GL traces

2015-08-28 16:24:24 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* ext/qt/gstqtsink.cc:
	  qtsink: explicitely fallthrough switch statement
	  In case ret is False, fallthrough to default case.
	  CID #1320705

2015-09-29 11:15:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/gdkpixbufoverlay.c:
	  tests: gdkpixbufoverlay: add minimal unit test
	  https://bugzilla.gnome.org/show_bug.cgi?id=755773

2015-09-29 11:12:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufsink: don't leak old pixel buffer when setting a new overlay
	  https://bugzilla.gnome.org/show_bug.cgi?id=755773

2015-09-28 20:25:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/flac/gstflacenc.c:
	  flacenc: avoid potential string overflow
	  We don't necessarily have full control over the input tags, so
	  it's possible that the ISRC tag contains a longer string than
	  expected, in which case we'd write over the end of the static-size
	  13 byte buffer that is FLAC__StreamMetadata_CueSheet_Track::isrc.
	  Make sure to only copy the ISRC if it's not too long, and make
	  sure the buffer we write to is always NUL-terminated by using
	  g_strlcpy().
	  CID 1324931.

2015-09-28 18:03:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Remove leftover assertion from 0.10
	  We now allocate memory via GstAllocator and as such can handle arbitrary
	  alignments, not only <= G_MEM_ALIGN.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755708

2015-09-29 00:25:00 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkbasesink.c:
	  gtk: fix assertion when the element has no peer
	  When proxying keyboard/navigation/mouse events, only unref a successfully
	  retreived peer pad.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755738

2015-08-28 16:35:39 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* ext/qt/qtitem.cc:
	  qml: remove overwritten value
	  Value in tex is overwritten before being used. Removing it.
	  CID 1320715
	  https://bugzilla.gnome.org/show_bug.cgi?id=754253

2015-09-02 23:45:07 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/Makefile.am:
	* ext/qt/gstqsgtexture.h:
	* ext/qt/gstqtgl.h:
	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	  qt: add support for building/running on android
	  Including:
	  - Necessary configure checks
	  - Necessary compile time platform checks
	  - Necessary runtime qt android platform detection
	  - Escaping GLsync definition with Qt's GLES2 implementation
	  https://bugzilla.gnome.org/show_bug.cgi?id=754466

2015-09-02 23:40:31 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/Makefile.am:
	  qt: don't use CPPFLAGS for tools that cannot use them
	  For example moc will bail out when given arguments it does not
	  know about.  The moc specific MOC_CPPFLAGS can still be used
	  to pass flags to moc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754466

2015-09-02 23:39:54 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/Makefile.am:
	  qt: rename library to include gst prefix
	  libqtsink -> libgstqtsink
	  https://bugzilla.gnome.org/show_bug.cgi?id=754466

2015-09-25 10:01:37 +0200  Guillaume Marquebielle <guillaume.marquebielle@parrot.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix uninitialized variables in LOAS config reading
	  On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
	  "Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
	  'channels' are pointing to uninitialized values can end on setting random values as
	  rate and channels on src caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755611

2015-09-18 00:58:23 +1000  Jan Schmidt <thaytan@noraisin.net>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	  Fix some compiler warnings when building with G_DISABLE_ASSERT
	  Touches rtpmanager and gdkpixbufsink

2015-08-18 14:30:57 +0100  Chris Bass <floobleflam@gmail.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: support timed-text subtitle tracks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752818

2015-09-26 00:12:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  gst: Don't use deprecated gst_segment_to_position()

2015-09-21 13:47:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time
	  https://bugzilla.gnome.org/show_bug.cgi?id=755125

2015-09-16 19:28:11 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin/session: Allow RTCP sync to happen based on capture time or send time
	  Send time is the previous behaviour and the default, but there are use cases
	  where you want to synchronize based on the capture time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755125

2015-09-25 23:51:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.6.0 ===

2015-09-25 23:15:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.6.0

2015-09-25 22:57:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2015-09-25 14:08:09 +0200  Thibault Saunier <tsaunier@gnome.org>

	* gst/smpte/gstsmptealpha.c:
	  smptealpha: Do not set width/height before comparing with old values
	  Otherwise we end up considering the values did not change and we wrongly
	  work with the old video format (which will lead to wrong
	  behaviour/segfaults).
	  https://bugzilla.gnome.org/show_bug.cgi?id=755621

2015-09-24 18:51:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gstgtkbasesink.c:
	  gtk: Only run from the main thread in stop() if we created the window
	  We're not doing anything at all from the main thread in other cases.

2015-09-24 15:52:40 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: When setting format check if pending format changed
	  In case the format changed fast and the pending format is different
	  than the currently set but the currently set is equal to the pending
	  one we could end up having mismatch between the finally set format
	  and the data stream format.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755542

2015-09-24 15:51:28 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gstgtkbasesink.c:
	  gtk: Do not forget to release OBJECT_LOCK on error path
	  https://bugzilla.gnome.org/show_bug.cgi?id=755542

2015-09-24 11:37:04 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/Makefile.am:
	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkutils.c:
	* ext/gtk/gstgtkutils.h:
	* ext/gtk/gtkgstglwidget.c:
	  gtk: Factor out a function to run a function on main thread
	  https://bugzilla.gnome.org/show_bug.cgi?id=755251

2015-09-24 10:51:31 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gstgtkbasesink.c:
	  gtk: Marshall state changes in the main thread
	  Gtk is not MT safe thus we need to make sure that everything is done
	  in the main thread when working with it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755251

2015-09-23 20:59:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Accumulate segments for edit lists before activating the next segment
	  eceb2ccc739092d964d78945e19c2ecedbd214e2 broke segment seeks by always
	  accumulating segments manually when activating a segment. This is only
	  needed when handling edit lists, not when activating a segment because of a
	  seek. Do the accumulation when switching edit list segments instead.
	  This fixes segment seeks again, while keeping edit lists playback working.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755471

2015-09-23 17:43:51 +0530  Vikram Fugro <vikram.fugro@gmail.com>

	* gst/spectrum/gstspectrum.c:
	  spectrum: send phase values in the GstMessage for Phase info
	  https://bugzilla.gnome.org/show_bug.cgi?id=755463

2015-09-23 11:42:51 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gstgtkbasesink.c:
	  gtksink: Do not show window until we reach the PAUSED state
	  https://bugzilla.gnome.org/show_bug.cgi?id=755459

2015-09-22 00:46:01 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Don't output a warning on MONO multiview mode.

2015-09-21 10:47:15 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gstgtkbasesink.c:
	  gtksink: Do not re destroy the GtkWindow if destroyed by the user
	  Otherwise we will get an ASSERT.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755249

2015-09-19 17:02:18 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: Fix memory leaks
	  The same memory leaks were fixed in identical fashion for
	  vorbisdepay in 06efeff5d979576a252e5dae57f46d6445b1df12 in 2009.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277

2015-09-19 17:04:07 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup
	  * use g_list_free_full(), don't iterate elements maually when freeing
	  * call gst_rtp_*_pay_clear_packet(), don't duplicate its code
	  * use gst_buffer_unref() to clarify that it is buffers being released,
	  instead of refering directly to gst_mini_object_unref()
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277

2015-09-19 18:44:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=755265

2015-09-19 11:46:37 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkbasesink.h:
	* ext/gtk/gstgtkglsink.c:
	  gtkglsink: Hide and clean the GtkWindow we might create
	  When stopping the sink we should always hide the window.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755249

=== release 1.5.91 ===

2015-09-18 19:33:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.5.91

2015-09-18 19:23:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2015-09-18 11:50:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/zh_CN.po:
	  po: Update translations

2015-09-17 10:50:01 +0900  Eunhae Choi <eunhae1.choi@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix taglist leak
	  gst_tag_list_insert() does not take ownership of the inserted taglist.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755138

2015-09-17 13:35:02 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* ext/gtk/gtkgstglwidget.c:
	  gl: Fix GError leaks during failures
	  https://bugzilla.gnome.org/show_bug.cgi?id=755140

2015-09-16 07:05:36 +1000  Jan Schmidt <jan@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Skip LOAS AAC until a valid config is seen.
	  It's normal when dropping into the middle of a stream to
	  not always have the config available immediately, so skip LOAS
	  until a valid config is seen without either setting invalid
	  caps or erroring out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751386

2015-09-13 15:41:38 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: reset just a bit more upon flush_stop

2015-09-13 15:40:09 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: remove dead struct member

2015-09-11 17:09:28 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: fix GError memory leak when hostname resolution fails
	  https://bugzilla.gnome.org/show_bug.cgi?id=754869

2015-09-10 15:26:54 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/matroska/ebml-write.c:
	  matroskamux: drop HEADER flag from output buffers
	  Drop HEADER flag from output buffers if they are not indeed
	  headers.
	  Fixes resending of headers in tcp connection handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=754768

2015-09-10 16:00:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/ebml-write.c:
	  matroskamux: fix matroskamux ! matroskademux
	  Don't carry over DISCONT flags from the input buffers to the
	  output buffer, or the demuxer might reset its state when it
	  receives the first data buffer just after parsing the simple
	  block header, and then expect sane data to follow.
	  Fixes matroskamux ! demux erroring out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754768
	  https://bugzilla.gnome.org/show_bug.cgi?id=657805

2015-09-09 12:51:40 -0700  Martin Kelly <martin@surround.io>

	* gst/rtsp/README:
	  rtsp: fix small README typo
	  https://bugzilla.gnome.org/show_bug.cgi?id=754807

2015-09-10 00:07:18 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  gtk, qt: more specifically define the compile time requirements
	  Otherwise we could include headers/configurations that will
	  never been installed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754732

2015-09-10 00:07:18 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk, qt: more specifically define the compile time requirements
	  Otherwise we could include headers/configurations that will
	  never been installed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754732

2015-09-10 00:00:11 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqsgtexture.cc:
	  qt: use our function table instead of directly calling gl functions
	  Otherwise when building with --as-needed we would need to link to
	  a GL or GLES library.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754732

2015-09-04 19:45:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	  wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
	  https://bugzilla.gnome.org/show_bug.cgi?id=752106

2015-09-04 19:34:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
	  flac contains the sample offset in the frame header, so after a seek
	  without index flacparse will know the exact position we landed on and
	  timestamp buffers accordingly. It only set the pts though, which means
	  the baseparse-set dts which was set to the seek position prevails, and
	  since the seek was based on an estimate, there's likely a discrepancy
	  between where we wanted to land and where we did land, so from here on
	  that dts/pts difference will be maintained, with dts possibly multiple
	  seconds ahead of pts, which is just wrong. The easiest way to fix this
	  is to just set both pts and dts based on the sample offset, but perhaps
	  parsed audio should just not have dts set at all.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752106

2015-09-06 16:33:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	  docs: remove properties and signals that no longer exist
	  https://bugzilla.gnome.org/show_bug.cgi?id=726443

2013-10-11 15:13:00 +0000  George Chriss <gschriss@gmail.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Make the element count in arrays not include end
	  One-line removal of tags_written++
	  This should fix rtmp output to crtmpserver, and hopefully
	  noone is expecting that the element count includes the end
	  element, as different bits of documentation say different
	  things about whether it should or not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661624

2015-07-30 00:59:15 +1000  Jan Schmidt <jan@centricular.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Store incoming bitrate tags and send in the metadata
	  Apparently the Microsoft Azure RTMP server requires that the
	  videodatarate and audiodatarate metadata be provided, so
	  set those, even if it's to 0. Use the actual input bitrate
	  tags if available.

2015-09-04 00:06:29 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't parse key data more than needed.
	  When an auxilliary streams are present in the SDP media,
	  there's no need to re-parse the SDP attributes multiple
	  times.

2015-09-03 20:56:55 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
	  In parse_keymgmt(), don't mutate the input string that's been passed
	  as const, especially since we might need the original value again if
	  the same key info applies to multiple streams (RTX, for example).
	  When a resource is 404, and we have auth info - retry with the auth
	  info the same as if we had receive unauthorised, in case the resource
	  isn't even visible until credentials are supplied.
	  Fix a memory leak handling Mikey data.
	  When generating a random keystring, don't overrun the 30 byte
	  buffer by generating 32 bytes into it.

2015-09-04 15:43:40 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: Do not consider GtkEvents as handled
	  Applications might still want to use them
	  after the sink transformed them into
	  GstNavigation events

2015-09-04 15:18:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix build with GLib < 2.44
	  G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.

2015-09-04 12:01:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
	  This happens on Windows if we use the same socket for sending packets,
	  and the remote sends ICMP port/host unreachable messages.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754534

2015-09-02 21:12:41 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbis/theoradepay: Fix handling of fragmented packets
	  This was broken in b1089fb520 by not considering the full packet length of a
	  fragmented packet but only the length of the first one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754417

2015-09-01 15:39:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmfsrc: Reply to latency query

2015-08-07 17:27:48 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/qtitem.cc:
	  qmlsink: Ensure that at least one windowing system is available
	  Otherwise, we'll just crash at runtime because the gl context is NULL
	  https://bugzilla.gnome.org/show_bug.cgi?id=754108

2015-08-31 16:42:30 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpsession.c:
	  tests: Fix rtpsession test failure
	  The time of the first RTCP packet is semi-random, so
	  sometimes it was produced before enough packets from
	  the second SSRC were received. First drop queued RTCP
	  packets, then advance the clock enough to ensure
	  that at least one new RTCP packet is produced.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750731

2015-08-31 18:06:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk, qt, gl: fix typo in debug and error messages

2015-08-31 18:06:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/qt/gstqtsink.cc:
	* ext/qt/qtitem.cc:
	  gtk, qt, gl: fix typo in debug and error messages

2015-08-31 13:56:04 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/level.c:
	  level: improve the test for multi-channel mode
	  Change the test to verify the read-index for multiple messages per buffer.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=754144

2015-08-31 12:46:52 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Align raw video frames to 32 bytes
	  Outputting unaligned video frames causes videoscale et al to
	  crash when attempting SIMD-accelerated conversion.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736965

2015-08-26 23:16:46 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: fix level calculations for mutliple channels
	  This was broken with 7b90bf32150897a141a29a12ecab555d8c5b7fab.

2015-08-27 10:28:55 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/smpte/gstsmpte.c:
	  smpte: Fix memory leak
	  In gst_smpte_collected(), check upfront if input formats are same
	  or not. This avoids allocation of in1 and in2 buffers and
	  subsequent memory leak when input formats do not match.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754153

2015-08-21 11:52:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: don't try to connect to dead radio server

2015-08-21 16:29:16 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Trivial fix to check correct condition
	  When checking for describe method, because of missing parentheses, wrong
	  condition is being checked, which will result in wrong behavior.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753912

2015-08-21 13:19:02 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/matroska/matroska-read-common.c:
	  matroska: read: fix tag list memory leak
	  gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
	  be unref'ed as soon as the tags are merged to the tocentry
	  https://bugzilla.gnome.org/show_bug.cgi?id=753904

2015-08-21 12:20:59 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: fix taglist memory leak
	  When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
	  by audiodecoder and the caller should free the taglist being passed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753903

2015-08-20 14:45:33 +0200  Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: fix pad closing
	  Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=753875

2015-08-19 13:52:21 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk/gl: Use our GL function table instead of directly calling GL functions
	  Otherwise we would have to link the plugin to the GL libraries directly.

=== release 1.5.90 ===

2015-08-19 13:29:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.5.90

2015-08-19 12:47:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2015-08-19 11:29:55 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	* po/zh_CN.po:
	  po: Update translations

2015-08-13 17:29:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: fix regression with starting from index set via index property
	  When we haven't started yet, set the start_index when we set the index property,
	  so that we start at the right index position after the initial seek. The index
	  property was never really meant to be for writing, but it used to work, so let's
	  support it for backwards compatibility.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739472

2015-08-18 10:52:11 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix offset calculation when parsing CENC aux info
	  Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for
	  DASH common encryption, however commit
	  bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master
	  shortly before the CENC commit caused the calculation of the CENC
	  aux info offset to be incorrect.
	  The base_offset was being added if present, but if the base_offset
	  is relative to the start of the moof, the offset was being added twice.
	  The correct approach is to calculate the offset from the start of the
	  moof and use that offset when parsing the CENC aux info.

2015-08-17 14:28:24 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacenc.c:
	  flacenc: actually return true for accept-caps query handling

2015-08-17 14:07:10 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpklvpay.c:
	  rtp: copy metadata in the (de)payloaders which is missed before
	  https://bugzilla.gnome.org/show_bug.cgi?id=753706

2015-08-16 15:21:51 -0400  Dustin Spicuzza <dustin@virtualroadside.com>

	* configure.ac:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: allow specifying audio playback device
	  https://bugzilla.gnome.org/show_bug.cgi?id=753670

2015-08-16 13:51:47 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacenc.c:
	  flacenc: remove single entry if from loop
	  Iterate from the 2nd channel on and create the 1 channel struct
	  outside to make loop structure simpler and only slightly faster.

2015-08-16 13:21:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacenc.c:
	  flacenc: implement proper accept-caps
	  Should just compare with what can be immediatelly accepted by
	  the element. flacenc can't renegotiate so if it has a caps already
	  it should only accept if it is that caps otherwise just use the
	  template caps

2015-08-16 13:03:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacenc.c:
	  flacenc: improve sink pad template caps
	  Removes the need for custom caps query handling and makes it more
	  correct from the beginning on the template. It is a bit uglier
	  to read because there is 1 entry per channel but makes code easier
	  to maintain.

2015-08-16 12:41:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/y4m/gsty4mencode.c:
	  y4mencode: fix gst-launch version in documentation

2015-08-15 22:32:21 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/speex/gstspeexenc.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-encode.c:
	  audioencoders: use template subset check for accept-caps
	  It is faster than doing a query that propagates downstream and
	  should be enough
	  Elements: speexenc, wavpackenc, mulawenc, alawenc

2015-08-15 22:29:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngenc.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	* gst/y4m/gsty4mencode.c:
	  videoencoders: use template subset check for accept-caps
	  It is faster than doing a query that propagates downstream and
	  should be enough
	  Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc

2015-08-16 17:21:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: use new baseparse API to fix tag handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=679768

2015-03-17 17:50:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: use new base parse API to fix tag handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=679768

2015-08-16 14:37:53 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: use new baseparse API and fix tag handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=679768

2015-08-16 13:04:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use signed integer type to be able to check for negative subtraction results
	  CID 1315829

2015-08-16 11:50:34 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbisdepay: remove dead code
	  payload_buffer must be NULL in ignore_reserved. Check will always be false.
	  Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf
	  CID #1316476

2015-08-15 22:45:53 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/law/alaw-encode.c:
	* gst/law/alaw-encode.h:
	  alawenc: port to AudioEncoder base class

2015-08-15 22:15:26 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/twolame/gsttwolamemp2enc.c:
	  audioencoders: use template subset check for accept-caps
	  It is faster than doing a query that propagates downstream and
	  should be enough
	  Elements: amrnbenc, lamemp3enc, twolamemp2enc

2015-08-15 22:15:26 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/lame/gstlamemp3enc.c:
	  audioencoders: use template subset check for accept-caps
	  It is faster than doing a query that propagates downstream and
	  should be enough
	  Elements: amrnbenc, lamemp3enc, twolamemp2enc

2015-08-15 09:16:23 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacdec.c:
	* ext/speex/gstspeexdec.c:
	* ext/wavpack/gstwavpackdec.c:
	* gst/law/alaw-decode.c:
	* gst/law/mulaw-decode.c:
	  audiodecoders: use default pad accept-caps handling
	  Avoids useless check of downstream caps when handling an
	  accept-caps query
	  Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec

2015-08-15 08:49:57 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  videodecoders: use default pad accept-caps handling
	  Avoids useless check of downstream caps when handling an
	  accept-caps query
	  Elements: jpegdec, pngdec, vp8dec, vp9dec

2015-08-15 11:31:04 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/law/alaw-decode.c:
	  alawdec: make error handling a bit nicer
	  Print the element along with the debug to make it easier to trace
	  the failures

2015-08-15 11:04:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-decode.h:
	  alawdec: port to audiodecoder base class
	  mulawdec was already ported, alawdec was left behind.

2015-08-15 10:34:14 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only look for more samples in moofs in pull-mode
	  For playback of some fragmented formats with qtdemux it will
	  try to look for the next moof after finishing one but it is only
	  possible for pull-mode. For playback of streaming fragmented formats
	  such as DASH it should just not try to look for another moof but
	  instead wait for more data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752602
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2015-08-15 14:31:15 +0200  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	  glsink: Enable sync meta on pools we offer
	  As the upload is asynchronous, we need to enable the sync meta to
	  gain correct rendering. The buffer pool receiver don't know about
	  that.

2015-08-15 15:12:27 +0200  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gtkgstglwidget.c:
	  gtkglsink: Add overlay composition support
	  Rendering composition overlay in GL with additional high resolution
	  overlay being added.

2015-08-15 15:08:11 +0200  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstwidget.c:
	  gtkglsink: Fix unsafe handling of buffer life time
	  We need to keep the active buffer (the one we have retreive a
	  texture id from) otherwise it's racy and upstream may upload
	  new content before we have rendered or during later redisplay.

2015-08-14 18:07:15 +0200  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	* ext/gtk/gtkgstglwidget.c:
	  gtkglsink: Remove reset path
	  The reset path is bogus and there is no reason to get rid of these
	  things during resize.

2015-08-15 12:58:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: Don't look for a second syncword
	  There are streams out there that consistently contain garbage between
	  every frame so we never ever find a second consecutive syncword.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=738237

2015-08-15 11:12:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vp8enc, vp9enc: reset multipass file index when stopping encoder
	  Fixes multipass encoding when re-using the same element/pipeline
	  for subsequent encoding runs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747728

2015-08-15 11:09:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	  vp9enc: provide support for multiple pass cache files
	  Some files may provide different caps insight of one stream. Since
	  vp9enc support caps reinit, we should support cache reinit too.
	  If more then file cache file will be created, the naming will be:
	  cache cache.1 cache.2 ...
	  Based on patch by: Oleksij Rempel <linux@rempel-privat.de>
	  https://bugzilla.gnome.org/show_bug.cgi?id=747728

2015-08-14 11:41:42 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/aacparse.c:
	  tests: aacparse: use caps query instead of accept-caps
	  The accept-caps query just does a shallow check at the current
	  element while at this test we want it to also look at downstream.
	  So use caps query there.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753623

2015-08-14 11:40:22 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: enable accept-template flag
	  Do a quick check with the pad template caps as it is enough. Users
	  should have figured the appropriate full caps on a previous caps query
	  https://bugzilla.gnome.org/show_bug.cgi?id=753623

2015-08-14 15:46:53 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: send the User-Agent header
	  Sometimes it is useful to know this information on the
	  server side. Other popular implementations (vlc, ffmpeg, ...)
	  also send this header on every message.
	  This includes a new "user-agent" property that the user
	  can set to use a custom User-Agent string. The default
	  is "GStreamer/<version>"
	  https://bugzilla.gnome.org/show_bug.cgi?id=750101

2015-08-14 15:42:42 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: wrap gst_rtsp_message_init_request in a local function
	  This will allow adding common request initialization, like the
	  user agent string, in just one place.

2015-08-14 09:36:09 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* gst/audiofx/audioecho.c:
	  audioecho: make sure buffer gets reallocated if max_delay changes
	  https://bugzilla.gnome.org/show_bug.cgi?id=753490

2015-07-09 09:51:26 +0200  Oleksij Rempel <linux@rempel-privat.de>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	  vp8enc: provide support for multiple pass cache files
	  Some files may provide different caps insight of one stream. Since vp8enc
	  support caps reinit, we should support cache reinit too.
	  If more then file cache file will be created, the naming will be:
	  cache
	  cache.1
	  cache.2
	  ...
	  https://bugzilla.gnome.org/show_bug.cgi?id=747728

2015-04-15 22:51:51 +0200  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
	  Use constantDuration to calculate the timestamp of non-first AU in the
	  RTP packet.
	  If constantDuration is not present in the MIME parameters, its value
	  must be calculated based on the timing information from two consecutive
	  RTP packets with AU-Index equal to 0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747881

2015-08-14 06:43:13 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: remove unnecessary if, g_free is null safe

2015-08-14 08:33:56 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: add property to set HTTP method
	  To allow souphttpsrc to be use HTTP methods other than GET
	  (e.g. HEAD), add a "method" property that is a string. If this
	  property is not set, GET is used.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752413

2015-08-14 11:13:01 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/generic/states.c:
	  check: Rename states unit test
	  Makes it easier to differentiate from other modules states unit test

2015-08-14 09:21:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/goom/gstaudiovisualizer.c:
	* gst/goom/gstaudiovisualizer.h:
	* gst/goom2k1/gstaudiovisualizer.c:
	* gst/goom2k1/gstaudiovisualizer.h:
	  goom: Rename get_type() function of base class to prevent symbol conflicts
	  This is a problem when statically linking.

2015-08-13 16:32:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
	  Otherwise we will just output buffers without timestamps after a reset if no
	  timestamps are provided by upstream, e.g. when using RTSP over TCP.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-08-12 17:16:01 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.h:
	  matroska: Remove unused variable
	  https://bugzilla.gnome.org/show_bug.cgi?id=753556

2015-08-12 00:18:20 +0200  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: fix motion event name
	  s/motion/mouse/
	  Fixes hover interaction with DVD menus

2015-08-12 00:14:14 +0200  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstbasewidget.c:
	  gtk: correct navigation events for window scaling
	  i.e. take into account the possiblity of scaling in the sink
	  or through GDK_SCALE.
	  Fixes DVD Menus with a scaled gtkwidget

2015-08-11 13:34:59 +0200  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	  gtk: implement GstNavigation interface
	  Now we can push key/mouse input into the pipeline for DVD use cases.

2015-08-04 20:59:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtptheorapay.h:
	* gst/rtp/gstrtputils.c:
	* gst/rtp/gstrtputils.h:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvorbispay.h:
	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: Copy metadata in the (de)payloader, but only the relevant ones
	  The payloader didn't copy anything so far, the depayloader copied every
	  possible meta. Let's make it consistent and just copy all metas without
	  tags or with only the video tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751774

2015-08-10 18:20:15 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix small typo in comment

2015-08-10 16:19:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/goom2k1/gstgoom.c:
	  goom2k1/doc: Fixup previous commit

2015-08-10 15:55:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	  goom2k1/doc: Use GstGoom2k1 namespace
	  The doc generator isn't happy when we have class name clash. Simply
	  use it's own namespace.

2015-08-10 17:10:42 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* gst/audiofx/audioecho.c:
	  audioecho: removed unused variable in set_property
	  unused local variable 'delay' is removed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753450

2015-08-10 12:45:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix suboptimal queue iteration code

2015-08-09 17:25:45 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't use glib 2.44-only API

2015-07-29 14:14:50 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: add support for ISOBMFF Common Encryption
	  This commit adds support for ISOBMFF Common Encryption (cenc), as
	  defined in ISO/IEC 23001-7. It uses a GstProtection event to
	  pass the contents of PSSH boxes to downstream decryptor elements
	  and attached GstProtectionMeta to each sample.
	  https://bugzilla.gnome.org/show_bug.cgi?id=705991

2015-08-10 14:13:50 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: checking if depay has sps/pps nals before insertion
	  https://bugzilla.gnome.org/show_bug.cgi?id=753430

2015-08-08 16:44:49 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix outdated comment
	  The default behaviour was changed in the 0.10 -> 1.x
	  transition, but the comment was not updated.

2015-08-08 17:42:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: If flushing a packet failed, go out of the loop immediately

2015-08-08 17:41:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: If flushing a packet failed, go out of the loop immediately

2015-08-08 17:34:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtptheorapay.h:
	  rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps
	  We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2
	  and 4:4:4 formats.

2015-08-08 17:28:03 +0200  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqsgtexture.cc:
	* ext/qt/gstqsgtexture.h:
	* ext/qt/qtitem.cc:
	  qml: implement the required multiple GL context synchonisation
	  From GStreamer's GL context into the QML context

2015-08-06 17:46:13 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvpay.c:
	  rtpklv(de)pay: add "RTP" in the klass string
	  GstRTSPMedia uses this classification to detect the real payloader
	  inside a dynpay bin and asserts if it doesn't find it, therefore
	  it is required
	  https://bugzilla.gnome.org/show_bug.cgi?id=753325

2015-08-05 11:13:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/rtpaux.c:
	  tests: rtpaux: use a dynamic pt in the test
	  1) Tests that using dynamic PT instead of the default ones work
	  2) If we ever decide to change the codec here we don't need to
	  worry about change the PT for the default one of the new codec
	  in the test
	  https://bugzilla.gnome.org/show_bug.cgi?id=746445

2015-08-05 10:53:15 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: print valid type where guint32 is expected
	  https://bugzilla.gnome.org/show_bug.cgi?id=746445

2015-08-06 11:33:37 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmupay.c:
	  rtppayload: set standard payload type as default
	  Initialize the PT to the default value of the codec and check if
	  it is still the default before declaring the pt to be dynamic or
	  not when setting the caps.
	  Also use the PT constants from the rtp lib when possible
	  https://bugzilla.gnome.org/show_bug.cgi?id=747965

2015-07-26 12:07:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: store the moof-offset also for push mode
	  It will be used in some cases for getting the correct offsets
	  from trun atoms.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2015-07-26 02:09:24 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.h:
	  qtdemux: handle default-base-is-moof flag
	  Handle the flag from the tfhd that signals the base offset to
	  start from the moof atom
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2015-07-29 18:54:35 -0600  Glen Diener <grd@loganmill.net>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroskademux: Preserve forward referenced track tags
	  https://bugzilla.gnome.org/show_bug.cgi?id=752850

2015-08-04 18:07:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/rtpaux.c:
	  tests: rtpaux: fix test failure
	  The RTP PT for alaw is 8.
	  Less than 50 packets are received in the length of this test so it
	  would never drop a buffer or would drop only the last buffer and
	  it would fail sometimes when the received wouldn't receive the
	  retransmission packet in time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746445

2015-08-04 20:59:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpstreamdepay.c:
	  rtpstreamdepay: Only allow activation in push mode
	  We need a proper caps event from upstream with the full RTP caps as we can't
	  create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
	  a filesrc or any other element that supports pull mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753066

2015-08-04 16:28:17 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  soup: fix typo in translated string
	  https://bugzilla.gnome.org/show_bug.cgi?id=753240

2015-08-04 12:25:46 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Put the profile and level into the caps

2015-08-04 12:09:12 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Only update the srcpad caps if something else than the codec_data changed
	  h264parse does the same, let's keep the behaviour consistent. As we now
	  include the codec_data inside the stream too here, this causes less caps
	  renegotiation.

2015-08-04 11:48:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id
	  The spec says:
	  When a picture parameter set NAL unit with a particular value of
	  pic_parameter_set_id is received, its content replaces the content of the
	  previous picture parameter set NAL unit, in decoding order, with the same
	  value of pic_parameter_set_id (when a previous picture parameter set NAL unit
	  with the same value of pic_parameter_set_id was present in the bitstream).

2015-08-03 13:45:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: remove extra \n at debug message

2015-08-03 13:42:20 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: prevent deadlock when states change too fast
	  If the GOP is completed, pads have to start gathering for the
	  next one but it is possible that the the state might go to
	  COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
	  thread has a chance to wake up and proceed, leaving it trapped in
	  the check_completed_gop loop and deadlocking the other threads
	  waiting for it to advance.
	  To solve it, this patch also checks that tha input running time
	  hasn't changed to prevent this scenario.

2015-08-03 17:55:01 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Insert SPS/PPS NALs into the stream
	  h264parse does the same and this fixes decoding of some streams with 32 SPS
	  (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
	  the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
	  As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
	  This looks like a mistake in the part of the spec about the codec_data.

2015-07-30 11:29:27 +0900  Eunhae Choi <eunhae1.choi@samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: handle empty http proxy string
	  1) If the system http_proxy environment variable is not set
	  or set to an empty string, we must not set proxy to avoid
	  http connection error.
	  2) In case of proxy property setting, if user want to clear
	  the proxy setting, they should be able to set it to NULL or
	  an empty string again, so this is fixed too.
	  3) Check if the proxy string was parsed correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752866

2015-07-29 15:46:20 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: remove unused variable
	  Remove unused variable 'framecount' from dvdemux
	  https://bugzilla.gnome.org/show_bug.cgi?id=753008

2015-07-30 15:32:09 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: assertion error due to wrong condition check
	  In media to caps function, reserved_keys array is being used for variable i,
	  leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
	  changed it to variable j
	  https://bugzilla.gnome.org/show_bug.cgi?id=753009

2015-07-30 15:21:20 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/rtp/gstrtpmp4vdepay.c:
	  rtpmp4vdepay: rtpbuffer is being unref'ed twice
	  process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay
	  the refernce should not be removed here
	  https://bugzilla.gnome.org/show_bug.cgi?id=753042

2015-07-29 11:26:46 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Strip keys from the fmtp that we use internally in our caps
	  Skip keys from the fmtp, which we already use ourselves for the
	  caps. Some software is adding random things like clock-rate into
	  the fmtp, and we would otherwise here set a string-typed clock-rate
	  in the caps... and thus fail to create valid RTP caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=753009

2015-07-29 19:28:33 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Support mpegtsmux as a muxer.
	  As a fallback, look for a pad template sink_%d on
	  the muxer when requesting pads, to support mpegtsmux
	  https://bugzilla.gnome.org/show_bug.cgi?id=752999

2015-06-25 01:35:27 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	  splitmuxsrc: Use a separate lock to delay typefind.
	  Don't hold the main splitmux part lock over
	  the parent state change function, as it prevents
	  posting error messages that happen. Since the purpose
	  is to prevent typefinding from proceeding, use a
	  separate mutex just for that.

2015-07-29 13:43:50 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/matroska/matroska-read-common.c:
	  matroska: fix memory leak
	  After adding to tag list, key_val is not being free'd
	  resulting in memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=752992

2015-07-27 13:34:14 +0900  Manasa Athreya <manasa.athreya@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc
	  'NONE' and 'raw ' fourcc don't always contain U8 audio, it can
	  be more bits as well, in which case it's just like 'twos'.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752613

2015-07-24 15:10:05 +0200  Dimitrios Katsaros <patcherwork@gmail.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Allow framerate to be large then 100pfs
	  This limit was arbitrary. We still fixate near 100pfs for compatibility.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752825

2015-07-25 03:25:28 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/avi/gstavidemux.c:
	  avidemux: Stop without posting error on flushing
	  This could just be a normal pipeline shutdown.

2015-07-23 15:00:08 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also
	  https://bugzilla.gnome.org/show_bug.cgi?id=752618

2015-07-22 15:13:48 +0200  Edward Hervey <edward@centricular.com>

	* ext/qt/Makefile.am:
	  qt: Don't dist files that might not exist
	  We only require moc building at build time.

2015-07-22 08:05:04 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/qt/Makefile.am:
	  qt: Tidy up makefile a bit more
	  Separate generated files, from disted files

2015-07-21 11:23:21 +0100  Julien Isorce <j.isorce@samsung.com>

	* ext/gtk/gtkgstglwidget.c:
	  gstglwidget: use gst_gl_display_create_context
	  Also handle the failure case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750310

2015-07-16 18:09:30 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/matroskademux.c:
	  tests: add minmal matroskademux test for subtitle output
	  Some of the subtitle chunks will have embedded
	  NUL-terminators (last three), some don't (first three),
	  some will have markup, some won't, some will be valid
	  UTF-8 (all but last), some won't (last stanza).
	  https://bugzilla.gnome.org/show_bug.cgi?id=752421

2015-07-16 18:49:26 +0300  Dimitrios Christidis <dchristidis@mykolab.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix for subtitle buffers with NUL terminators
	  Commit 45892ec8 created a regression where g_utf8_validate() would fail
	  if the subtitle buffer had a NUL terminator as part of the data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752421

2015-07-21 13:31:05 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpvp8depay: Check available bytes before copy
	  Need to check that the number of bytes we want to copy from the adapter
	  actually is available and handle the error case gracefully. This error
	  may happen if malformed packets are received and we don't have a
	  complete frame.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752663

2015-07-16 09:32:36 +0900  Paul Hyunil <paul.hyunil@lge.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Support subtitle when track subtype is fourcc_subt
	  https://bugzilla.gnome.org/show_bug.cgi?id=752655

2015-07-20 16:59:40 +0800  Song Bing <b06498@freescale.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Set timestamp when queue buffer.
	  Should set timestamp when queue buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752618

2015-07-20 11:09:20 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: Log GDK GL error when failling creating GdkGLContext

2015-07-18 17:19:18 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/qtitem.cc:
	  glcontext: fix get_current_gl_api on x11/nvidia drivers
	  They require to get_proc_address some functions through the
	  platform specific {glX,egl}GetProcAddress rather than the default
	  GL library symbol lookup.

2015-07-18 17:19:18 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  glcontext: fix get_current_gl_api on x11/nvidia drivers
	  They require to get_proc_address some functions through the
	  platform specific {glX,egl}GetProcAddress rather than the default
	  GL library symbol lookup.

2015-07-17 16:00:01 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtkgstglwidget: Cleanup unused private member
	  new_buffer has been moved to base class. Also cleanup
	  the properties comment, which are also all moved into
	  the base class.

2015-07-17 15:57:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkbasesink.c:
	  gtksink: "widget" must be access from main thread
	  Document that "widget" property must be accessed from the
	  main thread (where GTK is running). This is the same for
	  state transition on these elements. It is very natural to
	  do so un GTK applications.

2015-07-17 15:08:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtkglsink: Don't leak vertex array and buffers
	  This is now possible since reset is always called from the
	  main thread.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-17 14:36:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	  gtkgstbasewidget: Fix black frame on resize
	  This is solved by only applying the new format when the next
	  buffer is to be rendered and on the GTK thread.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-17 13:05:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	  gtkgstbasewidget: Pass already parsed VideoInfo
	  As the base sink already parse the caps into VideoInfo it
	  makes sense to pass in VideoInfo to the widget instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-16 16:49:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	  gtkglsink: Port to GstGtkBaseSink base class
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-16 16:00:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtksink.c:
	* ext/gtk/gstgtksink.h:
	  gtksink: Port to GstGtkBaseSink
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-16 15:59:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/Makefile.am:
	* ext/gtk/gstgtkbasesink.c:
	* ext/gtk/gstgtkbasesink.h:
	  gtkbasesink: Create a base class
	  This contains all the common code between the gtkglsink and
	  gtksink.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-16 14:30:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstglwidget.h:
	  gtkglsink: Port to GtkGstBaseWidget
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-16 12:55:11 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtksink.c:
	* ext/gtk/gstgtksink.h:
	* ext/gtk/gtkgstwidget.c:
	* ext/gtk/gtkgstwidget.h:
	  gtksink: Port to GtkGstBaseWidget
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-16 12:51:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/Makefile.am:
	* ext/gtk/gtkgstbasewidget.c:
	* ext/gtk/gtkgstbasewidget.h:
	  gtk: Add GtkGstBaseWidget
	  This is a "pseudo" base class. Basically it's a shared instance
	  and class structure and a shared set of function between the
	  two widget. It cannot have it's own type like normal base class
	  since the one instance will implement GtkGLArea while the other
	  implements GtkDrawingAreay. To workaround this, the parent instance
	  and class is a union of both.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752441

2015-07-15 17:35:22 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtkgstglwidget: Remove unused gl_caps

2015-07-15 16:56:33 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtksink.c:
	  gtksink: Create a window if the widget is unparented
	  The same way as it's now done with the gtkglsink, create a top
	  level window if the widget is not parented.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751104

2015-07-15 14:35:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtksink.c:
	  gtksink: Ensure the copy pasted code remains the same
	  Move back the default property at the same place they are in the
	  other sink. This helps when using a diff viewer to synchronized
	  this unfortunate copy paste.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751104

2015-07-15 14:32:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gstgtksink.c:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstwidget.c:
	  gtk: Fix race between queue_draw and destroy
	  In GTK dispose can be called before the last ref is reached. This
	  happens when you close the container window. The dispose will be
	  explicitly called, and destroyed notify will be fired. This patch
	  fixes this race by properly tracking the widget state.
	  In the sink, we now set the widget pointer to NULL, so the widget
	  will properly get created again if you set your pipeline to NULL
	  state after the widget was destroy, and set it back to PLAYING.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751104

2015-07-16 15:12:17 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	* tests/check/elements/rtpmux.c:
	  rtpmux: handle different ssrc's on sinkpads
	  Do this by not putting the ssrc from the src pads in the caps used to
	  probe other sinkpads, and then  intersecting with it later.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752491

2015-07-16 17:19:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/avi/gstavimux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	  Update mailing list address from sourceforge to freedesktop

2015-07-15 13:44:52 +0300  Dimitrios Christidis <dchristidis@mykolab.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix trailing '*' displayed with some text subtitles
	  The subtitle buffer we push out should not include a NUL terminator
	  as part of the data, we just add such a terminator for safety, but
	  it should not be included in the buffer size.
	  A NUL terminator is not valid UTF-8, so checks will fail if it's
	  included in the size, and the NUL will be replaced by the fallback
	  character specified when converting, i.e. '*'.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752421

2015-07-15 18:23:05 +0200  Wim Taymans <wtaymans@redhat.com>

	* ext/pulse/pulsedeviceprovider.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulse: add properties to GstDevice
	  Add the extra properties we get from pulse to the GstDevice we expose
	  with the device monitor

2015-07-15 11:47:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gtkgstwidget.c:
	  gtkgstwidget: Add missing break in get_property

2015-07-15 11:44:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gstgtksink.h:
	  gtksinks: Remove undefined private structure
	  The classes contains a private structure which are not defined,
	  hence unused.

2015-07-15 17:20:20 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiowsincband.c:
	  audiofx: Fix typo in example pipelines
	  Fix typo in example pipelines of audiowsincband and audioinvert.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752416

2015-04-15 18:27:04 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: add a "format-location" signal that allows better control over filenames
	  In certain applications, splitting into files named after a base
	  location template and an incremental sequence number is not enough.
	  This signal gives more fine-grained control to the application to
	  decide how to name the files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750106

2015-04-15 20:13:27 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudiosrc: no resampling on OS X
	  Unlike Remote IO, AUHAL doesn't have built-in resampling
	  for sources -- confirmed by Core Audio engineer Doug Wyatt:
	  http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-04-15 18:29:14 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudiosrc: avoid get_channel_layout
	  This only produces a warning and serves no purpose.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-04-07 15:40:14 +0530  Arun Raghavan <arun@centricular.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: Avoid making a duplicate structure in caps for mono/stereo case
	  For 1ch or 2ch devices, we just need to set the caps to allow both
	  options since CoreAudio will up/downmix appropriately.
	  Also fixes the condition for the 2ch case to be exact, rather than at
	  least 2 channels since the downmix will not take place in the >stereo
	  case.

2015-04-06 16:22:34 +0530  Arun Raghavan <arun@centricular.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: Don't set the format on an initialized AudioUnit
	  We need to initialize the AudioUnit early to be able to probe the
	  underlying device, but according to the AudioUnitInitialize() and
	  AudioUnitUninitialize() documentation, format changes should be done
	  while the AudioUnit is uninitialized. So we explicitly uninitialize the
	  AudioUnit during a format change and reinitialize it when we're done.

2015-04-06 15:55:59 +0530  Arun Raghavan <arun@centricular.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudio: Minor spelling fix (unitialize -> uninitialize)

2015-03-21 20:34:25 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudio: Fix lockup in _audio_unit_property_listener
	  _audio_unit_property_listener is called either from a Core Audio thread
	  or as a result of a Core Audio API (e.g. AudioUnitInitialize)
	  from our own thread. In the latter case, osxbuf can be already locked
	  (GStreamer's mutex is not recursive).
	  We introduce the flag cached_caps_valid and use it instead of nullifying
	  cached_caps when we cannot lock on osxbuf.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-03-12 12:15:12 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: Invalidate cached caps on format change
	  Listen for changes in hardware stream format and channel layout, and
	  invalidate cached caps (since they contain the preferred caps).
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-03-09 23:34:06 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxaudiosrc.h:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiocommon.h:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: Overhaul of probing caps
	  - Probing caps is unified between source and sink
	  - Hardware stream format is now reported as preferred capabilities
	  (dynamically updated when hardware configuration changes)
	  - Get hardware channel layout from Remote IO just like from HAL
	  - More comprehensive mapping between AudioChannelLabel and
	  GstAudioChannelPosition
	  - Support for unpositioned channel layouts
	  - Announce stereo-mono upmixing/downmixing in caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-03-09 23:15:56 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: AudioUnitInitialize on open
	  Call AudioUnitInitialize upon open. Otherwise, we cannot get
	  (hardware) stream format nor channel layout from the outer scope.

2015-07-12 14:27:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvp8depay.c:
	  rtp: depayloaders: implement process_rtp_packet() vfunc
	  For more optimised RTP packet handling: means we don't
	  need to map the input buffer again but can just re-use
	  the mapping the base class has already done.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750235

2015-05-27 19:19:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtpvrawdepay: implement process_rtp_packet() vfunc
	  For more optimised RTP packet handling: means we don't
	  need to map the input buffer again but can just re-use
	  the map the base class has already done.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750235

2015-07-10 14:01:43 +0200  Edward Hervey <edward@centricular.com>

	* ext/qt/qtitem.cc:
	  configure/qt: Fix build without Qt5X11Extras

2015-07-06 23:10:51 +1000  Matthew Waters <matthew@centricular.com>

	* ext/qt/.gitignore:
	* ext/qt/Makefile.am:
	* ext/qt/gstplugin.cc:
	* ext/qt/gstqsgtexture.cc:
	* ext/qt/gstqsgtexture.h:
	* ext/qt/gstqtsink.cc:
	* ext/qt/gstqtsink.h:
	* ext/qt/qtitem.cc:
	* ext/qt/qtitem.h:
	* tests/examples/qt/qml/.gitignore:
	* tests/examples/qt/qml/main.cpp:
	* tests/examples/qt/qml/main.qml:
	* tests/examples/qt/qml/play.pro:
	* tests/examples/qt/qml/qml.qrc:
	  new qt5 qml GL video sink
	  Very much in the same spirit as the Gtk GL sink
	  Two things are provided
	  1. A QQuickItem subclass that renders out RGBA filled GstGLMemory
	  buffers that is instantiated from qml.
	  2. A sink element that will push buffers into (1)
	  To use
	  1. Declare the GstGLVideoItem in qml with an appropriate
	  objectName property set.
	  2. Get the aforementioned GstGLVideoItem from qml using something like
	  QQmlApplicationEngine engine;
	  engine.load(QUrl(QStringLiteral("qrc:/main.qml")));
	  QObject *rootObject = engine.rootObjects().first();
	  QQuickItem *videoItem = rootObject->findChild<QQuickItem *> ("videoItem");
	  3. Set the videoItem on the sink
	  https://bugzilla.gnome.org/show_bug.cgi?id=752185

2015-07-10 00:13:32 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix indention

2015-07-09 23:59:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Always estimate DTS from the current clock time
	  Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
	  we would produce wrong DTS. As now the estimated DTS is based on the clock,
	  don't store it in the jitterbuffer items as it would otherwise be used in the
	  skew calculations and would influence the results. We only really need the DTS
	  for timer calculations.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-07-09 09:26:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/.gitignore:
	  gitignore: ignore rtph263 test

2015-07-09 13:03:23 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstwidget.c:
	  gtk: add to the generic/states test

2015-06-17 09:36:57 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	  GstGtkGLSink: Ensure widget has a toplevel parent
	  Checking for a parent is not enough, it must have a toplevel one.
	  If widget has no toplevel parent then add it in a GtkWindow, that
	  make it usable from gst-launch-1.0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751104

2015-06-17 09:36:40 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	  GstGtkGLSink: Post error if widget gets destroyed
	  https://bugzilla.gnome.org/show_bug.cgi?id=751104

2015-06-16 16:21:26 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	  GstGtkGLSink: fix possible warning in finalize
	  If the element is finalized before going in READY state
	  the widget could still be NULL.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751104

2015-07-08 23:47:44 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2
	  Replace static constants with macros to make gcc happy
	  CC       elements/elements_rtpjitterbuffer-rtpjitterbuffer.o
	  elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant
	  static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND;
	  ^
	  elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant
	  static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000;
	  ^
	  elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant
	  PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000;

2015-07-08 23:40:45 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: run indent and fix some comments
	  Fix indent on this file and break some comment lines into two to make
	  it fit 80 chars per line

2015-07-08 15:02:24 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: rework segment event handling for adaptive streaming
	  When a new time segment is received upstream is going to restart
	  with a new atom. Make the neededbytes and todrop variables
	  reflect that to avoid waiting too much or dropping the
	  initial bytes that contain the header.

2015-07-08 12:35:55 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: push data from adapter before starting new segment
	  The adapter might have data remaining from the previous segment,
	  push it all before clearing the adapter and starting a new segment.
	  It can accumulate data if it had pushed and got not-linked, returning
	  immediately without processing all the data. Before starting a new
	  segment this data should be handled.

2015-07-08 19:59:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-07-08 21:08:36 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix gap-time calculation and remove "late"
	  The amount of time that is completely expired and not worth waiting for,
	  is the duration of the packets in the gap (gap * duration) - the
	  latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
	  that we make a "multi-lost" packet for.
	  The "late" concept made some sense in 0.10 as it reflected that a buffer
	  coming in had not been waited for at all, but had a timestamp that was
	  outside the jitterbuffer to wait for. With the rewrite of the waiting
	  (timeout) mechanism in 1.0, this no longer makes any sense, and the
	  variable no longer reflects anything meaningful (num > 0 is useless,
	  the duration is what matters)
	  Fixed up the tests that had been slightly modified in 1.0 to allow faulty
	  behavior to sneak in, and port some of them to use GstHarness.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738363

2015-06-30 11:21:31 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected"
	  This reverts commit 05bd708fc5e881390fe839803b53144393d95ab0.
	  The reverted patch is wrong and introduces a regression because there
	  may still be time to receive some of the packets included in the gap
	  if they are reordered.

2015-07-07 23:53:02 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: flush samples before adding more from moof
	  Avoids accumulating all samples from a fragmented stream that could
	  lead to a 'index-too-big' error once it goes over 50MB of data. It
	  could reach that before 2h of playback so it doesn't take that long.
	  As upstream elements are providing data in time format they should
	  be the ones that have more information about the full media index
	  and should be able to seek if possible.

2015-07-07 23:56:12 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: rename upstream_newsegment to upstream_format_is_time
	  upstream_newsegment isn't really clear on what it means, it is set
	  to TRUE when the upstream element sends a segment in TIME format, so
	  rename it to be more clear about it.
	  It is important to know this because it means that upstream has
	  a notion of time and qtdemux is likely being driven by an upstream
	  element that is reading from a higher level abstraction than a file,
	  such as a DASH, MSS or DLNA element.

2015-07-07 21:31:08 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix leak by flushing previous sample info from trak
	  In fragmented streaming, multiple moov/moof will be parsed and their
	  previously stored samples array might leak when new values are parsed.
	  The parse_trak and callees won't free the previously stored values
	  before parsing the new ones.
	  In step-by-step, this is what happens:
	  1) initial moov is parsed, traks as well, streams are created. The
	  trak doesn't contain samples because they are in the moof's trun
	  boxes. n_samples is set to 0 while parsing the trak and the samples
	  array is still NULL.
	  2) moofs are parsed, and their trun boxes will increase n_samples and
	  create/extend the samples array
	  3) At some point a new moov might be sent (bitrate switching, for example)
	  and parsing the trak will overwrite n_samples with the values from
	  this trak. If the n_samples is set to 0 qtdemux will assume that
	  the samples array is NULL and will leak it when a new one is
	  created for the subsequent moofs.
	  This patch makes qtdemux properly free previous sample data before
	  creating new ones and adds an assert to catch future occurrences of
	  this issue when the code changes.

2015-07-07 16:46:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix index size check and debug message
	  It is allocating samples_count + n_samples, not only n_samples

2015-07-08 17:02:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Calculate receive time if we don't have any
	  This is required to properly schedule packet loss timers and make
	  sure all our calculations work properly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-07-08 15:13:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations
	  That is, handle DTS==GST_CLOCK_TIME_NONE correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-07-08 20:31:42 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix event leak
	  when seek fails in avidemux, event is not being freed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752117

2015-07-08 12:02:22 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph263depay.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtph263.c:
	  rtph263depay: Make sure payload is large enough
	  Plus new unit test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752112

2015-07-08 08:59:49 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/rtp/gstrtpklvdepay.c:
	  rtpklvdepay: fix printf format compiler warning
	  v_len is of type guint64, but while print the value(16 + len_size + v_len)
	  G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT
	  https://bugzilla.gnome.org/show_bug.cgi?id=752100

2015-07-07 20:25:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-rtp.xml:
	  docs: add new RTP elements to docs

2015-07-07 20:07:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: add basic unit test for KLV payloading
	  Also make it so that the mtu is always set if specified, not
	  only in case of the rather weird bufferlist test code path.
	  This allows us to easily make the payloader fragment a payload
	  across multiple output packets by setting a small MTU on it.

2015-07-07 19:58:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvdepay.h:
	  rtpklvdepay: improve start detection and handle fragmented KLV units

2015-07-05 20:25:10 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvdepay.h:
	  rtp: add SMPTE 336M KLV metadata depayloader
	  http://tools.ietf.org/html/rfc6597

2014-08-09 10:08:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpklvpay.c:
	* gst/rtp/gstrtpklvpay.h:
	  rtp: add SMPTE 336M KLV metadata payloader
	  http://tools.ietf.org/html/rfc6597

2015-07-07 16:59:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/properties.h:
	* gst/matroska/matroska-mux.c:
	* gst/rtpmanager/rtpsource.c:
	  docs: fix "Symbol name not found at the start of the comment block"
	  Add symbols or change comment into a regular comment.

2015-07-07 16:58:53 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audioparsers/gstamrparse.h:
	  docs: remove outdated doc strings

2015-07-03 23:10:40 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  docs: add missing plugins and ensure master doc is sorted

2015-07-07 15:54:41 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  Revert "imagefreeze: Remove impossible error condition"
	  This reverts commit d46631c5c7312ad613397f8238c7a9714ae3ae94.
	  pad only handle EOS events but not EOS flow, and will push the buffer again
	  resulting in an assertion error. So we should not handle the buffer
	  and return EOS flow.

2015-07-07 15:50:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpg729depay.c:
	  rtpg729depay: unmap rtp buffer in error path

2015-07-07 15:48:40 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: fix buffer leak
	  The handle_buffer vfunc takes ownership of the input buffer.
	  Fixes elements/rtp-payloading under valgrind.

2015-07-02 08:52:43 +0200  Tobias Mueller <muelli@cryptobitch.de>

	* gst/goom/goom_core.c:
	  goom: Initialised variables to remove compiler warnings
	  goom_core.c: In function 'goom_update':
	  goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur);
	  ^
	  goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur);
	  ^
	  https://bugzilla.gnome.org/show_bug.cgi?id=752053

2015-07-07 09:18:39 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: fix indentation

2015-07-06 19:11:00 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: Fix uninitialized variable compiler error
	  endpos variable does not correctly understand in the
	  4.6.3 GCC version. So compile error appears when we do
	  compile rtph261pay using jhbuild.
	  This patch is fixed the compile error in 4.6.3 GCC version.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751985

2015-07-06 19:33:35 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gtkgstglwidget.c:
	  gtkglsink: Release the widget lock when trying to get the GL context
	  Otherwise we might be waiting for the lock on the main loop (for
	  example in the ->render vmethod) and thus we will deadlock.

2014-11-12 12:08:58 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Handle seek flags properly
	  Allows for non-keyframe seeks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738570

2015-02-24 10:50:52 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid looping reading the 'moof' atom forever
	  It gets stuck if it only finds a moof and no mfra/mfro or moov
	  atoms. Skip the moof to continue the parsing to have it either
	  play or error out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745089

2015-06-26 13:24:17 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/flac/gstflacdec.c:
	  flacdec: improve error handling
	  for files which have corrupted header, libflac is not able to
	  process the metadata properly. We just try to ignore the error
	  and continue with the processing, since metadata parsing is not
	  making much of a difference to libflac
	  https://bugzilla.gnome.org/show_bug.cgi?id=751334

2015-07-06 20:16:38 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* sys/ximage/ximageutil.c:
	  ximagesrc: add meta transform function
	  ximage metadata can't be transformed or copied, but provide an empty
	  transformation function instead of NULL to allow unconditional calling
	  of metas' transform functions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751778

2014-06-16 16:14:28 +0200  Stian Selnes <stian.selnes@gmail.com>

	* gst/rtp/gstrtph263pdepay.c:
	  rtph263pdepay: init debug category
	  https://bugzilla.gnome.org/show_bug.cgi?id=752012

2014-06-20 10:59:14 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpv8depay: ignore reserved bit in payload descriptor
	  Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that:
	  R: Bit reserved for future use.  MUST be set to zero and MUST be
	  ignored by the receiver.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751929

2015-07-04 20:56:42 +0200  Stian Selnes <stian@pexip.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: rtph261depay: Add documentation
	  https://bugzilla.gnome.org/show_bug.cgi?id=751982

2015-07-03 21:58:14 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From f74b2df to 9aed1d7

2015-07-03 14:29:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: Fix compiler warning
	  gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init':
	  gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable]
	  GObjectClass *gobject_class;

2015-07-03 14:03:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph261depay.c:
	  rtph261depay: Let the base class push the buffer so it can deal with the flow return

2015-07-03 14:11:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: Remove unused adapter

2015-07-03 13:17:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpspeexpay.c:
	  speexpay: Directly attach payload to the output buffer instead of copying it

2015-07-03 13:07:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpsbcpay.c:
	  sbcpay: Attach payload directly to the output instead of copying

2014-12-01 14:18:40 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261depay.h:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph261pay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtp: add H.261 RTP payloader and depayloader
	  Implementation according to RFC 4587.
	  Payloader create fragments on MB boundaries in order to match MTU size
	  the best it can. Some decoders/depayloaders in the wild are very strict
	  about receiving a continuous bit-stream (e.g. no no-op bits between
	  frames), so the payloader will shift the compressed bit-stream of a
	  frame to align with the last significant bit of the previous frame.
	  Depayloader does not try to be fancy in case of packet loss. It simply
	  drops all packets for a frame if there is a loss, keeping it simple.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751886

2015-07-03 12:18:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpmpvdepay.c:
	  rtpmpvdepay: Don't forget to unmap the input buffer

2015-07-03 12:14:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpmpvpay.c:
	  rtpmpvpay: Create buffer lists instead of pushing each buffer individually

2015-07-03 12:03:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpmpapay.c:
	  rtpmpapay: Use buffer lists instead of pushing each fragment individually

2015-07-03 10:51:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpmp4apay.c:
	  rtpmp4apay: Create buffer lists and don't copy payload memory

2015-06-29 16:14:18 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Consider timers len to compare with RTP_MAX_DROPOUT
	  When there are a lot of small gaps, we can consider that there is
	  a big gap (too losses) to reset the buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751636

2015-06-29 15:53:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: If possible, always update the current time before looping over all timers
	  If we have a clock, update "now" now with the very latest running time we have.
	  If timers are unscheduled below we otherwise wouldn't update now (it's only updated
	  when timers expire), and also for the very first loop iteration now would otherwise
	  always be 0.
	  Also the time is used for the timeout functions, e.g. to calculate any times
	  for the next timeouts and we would otherwise pass too old times there.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751636

2015-07-02 14:34:57 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: fix memory leak
	  tmp needs to be freed before going out of scope in 'done'.
	  CID #1308954

2015-07-02 12:23:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Generate buffer lists and attach the payload directly instead of copying it

2015-07-02 09:48:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph263pdepay.c:
	  rtph263pdepay: Simplify code a bit and do less direct memcpy and let GstBuffer do that for us

2015-07-02 09:17:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	  rtph263pay: Stop using an adapter and directly use the buffer
	  We always pushed one buffer into the adapter, then handled exactly that one
	  buffer and flushed it from the adapter. Now also don't memcpy() the actual
	  payload but just attach the input buffer's data to the output buffer.
	  This code still needs some serious refactoring/rewriting.

2015-07-01 21:57:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgsmpay.c:
	  rtpgsmpay: Remove non-existing includes for now
	  git add -p mistake.

2015-07-01 19:29:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Use the return value of gst_buffer_append()

2015-07-01 19:19:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgsmpay.c:
	  rtpgsmpay: Attach payload to the output buffer instead of copying it

2015-07-01 17:58:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Attach payload directly to output buffers instead of copying

2015-07-01 17:43:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpg723pay.c:
	  rtpg723pay: Attach payload buffer to the output instead of copying

2015-07-01 17:30:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpdvdepay.c:
	  rtpdvdepay: Map the output buffer once instead of once every 80 bytes

2015-07-01 21:46:46 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix return type of index_entry_offset_search()
	  It's a compare function and may return a negative value,
	  so should for correctness and consistency return a signed
	  integer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751780

2015-07-01 14:12:57 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refactor handle_next_buffer
	  The goal of this patch is making handle_next_buffer function
	  more readable avoiding unnecesary gotos and adding other
	  cosmetic changes.

2015-07-01 15:40:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpac3pay.c:
	  rtpac3pay: Attach the payload to the output buffer instead of copying it
	  Might also want to produce buffer lists here if needed.

2015-07-01 15:38:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	  rtp: Fix indention

2015-07-01 12:37:11 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-VP8-OPUS.sh:
	* tests/examples/rtp/server-VTS-VP8-ATS-OPUS.sh:
	  rtp: Add examples with VTS/ATS for VP8/OPUS
	  Let's have an example with modern codecs.

2015-06-30 18:11:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()

2015-06-30 14:06:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvp8depay.c:
	  vp8depay: Don't lock/map every non-keyframe buffer twice
	  Just copy the complete header instead of first looking at the first byte
	  and then at the remaining 10 bytes.

2015-06-29 16:05:44 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: document fallthrough cases
	  Pacify coverity and document fallthrough cases in switch statements.
	  CID #1308948, #1308947, #1308946

2015-06-29 10:36:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Revert "rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout"
	  This reverts commit 0c21cd7177ea883c710999147ddcedb19004d182.
	  If we have multiple immediate timers, we want to first handle the one with the
	  lowest sequence number... which would be broken now.
	  Instead of this we should just use a GSequence for the timers, and have them
	  sorted first by timestamp, and for equal timestamps by sequence number. Then
	  we would always only have to take the very first timer from the list and never
	  have to look at any others.

2015-06-29 10:14:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout
	  If we have lots of such immediate timeouts, we would otherwise have quadratic
	  runtime in the number of timeouts.

2015-06-19 18:01:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: sticky events are sent automatically from the pad
	  No need to send them explicitly from the element
	  https://bugzilla.gnome.org/show_bug.cgi?id=751240

2015-06-19 18:00:40 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: make sure to push sticky events before adding pad
	  It allows the caps to be set on the pad before being added for
	  dynamic autoplugging to work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751240

2015-06-26 00:05:29 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property
	  Enable to use new ntp-time-source property of rtpbin
	  https://bugzilla.gnome.org/show_bug.cgi?id=751496

2015-06-25 23:19:58 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin/session: fix description
	  https://bugzilla.gnome.org/show_bug.cgi?id=751496

2015-06-25 10:57:25 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/matroska/matroska-demux.c:
	* tests/examples/shapewipe/shapewipe-example.c:
	  docs: decodebin2 -> decodebin

2015-06-25 10:47:06 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: update example pipeline
	  Update reference to decodebin2 to decodebin

2015-06-25 10:45:35 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: remove dead assignments
	  Values in fields_required and same_buffer are overwritten before used. Removing
	  assignment

2015-06-25 10:06:07 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/Makefile.am:
	* ext/mikmod/Makefile.am:
	* ext/mikmod/README:
	* ext/mikmod/drv_gst.c:
	* ext/mikmod/gstmikmod.c:
	* ext/mikmod/gstmikmod.h:
	* ext/mikmod/mikmod_reader.c:
	* ext/mikmod/mikmod_types.c:
	* ext/mikmod/mikmod_types.h:
	* m4/Makefile.am:
	* m4/libmikmod.m4:
	* win32/MANIFEST:
	* win32/vs8/libgstmikmod.vcproj:
	  mikmod: remove ancient unported plugin
	  This hasn't been touched in 11 years, and
	  clearly no one's been missing it.

2015-06-23 20:15:13 +0900  Gilbok Lee <gilbok.lee@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: does not detect orientation
	  Most files don't contain the values for transposing the coordinates
	  back to the positive quadrant so qtdemux was ignoring the rotation
	  tag. To be able to properly handle those files qtdemux will also ignore
	  the transposing values to only detect the rotation using the values
	  abde from the transformation matrix:
	  [a b c]
	  [d e f]
	  [g h i]
	  https://bugzilla.gnome.org/show_bug.cgi?id=738681

2015-06-25 00:04:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.5.2 ===

2015-06-24 23:30:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.5.2

2015-06-24 22:56:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2015-06-24 11:15:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/nl.po:
	  po: Update translations

2015-06-23 18:42:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/qtmux.c:
	  qtmux: Correctly test each segments
	  In presence of gaps, qtdemux will emit multiple segments. The
	  second segment start should match the CTTS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 17:54:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Correctly calculate the elst media start
	  The media start has nothing to do with the shift we have applied
	  but with the value of the first PTS. This is defined as:
	  Dt(0) = 0
	  Ct(0) = Dt(0) + CTTS(0)
	  So the media start is always the first CTTS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 11:49:32 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: accumulate previous edts entries into segment.base
	  Allows playing edts editted files with proper synchronization of
	  streams. This patch fixes the regression introduced by
	  bf95f93c0189aa04f18e264b86b6527e431c5d53 that was added to fix
	  segment seeks handling.
	  Having the accumulated_base separated from the main segment.base
	  allows handling both segment seeks and edts editted files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 00:56:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: improve some debug messages
	  Those messages are about the stream, use the pad as the
	  debug object to make it clear from the logs
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-22 22:22:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: store last_dts of the first buffer
	  Buffers need not to start at running-time 0 so the last_dts needs
	  to be the value of the first buffer's dts as it is used to compute
	  the duration of the buffers. If it was left at 0 the first buffer
	  would have a larger duration when it shouldn't
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 17:11:57 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix possible memory leak
	  when buffer is stored to seektable, and stop gets called due to
	  corrupt flac file, then the seektable is not being released
	  https://bugzilla.gnome.org/show_bug.cgi?id=751364

2015-06-23 16:28:40 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  Revert "splitmuxsink: Mask async-start/done while switching files."
	  This reverts commit d61e5393f110ed482815d77807245d78b52eff46.
	  Causes failures muxing larger GOP sizes for some reason. Reverting
	  while I figure it out

2015-06-18 23:22:06 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Fix startup and shutdown races.
	  Fix 2 startup races when things happen too quickly, and 1
	  at shutdown by holding a ref to the pads in use until the
	  loop functions exit.
	  Handle errors activating file parts and publish them on
	  the bus.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750747

2015-06-18 09:26:13 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Mask async-start/done while switching files.
	  Sometimes, extra async-start/done from the internal sink
	  while the element is still starting up can cause splitmuxsink
	  to stall in PAUSED state when it has been set to PLAYING
	  by the app. Drop the child's async-start/done messages while
	  switching, so they don't cause state changes at the
	  splitmuxsink level.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750747

2015-06-15 16:12:10 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Use gst_video_multiview_guess_half_aspect()
	  Use the gst_video_multiview_guess_half_aspect() utility function
	  to set the half-aspect flag (or not) on stereoscopic frame-packed
	  videos.

2015-06-15 16:10:37 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Move multiview caps calculations, add half-aspect heuristics
	  Move the multiview caps calculations to the configure_stream()
	  function, so the rest of the video info is available, and
	  use the gst_video_multiview_guess_half_aspect() function to
	  determine if the half-aspect flag should be set on frame-packed
	  video.

2015-06-18 16:06:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add cslg support
	  The cslg atom provide information about the DTS shift. This is
	  needed in recent version of ctts atom where the offset can be
	  negative. When cslg is missing, we parse the CTTS table as proposed
	  in the spec to calculate these values.
	  In this implementation, we only need to know the shift. As GStreamer
	  cannot transport negative timestamps, we shift the timestamps forward
	  using that value and adapt the segment to compensate. This patch also
	  removes bogus offset of ctts_soffset, this offset shall be included
	  in the edit list.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751103

2015-06-19 18:37:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/qtmux.c:
	  qtmux: Test gaps at start of stream
	  https://bugzilla.gnome.org/show_bug.cgi?id=751242

2015-06-19 18:40:43 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use PTS to figure-out presence of gaps
	  We need to look at the presentation timestamp in order to conclude if
	  there is a gap at the start of a stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751242

2015-06-19 16:45:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Set edit list to compensate DTS shift
	  We shift DTS forward to avoid negative timestamps which cannot be
	  represented with version 0 of the CTTS table. To stick with that
	  version (backward compatibility), the spec recommend using an
	  edit list entry to move back the presentation time to where it
	  should be.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751242

2015-06-22 14:35:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Insert AVC end of sequence
	  This FLV specific mark is needed to prevent Flow Player (most likely
	  all Flash base player) from going into buffering state when near EOS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751320

2015-06-22 13:05:29 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	  matroska: remove useless check
	  No need to check for context availability while freeing. We are inside
	  inside a code block with a condition that dereferences context.
	  if (context->type == 0 ...
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 19:35:57 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/matroska/lzo.c:
	  lzo: fix memory leak
	  the opened file is not being closed during test, which will result
	  in memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 19:30:58 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* ext/mikmod/mikmod_reader.c:
	  mikmod_reader: Possible null pointer dereference:
	  gst_reader variable is being used before actually checking if it
	  allocated properly
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 19:45:14 +0900  Sangkyu Park <sk1122.park@samsung.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Minor clean-up
	  1. Fix the code which is wrong coding style.
	  2. Fix a typing error of comment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751316

2015-06-22 11:28:13 +0200  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Do not try to push NULL buffers
	  If update_receiver_stats() fails, we can't really do anything with this buffer
	  anymore and have to drop it. This happens if there's a big seqnum
	  discontinuity for example.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751311

2015-06-22 13:10:02 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: trivial cleanup
	  trivial patch to add proper ( while checking for if(G_UNLIKELY())
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 13:16:08 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: initialize size variable
	  size can be used in cleanup without being initialized. Hence
	  setting it to 0 when declaring
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 13:13:29 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: initialze bpf variable
	  bpf variable might be used in cleanup without being intialized.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-19 14:50:59 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtprtxqueue.c:
	  rtprtxqueue: reverse pending list before pushing buffers
	  With this we send the RTX buffers in the same order
	  that they were requested.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751297

2015-06-21 19:22:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix DTS validity check
	  This check was up-side-down, causing a bad timestamp at start
	  and then all timestamp being delayed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751298

2015-06-17 15:19:47 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_types.c:
	  cslg: Add Composition Shift Least Greatest Atom
	  This simply add fourcc and dump function for the cslg Atom.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751103

2015-06-17 15:18:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/qtdemux_dump.c:
	  ctts_dump: Fix signess issues
	  It didn't bug, but use correct signess in traces. The number of
	  entries is unsigned while the offset can be signed according to
	  recent spec.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751103

2015-06-16 17:48:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 6015d26 to f74b2df

2015-06-16 11:43:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: gst_rtp_buffer_ext_timestamp() modifies its first argument, keep a copy around

2015-06-16 10:30:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Compare ext RTP times, not plain RTP time and ext RTP time when calculating elapsed time
	  Otherwise all RTP times after a wraparound would be considered as going
	  backwards, they will always be smaller than the ext RTP time.

2015-06-15 21:32:43 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtkglwidget: Const'ify another array

2015-06-15 21:29:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtkglwidget: Calculate the viewport size ourselves
	  Getting the current viewport and modifying it relatively will produce an
	  interesting feedback loop during widget resizing. Over a few frames we
	  will gradually move the viewport a bit until it converged again, adding
	  unnecessary additional borders at the top and left.

2015-06-15 21:24:01 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstwidget.c:
	  gtk: Use the display width/height for the widget's preferred width/height

2015-06-15 20:45:11 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gstgtksink.c:
	* ext/gtk/gtkgstwidget.c:
	  gtksink: Add support for xRGB/BGRx

2015-06-15 20:39:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gstgtksink.c:
	* ext/gtk/gtkgstwidget.c:
	  gtk: Cairo color formats are in native endianness, GStreamer's in memory order
	  CAIRO_FORMAT_ARGB32 is ARGB on big endian and BGRA on little endian.

2015-06-15 20:35:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gstgtksink.c:
	* ext/gtk/gstgtksink.h:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstwidget.c:
	  gtk: Implement ignore-alpha property and enable it by default

2015-06-15 20:13:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtksink.c:
	  gtk: Sync properties from the sink to the widget upon widget creation

2015-06-15 19:25:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: The default rtp-profile should be AVP, not AVPF

2015-06-15 18:28:37 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gstgtksink.c:
	* ext/gtk/gstgtksink.h:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstwidget.c:
	  gtk: implement pixel and display aspect ratio handling

2015-06-15 14:32:21 +0900  Sangkyu Park <sk1122.park@samsung.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Minor cleanup
	  1. Add Null check in 'free_item' function.
	  2. Fix a typing error of comment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750965

2015-06-15 14:35:35 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: silence unused variable warnings for unsupported winsys'

2015-06-15 14:33:08 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: implement basic wayland GL support

2015-06-12 17:44:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvmux.c:
	  flmux: Make sure best_time is initialized

2015-06-12 23:29:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpsession.h:
	  rtpbin/session: Add new ntp-time-source property and deprecate use-pipeline-clock property
	  The new property allows to select the time source that should be used for the
	  NTP time in RTCP packets. By default it will continue to calculate the NTP
	  timestamp (1900 epoch) based on the realtime clock. Alternatively it can use
	  the UNIX timestamp (1970 epoch), the pipeline's running time or the pipeline's
	  clock time. The latter is especially useful for synchronizing multiple
	  receivers if all of them share the same clock.
	  If use-pipeline-clock is set to TRUE, it will override the ntp-time-source
	  setting and continue to use the running time plus 70 years. This is only kept
	  for backwards compatibility.

2015-04-07 16:03:42 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: test for muxing with DTS outside the segment
	  https://bugzilla.gnome.org/show_bug.cgi?id=740575

2015-06-11 17:26:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Adjust segment according to ctts offset
	  In presence of a CTTS, the segment start/stop must be offset so
	  the segment start/stop include the PTS. This is needed since the
	  PTS cannot be negative in this format. This fixes issues where the
	  running time of the first buffer isn't at the start.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740575

2015-04-03 20:34:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Handle DTS with negative running time
	  As QT works with duration, simply bring back first DTS to 0 and shift
	  forward the PTS of the same amount.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740575

2015-06-10 18:15:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Add negative runtime DTS support
	  This is done by using new feature of the CollectPad clip function
	  which sets the DTS as a gint64 in the collected data. It also simplify
	  the code a bit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740575

2015-06-12 23:06:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Rename some variables and debug output to make more sense
	  Local and remote were mixed up in a few places, and the time we store here is
	  not UNIX time (1970 epoch), but NTP time (1900 epoch) in nanoseconds.

2015-06-12 19:21:10 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: fix latency property query on RemoteIO
	  AudioUnitGetProperty would fail with kParamErr (-50) every time,
	  simply because size wasn't initialized.
	  Now it returns zero latency, but at least it doesn't fail.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750868

2015-06-12 15:39:56 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: Do not try to activate a NULL GLContext
	  At that point in the code nothing guarantees it exists

2015-04-07 14:06:16 +0530  Arun Raghavan <git@arunraghavan.net>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Fix mapping of latency parameters to buffer attributes

2015-06-12 15:17:30 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gstgtksink.c:
	* ext/gtk/gstgtksink.h:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstwidget.c:
	  gtk: implement video aspect-ratio handling
	  For both the software and the GL sink's.
	  Doesn't deal with the pixel-aspect-ratio field at all yet.

2015-06-12 12:40:50 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtksink.c:
	  gtk: fix a couple of typos

2015-06-12 12:29:37 +1000  Matthew Waters <matthew@centricular.com>

	* ext/gtk/gstgtkglsink.c:
	  gtkglsink: reset the context/display in READY_TO_NULL
	  Fixes context propagation in pipelines with upstream GL elements.

2015-06-11 12:41:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/examples/gtk/gtkglsink.c:
	  gstgtk: No need to realize the widget
	  The widget already does that.

2015-06-11 12:38:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/examples/gtk/gtkglsink.c:
	* tests/examples/gtk/gtksink.c:
	  gstgtk: Don't leak the widget
	  g_object_get() returns a ref, gtk_container_add() only ref_sink().
	  That mean we still need to unref afterward. This leak was hiding
	  a reference bug previously present.

2015-06-11 12:10:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtksink.c:
	  gstgtk: Allow doing gst-inspect-1.0 on these elements
	  This patch allow going gst-inspect-1.0 on these elements removing
	  ugly crash that was previously occurring. The method consist of
	  making the widget creation as lazy as possible. This way we don't
	  endup doing gtk_init() before the application. We also ref_sink()
	  the widget, so we don't crash if the parent widget is discarded,
	  and cleanly error out with GL if the widget has no parent window,
	  because calling gtk_widget_realized() can only be done if the widget
	  has been parented to a window).

2015-06-12 01:56:37 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Actually set detected 3D info into output caps.
	  Use the information read from the StereoMode info
	  to configure multiview-mode and multiview-flags in the
	  video caps.

2015-06-11 13:36:54 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Take released-but-not-yet-output bytes into account
	  When deciding whether it's time to switch to a new file, take into
	  account data that's been released for pushing, but hasn't yet
	  been pushed - because downstream is slow or the threads haven't been
	  scheduled.
	  Fixes a race in the unit test and probably in practice - sometimes
	  failing to switch when it should for an extra GOP or two.
	  Also fix a problem in splitmuxsrc where playback sometimes
	  stalls at startup if types are found too quickly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750747

2015-06-11 15:02:44 +0200  Thibault Saunier <tsaunier@gnome.org>

	* ext/gtk/gtkgstglwidget.c:
	  gtk: Do not try to initialize display if we have not have a GLContext yet

2015-06-11 14:58:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/gtk/Makefile.am:
	  gtk: Add missing CFLAGS to example

2014-12-18 17:00:30 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gtk/Makefile.am:
	* ext/gtk/gstgtkglsink.c:
	* ext/gtk/gstgtkglsink.h:
	* ext/gtk/gstgtksink.c:
	* ext/gtk/gstgtksink.h:
	* ext/gtk/gstplugin.c:
	* ext/gtk/gtkgstglwidget.c:
	* ext/gtk/gtkgstglwidget.h:
	* ext/gtk/gtkgstwidget.c:
	* ext/gtk/gtkgstwidget.h:
	* tests/examples/gtk/Makefile.am:
	* tests/examples/gtk/gtkglsink.c:
	* tests/examples/gtk/gtksink.c:
	  Implement gtk sinks
	  two sinks are provided.  gtksink which is a cairo/software based renderer
	  and gtkglsink which utilises the GL support in gtk and gstreamer.

2015-06-11 01:04:51 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  atoms: remove custom gst_buffer_new function in favor of core version
	  Remove a custom specialized version of gst_buffer_new_wrapped by
	  using gst_buffer_new_wrapped_full inside a macro to simplify
	  parameters and give it a more meaningful name.
	  It is only used to create temporary buffers to have its data copied.

2015-06-11 00:14:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  atoms: simplify free form data atoms creation
	  Avoid creating an intermediary buffer or memory area just
	  to copy into an atom's data area.

2015-06-10 22:27:27 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: add AC-3 muxing support
	  Adds AC-3 muxing support. It is defined for mp4 and 3gp formats.
	  One extra feature that was added was the ability to add extension
	  atoms after set_caps as the AC-3 extension atom needs some data
	  that has to be extracted from the stream itself and is not
	  present on caps.

2015-06-10 22:36:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	  qtmux: remove unused type MP4S

2015-06-10 22:29:01 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: remove duplicate attribute value set
	  It is also set a few lines below

2015-06-11 00:22:54 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroska: Implement basic stereoscopic video support
	  Implement support for the packed video formats WebM
	  uses, not all the values that Matroska might use.
	  In practice, it's really hard to find any samples in the
	  wild of any.
	  Supported in both the muxer and demuxer.

2015-06-10 01:26:15 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add basic support for MPEG-A stereoscopic video
	  The MPEG-A format provides an extension to the ISO base media
	  file format to store stereoscopic content encoded with different
	  codecs like H.264 and MPEG-4:2. The stereo video media information(svmi)
	  atom declares the presence and storage method for the video.
	  Stereo video information for MPEG-A can also be supplied through
	  the 'stvi' atom (ref: ISO/IEC_14496-12, ISO/IEC_23000-11), which
	  is not implemented in this patch.
	  Also missing is support for stereo video encoded as separate video tracks
	  for now.
	  Based on a patch by Sreerenj Balachandran <sreerenj.balachandran@intel.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=611157

2015-06-02 16:15:35 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Add tls-database property
	  https://bugzilla.gnome.org/show_bug.cgi?id=750298

2015-06-10 14:33:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
	  The mix between all these in the RTP code is confusing, let's try to be
	  consistent.

2015-06-10 14:49:50 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: clarify negative lost packets in stats
	  Also:
	  - Move notes on units before field documentation.
	  - Unify documentation style.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750653

2015-06-10 06:38:39 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix getter of "ssl-use-system-ca-file"
	  https://bugzilla.gnome.org/show_bug.cgi?id=750298

2015-06-10 09:49:47 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix reverse playback
	  When performing seek, segment->start is being updated with desired_offset,
	  but in case of reverse playback segment->start should be 0 and
	  segment->stop should be updated with desired offset.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750675

2015-01-21 18:09:03 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	  gstv4l2bufferpool: handle -EPIPE from DQBUF to signal EOS
	  The V4L2 decoder signals EOS by returning -EPIPE from DQBUF after the
	  last buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743338

2015-06-06 21:09:19 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add a GTlsInteraction property
	  It can be used for TLS client authentication.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750471

2015-01-09 11:36:11 +0100  Enrico Jorns <ejo@pengutronix.de>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2: Allow scaling in the v4l2*convert element
	  This is inspired of videoscale and videoconvert elements.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742917

2015-06-09 19:02:55 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpmanager: document units of stats and arguments
	  Also, minor spelling and style corrections.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750653

2015-06-09 14:42:27 +0200  Stefan Sauer <ensonic@users.sf.net>

	* Makefile.am:
	  cruft: add the obsolete tmpl dir to cruft-dirs

2015-06-09 11:30:22 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From d9a3353 to 6015d26

2015-06-09 07:04:07 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Fix common version
	  Was accidently downgraded by 87a4884acd8655a6591d735a1d944ecb5ea3de16

2015-06-08 19:11:41 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Also set colorimetry on output devices
	  This completes the code that set the colorimetry on output
	  device.

2015-06-08 19:10:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* common:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add missing SMTP240M matrix
	  This is missing in the doc, but was in the header.

2015-06-08 23:00:16 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom/goom_core.c:
	  goom: possible uninitialized variables warning
	  Build fails with the latest snapshot of gcc-4.9 because param1 and param2 might
	  possibly be used uninitialized. They are set depending on the cases of a switch
	  statement and the compiler sees this as not a complete guarantee.
	  Set them to 0 if the switch statement falls down to the default case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750566#c6

2015-06-08 17:24:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fully implement colorimetry support
	  This fixes wrong mapping for sRGB as in GStreamer sRGB correctly
	  apply to RGB formats, while in V4L2 it's an alias for sYCC. Also
	  add support for the new quantization (range), ycbcr_encoding (matrix)
	  and xfer_func (transfer) enumeration.

2015-06-08 17:01:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/types-compat.h:
	* sys/v4l2/ext/v4l2-common.h:
	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	  v4l2: Update kernel headers to latest from media tree
	  This is the latest from media tree. This should enable more development
	  of the v4l2 elements. This includes new flags requires to fix draining
	  path in decoder, colorimetry and much more.

2015-06-08 23:07:55 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From d37af32 to d9a3353

2015-06-08 19:42:30 +0100  Chris Clayton <chris2553@googlemail.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8depay: potential access beyond end of array
	  Compiling (with gcc-4.9-20150603) produces an error because of an access beyond
	  the end of an array. This patch fixes the error by initializing the loop
	  control/array index variable (i) to 1 and returning i - 1 when a match is found.
	  Also, because the values stored in the array increase in value as the index
	  increases, the >= test unnecessary, so it is removed.

2015-04-30 02:52:58 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't accumulate more than 2 GOPs
	  Don't allow large amounts of data to queue up - we only need
	  the GOP we're writing, and the GOP we're accumulating.

2015-04-16 10:44:49 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: fsync after sending updates in robust mode
	  Use the new GstBuffer SYNC_AFTER flag to trigger an fsync
	  after updating the moov or mdat atom, and after updating the free
	  atom to make it visible.

2015-04-03 00:57:20 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: Only set moov header into streamheader at EOS
	  Only update the moov header into the caps if it's the finalised
	  moov at EOS time. Avoids posting a bogus moov at startup and
	  repeated updates in robust-recording mode

2015-04-03 01:44:15 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/qtmux.c:
	  tests: Update mp4 mux test for mdat placeholder change
	  The mp4 muxer now writes a place-holder mdat as a free
	  atom followed by a 0-byte mdat that covers the rest of the
	  file, making it possible to rewrite it as 64-bit, or leave
	  it as-is if nothing else is written afterward

2015-04-01 11:15:38 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  isomp4: Implement robust muxing using ping-pong strategy
	  Implement a robust recording mode, where the output
	  file is always in a playable state, seeking and rewriting
	  the moov header at a configurable interval. Rewriting
	  moov is done using reserved space at the start of
	  the file, and a ping-pong strategy where the moov
	  is replaced atomically so it's never invalid.
	  Track when tags have actually changed, and don't write them into
	  the moov unless they've changed. Clear any existing tags when
	  re-writing them, so we can do progressive moov updating in robust
	  recording mode.
	  Write placeholder mdat as a free atom plus a 32-bit mdat
	  with '0' size, which means "rest of the file" in the spec.
	  Re-write it later to a full 64-bit extended size atom if needed.

2015-04-01 00:58:52 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  isomp4: Update edit list when re-writing moov
	  Correctly update any edit lists each time the moov is recalculated,
	  updating existing table entries if they already exist instead of just
	  adding new ones.

2015-04-08 01:41:18 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: Remove an extra bracket in a comment.

2015-03-19 20:29:44 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Protect total_duration state variable with the object lock.
	  Prevent deadlocks from downstream querying duration from the streaming thread.

2015-06-07 23:06:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 21ba2e5 to d37af32

2015-06-07 19:24:20 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom/gstaudiovisualizer.c:
	  goom: clean dereferences of private structure
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-06-07 19:20:04 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom2k1/gstaudiovisualizer.c:
	  goom2k1: clean dereferences of private structure
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-06-07 17:32:01 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From c408583 to 21ba2e5

2015-06-07 17:01:37 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	  docs: remove variables that we define in the snippet from common
	  This is syncing our Makefile.am with upstream gtkdoc.

2015-06-07 17:16:19 +0200  Stefan Sauer <ensonic@users.sf.net>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From d676993 to c408583

2015-06-07 16:44:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.5.1 ===

2015-06-07 10:46:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* gst/deinterlace/tvtime-dist.c:
	* gst/videomixer/videomixerorc-dist.c:
	* win32/common/config.h:
	  Release 1.5.1

2015-06-07 10:38:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2015-06-07 10:32:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* tests/check/elements/rtpsession.c:
	  rtpsession: Override the SSRC from the packets' SSRC if none was given via caps or property

2015-06-07 09:35:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: Update translations

2015-06-05 15:32:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Don't warn when optional CID are not implement
	  gst_v4l2_get_attributre() shall only be used when the CID is expected
	  to be supported. Otherwise, we get unwanted warning posted to the bus.

2015-06-05 16:43:08 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Only suggest our internal ssrc if it's not a random one and was selected as internal ssrc
	  https://bugzilla.gnome.org/show_bug.cgi?id=749581

2015-06-04 14:18:01 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/interleave/interleave.c:
	  interleave: error when channel-positions-from-input=False
	  self->channels is being incremented only when
	  channel-positions-from-input is set as TRUE. So in case of FALSE
	  self->func is not set and hence creating assertion error.
	  Hence removing the condition to increment self->channels.
	  https://bugzilla.gnome.org/show_bug.cgi?id=744211

2015-06-05 10:33:11 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Add support for receiving reduced size RTCP
	  It worked before but gave warnings, now we just ignore RTCP
	  packets that don't start with a SR. As all we're interested
	  in here are SRs.

2015-06-03 12:22:42 +0200  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Add support for reduce size rtcp
	  According to RFC 5506, reduce size packages can be sent, this
	  packages may not be compound, so we need to add support for
	  getting ssrc from other types of packages.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750327

2015-06-03 13:14:44 +0200  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Add support for receiving reduced size rtcp
	  See RFC 5506
	  https://bugzilla.gnome.org/show_bug.cgi?id=750332

2015-06-04 16:09:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Add support for channel configurations 11, 12 and 14 and 7 actually has 8 channels
	  ISO/IEC 14496-3:2009/PDAM 4 added 11, 12 and 14.

2015-06-03 08:57:57 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/rtp/gstasteriskh263.c:
	  asteriskh263: Un-rank clashing depayloader
	  This depayloader clash with the standard one for H263p. It produces an
	  H263p stream with a modified header. It uses encoding-name that is the
	  same as H263p (H263-1998) though the resulting ES is not decodable or
	  parsable in GStreamer, making it unsuable in dynamic pipeline. This
	  patch unrank this specialized depayloader since it can only be used in
	  custom pipeline.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739935

2015-06-02 18:09:48 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	  goom2k1: remove variables not needed anymore
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-06-02 17:52:46 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom2k1/Makefile.am:
	* gst/goom2k1/gstaudiovisualizer.c:
	* gst/goom2k1/gstaudiovisualizer.h:
	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	  goom2k1: rebase to use the audiovisualizer class
	  Rebase to have goom2k1 using the common GstAudioVisualizer class
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-06-02 17:29:36 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom/Makefile.am:
	* gst/goom/gstaudiovisualizer.c:
	* gst/goom/gstaudiovisualizer.h:
	* gst/goom/gstgoom.c:
	* gst/goom/gstgoom.h:
	  goom: rebase to use the audiovisualizer class

2015-06-02 16:31:10 +0200  Edward Hervey <edward@centricular.com>

	* tests/check/pipelines/lame.c:
	  check: Use GST_CHECK_MAIN () macro everywhere
	  Makes source code smaller, and ensures we go through common initialization
	  path (like the one that sets up XML unit test output ...)

2015-06-02 16:27:24 +0200  Edward Hervey <edward@centricular.com>

	* tests/check/elements/aacparse.c:
	* tests/check/elements/ac3parse.c:
	* tests/check/elements/apev2mux.c:
	* tests/check/elements/aspectratiocrop.c:
	* tests/check/elements/audioamplify.c:
	* tests/check/elements/audiochebband.c:
	* tests/check/elements/audiocheblimit.c:
	* tests/check/elements/audiodynamic.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/audiowsincband.c:
	* tests/check/elements/audiowsinclimit.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/equalizer.c:
	* tests/check/elements/flacparse.c:
	* tests/check/elements/id3v2mux.c:
	* tests/check/elements/jpegdec.c:
	* tests/check/elements/jpegenc.c:
	* tests/check/elements/matroskamux.c:
	* tests/check/elements/mpegaudioparse.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rglimiter.c:
	* tests/check/elements/rgvolume.c:
	* tests/check/elements/rtpbin.c:
	* tests/check/elements/rtpsession.c:
	* tests/check/elements/spectrum.c:
	* tests/check/elements/videobox.c:
	* tests/check/elements/videocrop.c:
	* tests/check/elements/videofilter.c:
	* tests/check/elements/wavpackdec.c:
	* tests/check/elements/wavpackenc.c:
	* tests/check/elements/wavpackparse.c:
	* tests/check/elements/y4menc.c:
	* tests/check/pipelines/simple-launch-lines.c:
	* tests/check/pipelines/tagschecking.c:
	* tests/check/pipelines/wavpack.c:
	  check: Use GST_CHECK_MAIN () macro everywhere
	  Makes source code smaller, and ensures we go through common initialization
	  path (like the one that sets up XML unit test output ...)

2015-05-26 14:47:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Only schedule a timer when we actually have to send RTCP
	  Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create
	  RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no
	  feedback is actually pending and no regular RTCP has to be sent).
	  This improves CPU usage and battery life quite a lot.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746543

2015-05-22 13:44:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Remove useless goto
	  https://bugzilla.gnome.org/show_bug.cgi?id=746543

2015-05-21 12:54:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-H264-rtx.sh:
	* tests/examples/rtp/client-rtpaux.c:
	* tests/examples/rtp/server-VTS-H264-rtx.sh:
	* tests/examples/rtp/server-rtpaux.c:
	  examples: Set RTP profile to AVPF for rtpaux examples
	  https://bugzilla.gnome.org/show_bug.cgi?id=746543

2015-05-04 16:41:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Set RTP profile on the rtpsession objects
	  https://bugzilla.gnome.org/show_bug.cgi?id=746543

2015-05-21 14:13:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: Add rtp-profile property for setting the default profile of newly created sessions
	  https://bugzilla.gnome.org/show_bug.cgi?id=746543

2015-05-04 11:51:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Only put RRs and full SDES into regular RTCP packets
	  If we may suppress the packet due to the rules of RFC4585 (i.e. when
	  below the t-rr-int), we can send a smaller RTCP packet without RRs
	  and full SDES. In theory we could even send a minimal RTCP packet
	  according to RFC5506, but we don't support that yet.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746543

2015-05-04 13:51:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Keep track of tp/tn and t_rr_last separately
	  Otherwise we can't properly schedule RTCP in feedback profiles as we need to
	  distinguish the time when we last checked for sending RTCP (tp) but might have
	  suppressed it, and the time when we last actually sent a non-early RTCP
	  packet.
	  This together with the other changes should now properly implement RTCP
	  scheduling according to RFC4585, and especially allow us to send feedback
	  packets a lot if needed but only send regular RTCP packets every once in a
	  while.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746543

2015-05-04 11:42:08 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc)
	  And modify our RTCP scheduling algorithm accordingly. We now can send more
	  RTCP packets if needed for feedback, but will throttle full RTCP packets by
	  rtcp-min-interval (t-rr-int from RFC4585).
	  In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is
	  statically set to 1s or 0s by RFC4585. Tmin defines how often we should
	  send RTCP packets at most.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746543

2015-05-30 17:41:05 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/law/mulaw-decode.c:
	  mulawdec: Let baseclass estimate bitrate
	  This makes playback directly from a file work with the right caps.

2015-05-27 16:31:23 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstdynudpsink.h:
	  dynudpsink: keep GCancellable fd around instead of re-creating it constantly
	  And create it only when starting the element.

2015-05-27 15:55:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  udpsink, multiudpsink: keep GCancellable fd around instead of re-creating it constantly
	  Otherwise we constantly create/close event file descriptors,
	  every time we call g_socket_condition_timed_wait() or
	  g_socket_send_message(s)(), i.e. a lot. Which is not
	  particularly good for performance.
	  Can't create GCancellable in ::start() here because it's used
	  in client_new() which may be called via the add-client action
	  signal which may be called before the element is up and running.

2015-05-19 18:13:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: keep GCancellable fd around instead of re-creating it constantly
	  Otherwise we constantly create/close event file descriptors,
	  every single time we call g_socket_condition_timed_wait() or
	  g_socket_receive_message(), i.e. twice per packet received!
	  This was not particularly good for performance.
	  Also only create GCancellable on start-up.

2015-05-26 15:33:37 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/matroska/matroska-read-common.c:
	  matroska: overwritten value assignment
	  curpos is set and immediately after, set again. Remove the redundant
	  assignment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749909

2015-05-23 13:47:17 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtpvrawdepay: don't shadow existing outbuf variable
	  And fix unref of the wrong one which will contain NULL
	  in an error code path.

2015-05-23 13:23:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawdepay.h:
	  rtpvrawdepay: map/unmap output frame only once, not for every input packet
	  Map output buffer after creating it and keep it mapped
	  until we're done with it instead of mapping/unmapping
	  it for every single input buffer.

2015-05-25 08:47:47 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: remove fixme from 2006
	  It has been verified by use over time.

2015-05-23 14:36:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix reverse playback of fragmented media
	  qtdemux creates a samples array and gets the timestamps for buffers by
	  accumulating their durations. When doing reverse playback of fragments,
	  accumulating samples will lead to wrong timestamps as the timestamps
	  should go decreasing from fragment to fragment and the accumulation
	  will produce wrong results.
	  In this case, when receiving a discont for fragmented reverse playback,
	  the previous samples information should be flushed before new data
	  is processed.

2015-05-23 01:03:18 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: Implement binary search in find_part_for_offset
	  Implement binary search using gst_util_array_binary_search
	  https://bugzilla.gnome.org/show_bug.cgi?id=749690

2015-05-21 13:26:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Don't crash if we receive FIR/PLI from a source we don't know

2015-05-21 09:35:58 +0200  Santiago Carot-Nemesio <sancane@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Fix collection of statistics
	  Stats should be collected on the media rtp source not in the
	  sender one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749669

2015-04-20 10:07:30 +0200  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: Add a new max-duration file switching mode
	  This new mode ensures that files will never exceed a certain duration
	  based on incoming buffer PTS (and duration if present)
	  Note:
	  * You need timestamped buffers (duh). If some of the incoming buffers don't
	  have PTS, then it will just accept them in the current file

2015-04-17 16:18:32 +0200  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: streamline the file-switch code a bit
	  Use the same functions regardless of the mode we are using

2015-04-02 13:35:18 +0100  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: add "aggregate-gops" property to process GOPs as a whole
	  This property can be used in combination with next-file=max-size
	  (and perhaps a future next-file=max-duration) to make sure that
	  each file part starts cleanly with a key frame and the appropriate headers.
	  In order for this property to work correctly, upstream elements should make
	  sure than any headers that need to be written in a standalone file are:
	  1) in the streamheader caps field
	  2) and/or in the stream as one or more buffers marked with GST_BUFFER_FLAG_HEADER
	  that are just before the keyframe buffer
	  This is useful for MPEG-TS/MPEG-PS file segmenting in
	  combination with mpegtsmux or mpegpsmux.
	  Original patch by: Tim-Philipp Müller <tim@centricular.com>

2015-05-20 16:37:22 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Use single-include header for the RTSP library

2014-10-24 23:47:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	  udp: don't use soon-to-be-deprecated g_cancellable_reset()
	  From the API documentation: "Note that it is generally not
	  a good idea to reuse an existing cancellable for more
	  operations after it has been cancelled once, as this
	  function might tempt you to do. The recommended practice
	  is to drop the reference to a cancellable after cancelling
	  it, and let it die with the outstanding async operations.
	  You should create a fresh cancellable for further async
	  operations."
	  https://bugzilla.gnome.org/show_bug.cgi?id=739132

2015-05-18 20:13:01 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/cutter/gstcutter.c:
	* gst/equalizer/gstiirequalizernbands.c:
	* gst/multifile/gstmultifilesink.c:
	  Revert "doc: Workaround gtkdoc issue"
	  This reverts commit 1797c8f8b12d7f4c7a9444c94f34f4d08ec85945.
	  This is fixed by the gtk-doc 1.23 release.
	  <para> cannot contain <refsect2>:
	  http://www.docbook.org/tdg/en/html/para.html
	  http://www.docbook.org/tdg/en/html/refsect2.html

2015-05-18 16:40:21 +0200  Nicola Murino <nicola.murino@gmail.com>

	* gst/rtp/gstrtpg726pay.c:
	  rtpg726pay: fix caps leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=749544

2015-05-18 16:34:13 +0200  Nicola Murino <nicola.murino@gmail.com>

	* gst/rtp/gstrtpg726depay.c:
	  rtpg726depay: don't leak input buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=749543

2015-05-18 17:38:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Queue bad packets instead of dropping them
	  So we can send them out once we found the next, consecutive sequence number in
	  case one is following.

2015-05-18 17:38:14 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Use g_queue_foreach() to unref all buffers in queues

2015-05-18 17:19:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Refactor seqnum comparison code a bit

2015-05-18 17:08:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Allow sequence number wraparound during probation

2015-05-18 17:07:23 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Make sequence number comparison code more readable
	  ... by using gst_rtp_buffer_compare_seqnum() and signed integers
	  instead of implictly using effects of integer over/underflows.

2015-04-22 18:54:06 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything
	  It might just be a late retransmission or spurious packet from elsewhere, but
	  resetting everything would mean that we will cause a noticeable hickup. Let's
	  get some confidence first that the sequence numbers changed for whatever
	  reason.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747922

2015-05-16 23:37:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/cutter/gstcutter.c:
	* gst/equalizer/gstiirequalizernbands.c:
	* gst/multifile/gstmultifilesink.c:
	  doc: Workaround gtkdoc issue
	  With gtkdoc 1.22, the XML generator fails when a itemizedlist is
	  followed by a refsect2. Workaround the issue by wrapping the
	  refsect2 into para.

2015-01-23 13:57:40 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/isomp4/qtdemux_types.c:
	  qtdemux: avoid wrong warnings on unknown node types
	  Add 'name' and 'mean' fourccs, as we handle them. Right now each use would
	  trigger a warning.

2015-05-08 19:13:00 +0200  Nicola Murino <nicola.murino@gmail.com>

	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726depay.h:
	  rtpg726depay: add block_align to output caps
	  It is needed to correctly negotiate caps with matroskamux
	  and most other muxers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749129

2015-05-12 13:41:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Fix time-domain convolution with >1 channels
	  input_samples is the number of frames, but we used it as the number of
	  samples.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747204

2015-05-12 12:13:16 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vp[89]enc: Properly convert between GStreamer and encoder timebase
	  ... by switching numerator and denominator when scaling.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749122

2015-05-11 13:33:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vp[89]enc: Don't set timebase from the framerate
	  The framerate very often is just an indication of the ideal framerate, not the
	  actual framerate of the stream. By just using the framerate, we confuse the
	  rate control algorithm algorithm as multiple frames will map to the same PTS
	  or have durations of 0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749122

2015-05-10 14:21:04 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* tests/check/elements/wavpackparse.c:
	  tests: wavpackparse: fix unit test
	  See also https://bugzilla.gnome.org/show_bug.cgi?id=738237

2015-05-10 11:34:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/twolame/gsttwolamemp2enc.c:
	  docs: update example pipelines in element docs
	  Mostly gst-launch -> gst-launch-1.0, but also
	  use autoaudiosink/autovideosink in more places
	  and update pipelines a little or flesh out
	  descriptions.

2015-05-10 11:34:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/lame/gstlamemp3enc.c:
	  docs: update example pipelines in element docs
	  Mostly gst-launch -> gst-launch-1.0, but also
	  use autoaudiosink/autovideosink in more places
	  and update pipelines a little or flesh out
	  descriptions.

2015-05-10 11:05:00 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/shout2/gstshout2.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpL24pay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtpmanager/gstrtpmux.c:
	* tests/check/pipelines/wavenc.c:
	* tests/examples/rtp/client-PCMA.c:
	* tests/examples/rtp/server-alsasrc-PCMA.c:
	  docs: update example pipelines in element docs
	  Mostly gst-launch -> gst-launch-1.0
	  Use autovideosink/autoaudiosink more often.
	  Sprinkle some converters here and there.

2015-05-09 19:48:55 +0200  Piotr Drąg <piotrdrag@gmail.com>

	* po/POTFILES.in:
	  po: update POTFILES.in
	  https://bugzilla.gnome.org/show_bug.cgi?id=749163

2015-05-10 10:52:18 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: minor error message clean-up
	  Don't put filename in error message shown to user.

2015-05-07 16:25:36 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix buffer leak when stored to seektable
	  Fix a leak with the
	  validate.file.playback.change_state_intensive.samples_multimedia_cx_flac_Yesterday_flac
	  scenario.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749072

2015-05-07 17:10:37 +0900  Paul Hyunil <paul.hyunil@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix example pipeline in docs
	  The gst-launch script for example launch line to test qtdemux is
	  missing a queue before the decodebins, otherwise the gst-launch-1.0
	  command won't work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749054

2015-05-07 14:51:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  Revert "rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active"
	  This reverts commit d22ec496328e6ba8edbf2d071d5608b2af2831e8.
	  Application code might expect that it only gets external sources on those
	  signals, and get confused by this. If anything we would need to add new
	  signals.

2015-03-25 15:27:34 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active
	  Without this it seems impossible for an application to easily get notified
	  about the internal ssrcs that are created, e.g. sender sources, and also
	  to know when they are active and produce RTCP packets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746747

2015-05-04 19:26:14 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix frame leaks in handle_frame() implementation
	  handle_frame() is supposed to consume @frame, so if we don't call
	  gst_video_decoder_drop_frame() or gst_video_decoder_finish_frame() we have to
	  release it manually.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748909

2015-05-04 16:50:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix up last commit

2015-05-04 16:46:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Only do RTX when using a feedback profile

2015-05-04 13:50:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: The stats min_interval is in seconds, not nanoseconds
	  We have to scale it to compare it against our clock times.

2015-05-04 11:38:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Only return TRUE if early feedback was requested already and it's early enough

2015-04-30 15:42:34 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/matroska/matroska-parse.c:
	  matroska: remove unused property enum items

2015-04-30 12:13:59 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix buffer leak on eos in push mode
	  Based on patch by Guillaume Desmottes.
	  scenario: validate.http.playback.seek_with_stop.raw_h264_1_mp4
	  https://bugzilla.gnome.org/show_bug.cgi?id=748617

2015-04-29 19:41:29 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check for sizes of the rdrf (redirect) atom before accessing the data and use g_strndup() instead of g_strdup()
	  Thanks to Ralph Giles for reporting this.

2015-04-29 15:52:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Only enable retransmissions if there is retransmission info in the SDP
	  Otherwise we're going to send early RTCP and NACKs in non-feedback sessions
	  too, which will confuse servers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748627

2015-02-11 18:09:24 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: extract recording time
	  Extracts the recorded time of the dv file from
	  the metadata and puts it into the global tags.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743657

2015-04-28 15:59:25 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix seek event leak
	  gst_matroska_demux_handle_seek_event() doesn't consume the
	  event so we have to unref it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748584

2015-04-28 15:42:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Send pending tags when adding a new pad
	  We might've parsed those tags before already and tried to push them to
	  non-existing pads before. Now let's do it for real.

2015-04-23 18:57:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpstats.c:
	  rtpstats: Average RTCP packet size is in bytes, bandwidths in bits
	  We need to convert the size to bits for our calculations.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747863

2015-04-23 18:53:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpstats.c:
	  rtpstats: Use the same lower limit for RTCP bandwidth to stop sending RTCP everywhere
	  https://bugzilla.gnome.org/show_bug.cgi?id=747863

2015-04-14 18:41:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value
	  https://bugzilla.gnome.org/show_bug.cgi?id=747863

2015-04-23 18:49:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Bandwidth is supposed to be in bits/s, not bytes/s
	  https://bugzilla.gnome.org/show_bug.cgi?id=747863

2015-04-27 16:36:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix RTX unit test
	  The calculations were a bit off everywhere, even before the changes done
	  recently to the delay for RTX of expected future packets. It only worked by
	  accident, but now the calculations are all correct again. Hopefully.

2015-04-27 11:22:11 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/avi/gstavimux.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/flv/gstindex.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/id3demux/gstid3demux.c:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multipart/multipartmux.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/y4m/gsty4mencode.c:
	  Rename property enums from ARG_ to PROP_
	  Property enum items should be named PROP_ for consistency and readability.

2015-04-27 10:55:13 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/audiofx/gststereo.c:
	  Rename property enums from ARG_ to PROP_
	  Property enum items should be named PROP_ for consistency and readability.

2015-04-25 02:49:58 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix "stats" property docs
	  https://bugzilla.gnome.org/show_bug.cgi?id=748436

2015-04-26 17:54:52 +0100  Tim-Philipp Müller <tim@centricular.com>

	* Android.mk:
	* gst/alpha/Makefile.am:
	* gst/apetag/Makefile.am:
	* gst/audiofx/Makefile.am:
	* gst/auparse/Makefile.am:
	* gst/autodetect/Makefile.am:
	* gst/avi/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debugutils/Makefile.am:
	* gst/deinterlace/Makefile.am:
	* gst/dtmf/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/equalizer/Makefile.am:
	* gst/flv/Makefile.am:
	* gst/flx/Makefile.am:
	* gst/goom/Makefile.am:
	* gst/goom2k1/Makefile.am:
	* gst/icydemux/Makefile.am:
	* gst/id3demux/Makefile.am:
	* gst/imagefreeze/Makefile.am:
	* gst/interleave/Makefile.am:
	* gst/isomp4/Makefile.am:
	* gst/law/Makefile.am:
	* gst/level/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/monoscope/Makefile.am:
	* gst/multifile/Makefile.am:
	* gst/multipart/Makefile.am:
	* gst/replaygain/Makefile.am:
	* gst/rtp/Makefile.am:
	* gst/rtpmanager/Makefile.am:
	* gst/rtsp/Makefile.am:
	* gst/shapewipe/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/spectrum/Makefile.am:
	* gst/udp/Makefile.am:
	* gst/videobox/Makefile.am:
	* gst/videocrop/Makefile.am:
	* gst/videofilter/Makefile.am:
	* gst/videomixer/Makefile.am:
	* gst/wavenc/Makefile.am:
	* gst/wavparse/Makefile.am:
	* gst/y4m/Makefile.am:
	  Remove obsolete Android build cruft
	  This is not needed any longer.

2015-04-24 13:55:08 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: print the property values when set
	  Instead of printing the currently used values. The log is meant
	  to show what the properties changed to, not what is being currently
	  used.

2015-04-24 17:01:10 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/alpha/gstalpha.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	  remove unused enum items PROP_LAST
	  This were probably added to the enums due to cargo cult programming and are
	  unused. Removing them.

2015-04-24 00:30:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/level/gstlevel.c:
	  level: fix infinite loop for very low interval values
	  https://bugzilla.gnome.org/show_bug.cgi?id=745515

2015-04-23 16:08:54 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	  tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
	  Make sure the test environment is set up.
	  https://bugzilla.gnome.org//show_bug.cgi?id=747624

2015-04-23 16:08:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	  configure: bump automake requirement to 1.14 and autoconf to 2.69
	  This is only required for builds from git, people can still
	  build tarballs if they only have older autotools.
	  https://bugzilla.gnome.org//show_bug.cgi?id=747624

2015-04-23 16:06:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* .gitignore:
	  Update .gitignore

2015-04-23 09:55:59 +0200  Jesper Larsen <knorr.jesper@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix RTCP caps leak
	  https://bugzilla.gnome.org//show_bug.cgi?id=748353

2015-04-22 20:24:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: When request retransmissions for future packets, consider the packet spacing in the extra delay
	  We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra
	  delay. If jitter is very low, this should prevent unnecessary retransmission
	  requests to some degree.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748041

2015-04-22 19:41:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Take a running average of the packet spacings instead of just the latest
	  https://bugzilla.gnome.org/show_bug.cgi?id=748041

2015-04-13 11:20:40 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Add "rtx-next-seqnum" property
	  If this is set to FALSE, rtpjitterbuffer will not request retransmissions for
	  future packets based on when they are estimated to arrive.
	  See also https://bugzilla.gnome.org/show_bug.cgi?id=748041
	  https://bugzilla.gnome.org/show_bug.cgi?id=739868

2015-04-22 19:29:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtxreceive: Put debug output for retransmission requests at the right place
	  Before it was only ever printed once for every time a ssrc was associated with
	  a specific stream.

2015-04-22 18:05:24 +0200  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: don't add the same interlace mode twice
	  Some drivers modify the interlace mode to progressive, no matter what
	  input you give them, make sure that we don't add the same interlace mode
	  twice.

2015-04-21 16:34:21 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: fix dynamic changes on bands
	  When we are in passthrough, the transform function doesn't run and if the
	  passthrough check is in this function it will never be deactivated. Fix this by
	  checking directly whenever a gain is changed.
	  Also set the passthrough to TRUE at init because the gains default to 0, so we
	  can passthrough until any gain property is changed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748068

2015-04-22 10:30:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* INSTALL:
	  Remove INSTALL file
	  autotools automatically generate this, and when using different versions
	  for autogen.sh there will always be changes to a file tracked by git.

2015-04-22 10:30:14 +0200  Sebastian Dröge <sebastian@centricular.com>

	* LICENSE_readme:
	  Remove LICENSE_readme
	  It's completely outdated and just confusing, better if people are
	  forced to look at the actual code in question than trusting this file.

2015-04-21 15:21:33 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: cast unused return to void
	  Quell unchecked return value defect by casting the return value to void and
	  making it explicit it is going to be ignored.
	  CID #206031

2015-04-17 13:08:02 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: optimize vpx image to gstbuffer copy when strides match
	  Solving this FIXME. Copy the full plane when strides are the same

2015-04-16 15:11:05 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/vpx/gstvp9dec.c:
	  vp9dec: optimize vpx image to gstbuffer copy when strides match
	  Solving this FIXME. Copy the full plane when strides are the same

2015-04-17 13:32:54 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: fix memory leak

2015-04-17 06:51:46 +0000  Alex O'Konski <alexanderokonski@gmail.com>

	* gst/icydemux/gsticydemux.c:
	  icydemux: Fix segfault if metadata-interval is 0
	  Prevents an extra unref of GstBuffer when passing a non-icy stream through
	  icydemux with metadata-interval set to 0.
	  Reproducible with:
	  gst-launch-1.0 filesrc location=~/testsong.mp3 ! \
	  'application/x-icy,metadata-interval=(int)0' ! icydemux ! decodebin ! wavenc ! \
	  filesink location=~/testsong.wav
	  https://bugzilla.gnome.org/show_bug.cgi?id=748024

2015-04-17 11:54:23 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiodynamic.c:
	  audiofx: fix typo in example pipelines
	  Fix typo in example pipelines
	  https://bugzilla.gnome.org/show_bug.cgi?id=748022

2015-04-15 18:22:37 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: fix spelling in debug message
	  https://bugzilla.gnome.org//show_bug.cgi?id=747936

2015-04-16 16:33:44 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* tests/examples/equalizer/demo.c:
	  tests: selectable amount of bands in equalizer demo
	  Adding an option in the equalizer demo to make the number of bands selectable.

2015-04-16 15:31:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource
	  https://bugzilla.gnome.org/show_bug.cgi?id=747394

2015-04-06 12:56:50 +0530  Arun Raghavan <arun@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Track RTX ssrc caps
	  This is needed so that we can generate SR for RTX stream correctly (the
	  clock rate is required).
	  https://bugzilla.gnome.org/show_bug.cgi?id=747394

2015-04-14 13:56:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: Copy over timestamps from the orignal buffers to the RTX buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=747394

2015-04-16 16:01:50 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* tests/examples/equalizer/demo.c:
	  tests: switch equalizer demo to play from uri
	  Switch the equalizer-nbands demo to use uridecodebin, so users can listen to
	  something more pleasant than white noise. If anybody misses the white noise
	  a uri handler to audiotestsrc can be used.

2015-04-16 11:17:38 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* tests/examples/equalizer/demo.c:
	  tests: improve readability of equalizer demo
	  Rename variable name to make it more readable, add comments for the three
	  scales created per block, and set the window title.

2015-04-15 17:32:37 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* tests/examples/equalizer/demo.c:
	  tests: add missing license header for equalizer demo

2015-04-16 13:09:19 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix tag list leaks on error paths

2015-04-16 12:23:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix tag list leak on unknown stream type

2015-04-09 13:19:49 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/gst-plugins-good.supp:
	  suppressions: ignore an apparent bug in strtod
	  A buffer overread.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747554

2015-04-15 11:07:27 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: do not access property variable without the object lock, use the local stack copy instead

2015-04-14 18:45:44 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: add probe on the multiqueue's sink pad instead of the ghost pad
	  because _release_pad tries to release it from ctx->sinkpad, which is
	  multiqueue's sink pad, and currently fails because the probe is not
	  installed there

2015-04-14 19:08:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtx*: Fix typos

2015-04-14 17:24:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Not sending early RTCP now because of dithering means we send it with the next compound packet

2015-04-14 16:27:18 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Improve debug output a bit if we can't allow early feedback

2015-04-07 18:00:53 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpvp8depay: When dropping intra packet, request keyframe
	  https://bugzilla.gnome.org/show_bug.cgi?id=747208

2015-04-13 20:25:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Change resyncing GST_WARNING to GST_INFO
	  This also happens in the very beginning when we receive the first packet, a
	  warning would be very confusing here. In all places where we should warn about
	  this, we would've printed a warning already before.

2015-04-02 13:26:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: minor docs improvement

2014-11-06 12:08:03 +0100  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Add "rtx-max-retries" property
	  This property allows to limit the maximum number of retransmission
	  for a specific packet.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739868

2014-11-04 15:00:52 +0100  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix expected_dts calc in calculate_expected
	  Right above we consider lost_packet packets, each of them having duration,
	  as lost and triggered their timers immediately. Below we use expected_dts
	  to schedule retransmission or schedule lost timers for the packets that
	  come after expected_dts.
	  As we just triggered lost_packets packets as lost, there's no point in
	  scheduling new timers for them and we can just skip over all lost packets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739868

2015-03-20 18:21:57 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Make the next output buffer discont after resetting the jitterbuffer
	  Resetting the jitterbuffer drops all packets and other things, and will cause
	  a discontinuity in the packets received by the depayloaders. They should now
	  also flush anything they had pending as the new data will start at a different
	  position.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739868

2015-04-10 09:17:26 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Update segment.start after key-unit seek
	  When doing key uint seek, qtdemux calls gst_qtdemux_adjust_seek
	  to get proper offset. And then this offset is set to
	  segment.position and segment.time in gst_qtdemux_perform_seek but
	  segment.start is not updated.
	  After that, application sends segment query,
	  qtdemux sets start and stop to query using gst_segment_to_stream_time. Due
	  to the wrong value in segment.start, the stop position is smaller than
	  it should.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746822

2015-04-07 16:12:40 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: remove useless variable do_pts
	  We always write the CTTS in qtmux. Ideally we only want to do that
	  for streams that need DTS, it should be present on the track information
	  rather than be decided based on each buffer

2015-04-07 00:53:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: remove subtraction that makes PTS/DTS start from 0
	  As qt uses durations, it doesn't matter, only the difference
	  between consecutive buffers is important. Also, collectpads
	  already replaces PTS/DTS with the running times for them.

2015-04-06 22:36:43 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: add tests to verify it handles non-0 segments
	  Both input streams in this test have a segment.start = 10s, so
	  output should start from 0 anyway.
	  Another test has both starting at non-0 segments, but the running
	  time of both streams should still start from 0

2015-04-06 20:03:19 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: simple muxing test
	  Adds a new simple test that verifies that data is properly muxed
	  and preserved.  PTS, DTS, duration and caps are verified.

2015-04-10 10:59:26 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/smpte/gstsmpte.h:
	  smpte: remove unused fields
	  Remove the fields - format and fps from smpte
	  as they are unused.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747597

2015-04-10 10:29:47 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/alpha.c:
	  tests: add test suite for alpha
	  Added test suite for alpha element with test cases
	  1. alpha
	  2. chroma keying
	  https://bugzilla.gnome.org/show_bug.cgi?id=747595

2015-04-09 12:58:46 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/gst-plugins-good.supp:
	  suppressions: add a well known zlib inflate bug

2015-04-09 12:58:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: fix mutex leak

2015-04-09 12:58:04 +1000  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/rtprtx.c:
	  tests: Fix rtprtx test by handling buffer lists
	  Commit #1018aa made rtprtxsend handle buffer lists, breaking
	  the test which probes for buffers, but not buffer lists.
	  Use a utility function to run the probe callback on each buffer
	  in the list in turn and remove any buffers that are dropped.

2015-04-01 11:15:38 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  isomp4: Refactor various state variables into a mux_mode var
	  Instead of checking various state variables around the muxer,
	  track the current muxing mode in a single 'mux_mode' enum.
	  Add some implementation notes about the different mux modes

2015-04-08 16:40:02 +0200  Edward Hervey <edward@centricular.com>

	* common:
	* tests/check/Makefile.am:
	  tests: Use AM_TESTS_ENVIRONMENT
	  Needed by the new automake test runner

2015-04-08 11:17:31 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: Fix framesize parsing
	  The string passed to the parsing function only contains a framesize, and
	  not <pt> + <framesize>
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416

2015-03-20 12:18:37 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: clip chunk size above the valid maximum (0x7fffffff)
	  https://bugzilla.gnome.org/show_bug.cgi?id=722567

2015-03-20 09:07:35 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: clip chunk length to available data (when known)
	  This prevents silly chunk lengths from possibly overflowing
	  (at least when we know the actual data length).
	  https://bugzilla.gnome.org/show_bug.cgi?id=722567

2015-04-06 20:17:52 -0700  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't accumulate segment bases manually
	  gst_segment_do_seek() does that for us already, and doing it twice
	  will break non-flushing seeks in interesting ways. Leftover from 1.0
	  porting.
	  Also copy over segment offset and applied_rate, just in case.

2015-04-06 19:08:10 -0700  Sebastian Dröge <sebastian@centricular.com>

	* tests/icles/test-segment-seeks.c:
	  icles: Fix waiting for segment-done if it happens too fast
	  Sometimes we can get segment-done before we got async-done. If we waited
	  for async-done only, the segment-done would be dropped and we would wait
	  forever for it a few lines below.

2015-04-06 18:55:08 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: stbl_index is valid from 0 onwards
	  It indicates the last sample parsed, not the next one to parse.
	  As it starts in -1, any value from 0 onwards means that it has
	  some valid data.

2015-04-05 20:06:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  docs: make GstRTCPSync enum show up in rtpbin docs
	  https://bugzilla.gnome.org/show_bug.cgi?id=747358

2015-04-05 11:45:45 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: add RTPJitterBufferMode enum to rtpbin docs
	  https://bugzilla.gnome.org/show_bug.cgi?id=747358

2015-04-04 11:55:00 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: close files before posting message
	  Makes sure the files were properly flushed and closed before
	  the message reaches the application

2015-03-30 13:54:23 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/multifile.c:
	  tests: multifile: increment tests to check for multifile messages
	  Also verify that the multifilesink file messages are being correctly
	  posted to the bus

2015-03-30 12:51:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/multifile.c:
	  tests: multifile: handle FIXME for proper checking when test finished
	  Use a GstBus and wait for EOS to finish the tests instead of
	  relying on sleeping

2015-03-30 11:14:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: post file message on EOS
	  When multifilesink is operating in any mode other than one file
	  per buffer, the last file created won't have a file message posted
	  as multifilesink doesn't handle the EOS event.
	  This patch fixes it by using the last position to post a file
	  message when EOS is received. This should ensure at least the
	  time related data and the filename are posted to the application
	  or other elements
	  https://bugzilla.gnome.org/show_bug.cgi?id=747000

2015-04-03 18:57:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From bc76a8b to c8fb372

2015-04-03 02:08:50 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Guard against 64-bit overflow
	  For large-file atoms, guard against overflow in the size field,
	  which could make us jump backward in the file and cause
	  infinite loops.

2015-04-01 23:46:13 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* tests/check/elements/qtmux.c:
	  isomp4: Make non-seekable downstream an error in normal mode
	  When not in fast-start or fragmented mode, we need to be able
	  to rewrite the size of the mdat atom, or else the output just
	  won't be playable - the mdat placeholder with size == 0 will
	  cover the rest of the file, including any moov atom we write out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=708808

2014-03-15 15:23:01 +0100  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtph263pay/-depay: add framesize SDP attribute
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416

2014-03-15 13:33:56 +0100  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay/-depay: Remove incorrectly introduced framesize SDP attribute
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726415

2015-03-27 21:09:44 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2src: device sequence/offset correction in case of renegotiation
	  The v4l2 device restarts the sequence counter in case of streamoff/streamon,
	  the GST offset values are supposed to increment strictly monotonic, so
	  adjust the sequence counter/offset values in case of caps
	  renegotiation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745441

2014-11-14 14:18:51 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: add frame loss detection
	  In case of v4l2 driver filled offset/sequence values add frame
	  loss detection (and write a warning message).
	  Move offset meta data setting and frame loss checking after the
	  timestamp adjustment code to get proper timestamps for the
	  warning message.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745441

2014-11-14 13:48:51 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: use v4l2 capture device sequence counter
	  Use the v4l2 capture device sequence counter for
	  setting the GstBuffer offset/offset_end values.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745441

2015-03-30 13:12:35 +0200  Tobias Modschiedler <tobias.modschiedler@cetitec.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: Ask the driver about its requirements for min_buffers before initiating buffer pool.
	  If propose_allocation() had not been called yet, it was possible that the driver was not asked at all.
	  In buffer pool: Consider minimum number of buffers requested by driver when setting config.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746834

2015-04-01 19:30:27 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	  rtpvp8depay: Parse width/height/profile from keyframes
	  This makes it possible to mux the result into a container
	  such as matroska.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747208

2015-04-01 19:01:49 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Expose VP8 width/height limitations in the caps template
	  The VP8 format specification (RFC 6386 section 18.1) specifies
	  that the maximum size is 16383x16383.

2015-03-31 00:20:13 +1100  Jan Schmidt <jan@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flv: When passing seek event upstream, hold a ref.
	  In case upstream can't handle the seek, make sure we
	  keep a ref on the event to attempt to handle it ourselves.

2015-03-26 13:34:53 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  matroska: fix GValue leaks when parsing tags
	  gst_tag_list_add_value() doesn't consume the GValue we pass to it so there is
	  no point copying it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746810

2015-03-23 20:58:25 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: resurrect some flow return handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=744572

2015-03-23 20:57:56 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: resurrect some flow return handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=744572

2015-03-23 20:56:41 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: resurrect some flow return handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=744572

2015-03-27 18:58:31 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-read-common.c:
	  matroska: store stream tags and push as updated
	  New tags can be found on different parts of the file, so this patch
	  keeps the stream taglists around for the life cycle of the pad
	  and adds those new tags as found. Then a new tag is found, the
	  pad's is marked with a tags changed flag, making the element push
	  a new tag event on the next check. Before this, we were sending
	  only the newly found tags, as the element was losing its taglist
	  when pushing the event.

2015-03-15 14:40:36 +0100  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: send global tags incrementally
	  Instead of sending only new tags once they are found, merge the taglist
	  and send them incrementally.

2015-03-14 17:07:05 +0100  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroskaparse: send global tags
	  Global tags are already being read in matroskaparse, but they are not
	  currently being sent.
	  This patch makes global tags get sent incrementally whenever new ones
	  are found.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746242

2015-02-03 10:18:58 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* gst/effectv/gstquark.c:
	  quarktv: fix "planes" property range, a value of 0 is not allowed
	  When planes property is set to 0, the pipeline executes in
	  an infinite loop and never exits. Since planes must never
	  be 0, set the minimum value in the property description
	  to 1.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743906

2015-03-26 13:42:02 -0700  David Schleef <ds@schleef.org>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix up comments regarding DTS

2015-03-25 15:11:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Fix segment in TCP mode
	  It is expected that buffers are time-stamped with running time. Set
	  a segment accordingly. In this case we pick 0,-1 as this is what udpsrc
	  would do. Depayloaders will update the segment to reflect the playback
	  position.
	  https://bugzilla.gnome.org/show_bug.cgi?id=635701

2015-03-26 12:21:25 -0700  David Schleef <ds@schleef.org>

	* gst/wavparse/gstwavparse.c:
	  wavparse: be more strict about typefinding DTS
	  Code now matches comments.

2015-03-25 15:10:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Remove useless function
	  This function didn't do anything special, let's not use a function for
	  that.

2015-03-20 13:03:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitter: Account for rtx_retry in overflow check
	  As rtx_retry is part of the substraction, we need to take it into
	  account, otherwise we may endup with a big value.

2015-03-24 23:15:15 +0000  Julien Isorce <j.isorce@samsung.com>

	* sys/osxvideo/cocoawindow.m:
	  osxvideosink: check for deprecated constants prior to OSX 10.10
	  cocoawindow.m:339:5: error: 'NSOpenGLPFAWindow'
	  is deprecated: first deprecated in OS X 10.9
	  cocoawindow.m:576:7: error: 'NSOpenGLPFAFullScreen'
	  is deprecated: first deprecated in OS X 10.6
	  cocoawindow.m:605:24: error: 'setFullScreen'
	  is deprecated: first deprecated in OS X 10.7

2015-03-24 16:51:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix seeking query
	  The segment start/stop in the query is meant to represent the seekable
	  portion of the stream. It does not match the segment start/stop. Instead
	  export 0 to duration.

2015-03-24 16:18:53 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Only set caps once if they don't change
	  Previously we were setting new caps with the same content for every H264 or
	  AAC codec_data we found in the stream, spamming everything and causing
	  renegotiations.

2015-03-24 12:46:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't create AAC/H264 caps without codec_data
	  Instead delay creating the caps until we read the codec_data from the stream,
	  or fail if we get normal data before the codec_data.
	  AAC raw caps and H264 avc caps always need codec_data, setting caps on the pad
	  without them is going to make negotiation fail most of the time. Even if we
	  later set new caps with the codec_data, that's usually going to be too late.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746682

2015-03-24 15:39:22 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Fix indention

2015-03-22 13:23:44 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudio: Fix string format warning on 32-bit
	  UInt32 (Darwin, not C99's uint32_t) is 'unsigned long' on 32-bit
	  platforms.

2015-03-21 17:50:40 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Fix another instance of sticky event misordering warnings
	  Make sure that the sync_src pad has caps before the segment event.
	  Otherwise we might get a segment event before caps from the receive
	  RTCP pad, and then later when receiving RTCP packets will set caps.
	  This will results in a sticky event misordering warning
	  This fixes warnings in the rtpaux unit test but also in the
	  rtpaux and rtx examples in tests/examples/rtp
	  https://bugzilla.gnome.org/show_bug.cgi?id=746445

2015-03-21 17:18:47 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Also start the RTCP send thread when receiving RTP or RTCP
	  Before we only started it when either:
	  - there is no send RTP stream
	  or
	  - we received an RTP packet for sending
	  This could mean that if the send RTP pads are connected but never receive any
	  RTP data, and the same session is also used for receiving RTP/RTCP, we would
	  never start the RTCP thread and would never send RTCP for the receiving part
	  of the session.
	  This can be reproduced with a pipeline like:
	  gst-launch-1.0 rtpbin name=rtpbin \
	  udpsrc port=5000 ! "application/x-rtp, media=video, clock-rate=90000, encoding-name=H264" ! rtpbin.recv_rtp_sink_0 \
	  udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
	  rtpbin.send_rtcp_src_0 ! fakesink name=rtcp_fakesink silent=false async=false sync=false \
	  rtpbin.recv_rtp_src_0_2553225531_96 ! decodebin ! xvimagesink \
	  fakesrc ! valve drop=true ! rtpbin.send_rtp_sink_0 \
	  rtpbin.send_rtp_src_0 ! fakesink name=rtp_fakesink silent=false async=false sync=false -v
	  Before this change the rtcp_fakesink would never send RTCP for the receiving
	  part of the session (i.e. no receiver reports!), after the change it does.
	  And before and after this change it would send RTCP for the receiving part of
	  the session if the sender part was omitted (the last two lines).

2015-03-19 11:54:12 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: Add support for buffer lists

2015-03-19 11:39:38 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtprtxqueue.c:
	  rtprtxqueue: Implement support for buffer lists

2015-03-18 17:32:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Improve trace readability
	  Change the command number into strings.

2015-01-20 10:18:56 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Don't repeatedly warn after no_more_pads (v2)
	  This can get rather spammy for such a high log level.
	  Only warn once per stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746274

2015-03-16 11:23:52 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Introduce constant for no-more-pads threshold
	  https://bugzilla.gnome.org/show_bug.cgi?id=746274

2015-01-20 10:18:29 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Fix warning to contain 'video'
	  https://bugzilla.gnome.org/show_bug.cgi?id=746274

2015-03-11 21:25:40 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: for dts only stream set pts=dts for intra only formats
	  https://bugzilla.gnome.org/show_bug.cgi?id=745192

2015-03-14 16:39:09 +0100  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-read-common.c:
	  matroskademux: fix sending of tags
	  * Fix critical when new tags are found after segment event has already
	  been sent.
	  * Send global tags before stream tags.
	  * Split sending of tags out of gst_matroska_demux_send_event() into its
	  own function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745973

2015-03-13 18:26:06 +0000  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: properly escape percent sign in documentation

2015-03-13 18:26:44 +0000  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: properly escape percent sign in documentation

2015-03-13 18:48:03 +0000  Thiago Santos <thiagoss@osg.samsung.com>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2src: delay renegotiation until it is likely buffers were reclaimed
	  Allow renegotiation to happen when buffers have returned after an allocation
	  query. As the allocation query is serialized, all buffers from the pool
	  should have returned and we can stop it to create a new one for the
	  new format
	  https://bugzilla.gnome.org/show_bug.cgi?id=682770

2015-03-13 18:47:55 +0000  Thiago Santos <thiagoss@osg.samsung.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: add gst_v4l2_object_try_format
	  Similar to set_format but it uses TRY_FMT instead of S_FMT
	  https://bugzilla.gnome.org/show_bug.cgi?id=682770

2015-03-13 18:38:42 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: fix crash with GST_DEBUG enabled
	  g_inet_socket_address_get_address() does not give
	  us a ref to the address, so don't unref it.

2015-03-12 13:49:56 +0000  Sebastian Dröge <sebastian@centricular.com>

	* gst/level/gstlevel.c:
	  level: Don't read over the end of the input memory
	  Previously we advanced the in_data pointer by bps for every channel, and then
	  later again for block_size*bps. This caused us to be one sample further than
	  expected if an input buffer covered two analysis frames. And in the end lead
	  to completely bogus values reported by level.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746065

2015-03-12 01:37:08 +1100  Jan Schmidt <jan@centricular.com>

	* sys/oss/gstossdmabuffer.c:
	  Remove a couple of superfluous trailing semi-colons

2015-03-10 09:31:20 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/alpha/gstalpha.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/isomp4/gstisoff.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/udp/gstmultiudpsink.c:
	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	  Fix double semicolons

2015-03-10 15:46:40 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmux: Shut down element before downward state change
	  Make sure the state change won't hang trying to shut down pads
	  by making sure the streaming has stopped before chaining up.

2015-03-09 22:58:05 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudio: stream format is an SPDIF-only field

2015-03-09 22:53:41 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudiosrc.h:
	  osxaudio: fix spaces

2015-03-09 22:52:46 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudiosrc.h:
	  osxaudio: add type check macro

2015-03-09 22:51:51 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiocommon.h:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: rename gst_core_audio_set_channels_layout()
	  to gst_core_audio_get_channel_layout().

2015-03-09 22:30:28 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	  osxaudio: remove unused finalize

2015-03-09 16:25:43 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* ext/vpx/gstvp9enc.c:
	  vp9enc: remove duplicate declaration of function

2015-03-09 16:22:29 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: remove unused value
	  CID #1226474

2015-03-09 16:14:34 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: fix leak
	  CID 1212156

2015-03-09 15:58:33 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: remove uneeded variable
	  We just need to save the ebit information in case there is an error decoding.

2015-03-09 16:46:02 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vp[89]enc: Reset the encoder when flushing
	  https://bugzilla.gnome.org/show_bug.cgi?id=745704

2015-03-09 12:51:17 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/matroska/matroska-parse.c:
	  matroska: error mode if can't push buffer
	  If gst_pad_push() fails, inform and return flow error.

2015-03-09 12:13:34 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/matroska/matroska-parse.c:
	  matroska: unused value
	  Value set in ret will be overwritten just before exiting the function.
	  CID #1226469

2015-03-09 11:10:35 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Drop packets with sequence numbers before the seqnum-base
	  These are outside the expected range of sequence numbers and should be
	  clipped, especially for RTSP they might belong to packets from before a seek
	  or a previous stream in general.

2014-02-27 10:52:16 +0100  Linus Svensson <linussn@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't include payload type in the caps for framesize
	  When the sdp media attribute framesize are converted to caps
	  the <payload> should not be included.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335

2015-03-09 10:05:14 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Don't forget to unlock the mutex when receiving GAPs in TCP streams

2015-03-09 11:24:58 +0530  Arun Raghavan <arun@centricular.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Make sure to filter caps in all cases during CAPS query
	  We were skipping the filter step while returning template caps, for
	  example.

2015-03-08 21:15:53 +0000  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Don't update buffer for OUTPUT
	  For output device, we should not update the buffer with flags and
	  timestamp when we dequeue. The information in the v4l2_buffer is not
	  meaningful and it breaks the case where the buffer is rendered at
	  multiple places.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745438

2015-03-08 18:04:34 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Implement cookies property

2015-03-08 18:02:51 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Implement automatic-redirect property

2015-03-08 17:54:07 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Implement proxy support
	  The properties were there before, but not used anywhere.

2015-02-21 20:05:24 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavidemux.c:
	  avidemux: resurrect some flow return handling

2015-03-04 10:27:17 +0100  Nicolas Huet <nicolas.huet@parrot.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix LOAS parsing issue
	  Fix missing index in syncword searching
	  https://bugzilla.gnome.org/show_bug.cgi?id=745585

2015-03-05 17:54:43 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: fix modulo math with ringbuffer parameters
	  To get a multiple of bpf use a subtraction and not an addition
	  https://bugzilla.gnome.org/show_bug.cgi?id=745684

2015-03-07 00:55:47 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Protect property variables with the object lock.
	  Use the object lock instead of the splitmux lock to protect
	  internal property variables, so they're not locked when
	  switching to a new file.
	  https://bugzilla.gnome.org/show_bug.cgi?id=744420

2015-03-06 11:39:39 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  check: add jitterbuffer unit test
	  See https://bugzilla.gnome.org/show_bug.cgi?id=745539

2015-03-05 09:18:52 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix handling of interleaved (TCP) streams
	  We need to set up the transport in any case, not just if we have a container
	  stream or a non-interleaved stream. Only if we have an interleaved stream and
	  are retrying, we should not set up the stream again.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745599

2015-03-05 10:00:33 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  vp[89]dec: Drop frames that have no output buffer because of errors
	  finish_frame() assumes that there is an output buffer.

2015-03-05 09:56:23 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't unref caps we don't own

2015-03-05 09:46:17 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Push RTCP caps on the RTCP pads
	  Otherwise we will get not-negotiated later from rtpbin, and will never be able
	  to send RTCP packets back to the server. Note that error flow returns from the
	  RTCP pads are ignored, that's why it didn't fail more visible before.

2015-03-05 09:35:32 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Make sure to send SEGMENT events on all pads

2015-03-03 16:23:15 +0100  Santiago Carot-Nemesio <sancane@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpstats.h:
	  rtp: Add Full Intra Request (FIR) packets to statistics
	  https://bugzilla.gnome.org/show_bug.cgi?id=745587

2015-03-03 16:01:53 +0100  Santiago Carot-Nemesio <sancane@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpstats.h:
	  rtp: Add Packet Loss Indication (PLI) to statistics
	  This is helpful to provide statistics in the format defined in
	  http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745587

2015-03-03 19:19:50 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: Remove duration accumulation logic
	  Duration accumulation can cause rounding errors and generate wrong
	  duration with different buffers that share the same timestamp.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745192

2015-03-03 18:40:16 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroska: Add an helper method to get buffer timestamps
	  ... and replace GST_BUFFER_TIMESTAMP that always return PTS with this method
	  that return PTS or DTS based on stream type.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745192

2015-03-04 11:28:12 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Add explanation why we have space for 32 hash tables
	  And also create only one, there's no need yet to create all 32 until
	  we implement RFC2762.

2015-03-04 11:26:57 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  Revert "rtpsession: Do not use an array of maps if they are not being used"
	  This reverts commit 1591adf4cd843d13d8622a30c619425691a84128.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
	  It's the beginning of an implementation of RFC 2762, which is needed for
	  large multicast groups. The implementation is not yet complete but why
	  not leave what is there and implement RFC 2762 instead?

2015-03-04 10:35:12 +0100  Santiago Carot-Nemesio <sancane@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Do not use an array of maps if they are not being used
	  rtpsession declares an array of maps to store srrcs but only the
	  the key 0 is being used. This patch replaces the array of maps
	  for just one map and remove useless parameters in rtpsession
	  https://bugzilla.gnome.org/show_bug.cgi?id=745586

2015-02-27 18:12:09 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/avi/gstavidemux.c:
	  avidemux: remove not needed code
	  In gst_avi_demux_handle_src_query, there is not needed code.
	  We already check about stream is vbr or not at the upper line.
	  o, we don't need to check this condition becase stream is not
	  vbr 100% in this case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745276

2015-03-03 23:25:35 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/gdkpixbufoverlay-test.c:
	  tests: gdkpixbufoverlay-test: replace deprecated function
	  Just avoid using the deprecated function entirely,
	  it's easy enough. Defining the macro is not enough.

2015-03-03 19:04:48 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/gdkpixbufoverlay-test.c:
	  tests: gdkpixbufoverlay-test: fix compilation against newer gdk-pixbuf
	  gdk_pixbuf_new_from_inline() has been deprecated in favour
	  of GResource.

2015-03-03 18:39:15 +0530  Arun Raghavan <arun@centricular.com>

	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudiosrc: Allow caps renegotiation
	  The ringbuffer does allow renegotiation, so we do not have to report
	  fixed caps once it is acquired (based on a similar patch for the sink
	  side by Ilya Konstantinov <ilya.konstantinov@gmail.com>).

2015-02-21 14:41:08 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: Allow renegotiating caps
	  Once osxaudiosink's device is open, it fixates on the initial caps and
	  refuses to accept new caps. This is erroneous since the Audio Unit is
	  can accept a new ASBD, and GstAudioRingBuffer supports reconfiguration
	  as well.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743925

2015-03-02 12:04:00 +0100  Gwenole Beauchesne <gwenole.beauchesne@intel.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2allocator: fix fd leak in DMABUF import mode.
	  Ensure gst_v4l2_buffer_pool_release_buffer() releases the associated
	  GstV4l2MemoryGroup. In particular, this allows for closing the DMABUF
	  handles prior to instantiating new ones.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745443

2015-03-02 15:06:09 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Use 0 as duration for the EOS "frame"

2015-03-02 15:02:20 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	  vp{8,9}enc: Tell the encoder about actual timestamps and durations of frames
	  ... instead of just counting frames. The values are supposed to be in timebase
	  units, not frame units. This fixes various quality problems with VP8/VP9
	  encoding and in general makes the encoder behave better.
	  Thanks to Nirbheek Chauhan for noticing this bug.

2015-03-01 13:56:17 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  vpxdec: Fix calculation of width in bytes
	  Right now we only support I420, but vpx seems to support more formats.
	  This will prevent hard to find bug in the future.

2015-03-01 13:52:50 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  vpxdec: Don't memcpy in frame map failed
	  This avoid a crash if mapping the frame failed.

2015-03-01 13:48:45 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Add missing break
	  This is cosmetic change.

2015-03-01 13:46:18 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: Workaround driver not setting field correctly
	  As it's very common, handle driver not setting field in buffers
	  by using the field value from the format. This workaround a long time
	  bug in UVC driver. For even buggier driver, we simply assume
	  progressive as before. We also only warn once, to avoid spamming.

2015-02-28 18:10:06 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix key unit seek
	  Unlike many other seek flags, the KEY_UNIT seek
	  flag is not copied over into the GstSegment,
	  since it's only relevant for the seek itself,
	  so we need to pass it explicitly to the seek
	  handler here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745339

2015-02-27 09:38:01 +0100  Edward Hervey <bilboed@bilboed.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	  docs/plugins: Updates

2015-02-26 23:41:47 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroskamux/demux: initialize dts_only
	  https://bugzilla.gnome.org/show_bug.cgi?id=745192

2015-02-26 23:28:11 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: store DTS for V_MS/VFW/FOURCC streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=745192

2015-02-26 19:48:33 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsrc.c:
	  multifile: attempt to fix docs build issue on build bot

2015-02-27 00:41:46 +0530  Arun Raghavan <git@arunraghavan.net>

	* gst/interleave/interleave.c:
	  interleave: Drop custom latency query handling
	  This is implemented by the default query handler now.

2015-02-27 00:40:05 +0530  Arun Raghavan <git@arunraghavan.net>

	* gst/videomixer/videomixer2.c:
	  videomixer: Drop custom latency querying logic
	  This is now implemented in the default latency query handler.

2015-02-26 16:10:41 +0100  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: fix payloader description and author e-mail
	  https://bugzilla.gnome.org/show_bug.cgi?id=745226

2014-09-05 16:34:26 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2: query crop configuration after each call of S_CROP
	  S_CROP ioctl is write-only and the device can adjust crop rectangle so
	  we query back the crop configuration after each S_CROP to know what has
	  been done.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736133

2015-02-26 02:12:18 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: V_MS/VFW/FOURCC streams have DTS instead of PTS
	  When such stream is present demuxer should set DTS on buffers instead
	  of PTS. This is consistent with how VLC and libav/ffmpeg handle VFW
	  streams.
	  Sample file
	  https://s3.amazonaws.com/MatejK/Samples/Matroska-VFW-DTS-Only.mkv
	  https://bugzilla.gnome.org/show_bug.cgi?id=745192

2015-02-25 16:45:11 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Check corruption flag on the right buffer
	  We where checking the buffer we are copying to instead of the buffer we
	  are copying from.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740040

2015-01-19 15:29:24 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: set colorspace in caps for capture devices
	  This information is set by the driver for a capture device, and so could
	  be forwarded to pipeline by setting the colorimetry in caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743186

2014-10-06 17:30:06 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2bufferpool: fix import_userptr() in single-planar API when n_planes > 1
	  In the V4L2 single-planar API, when format is semi-planar/planar,
	  drivers expect the planes to be contiguous in memory.
	  So this commit change the way we handle semi-planar/planar format
	  (n_planes > 1) when we use the single-planar API (group->n_mem == 1).
	  To check that planes are contiguous and have expected size, ie: no
	  padding. We test the fact that plane 'i' start address + plane 'i'
	  expected size equals to plane 'i + 1' start address. If not, we return
	  in error.
	  Math are done in bufferpool rather than in allocator because the
	  former is aware of video info.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738013

2015-01-23 10:15:46 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2allocator: let bufferpool calculate image size when importing userptr
	  Offset are relative to the buffer and there is no guarantee substracting
	  them will give us the plane size. So we let bufferpool make the math as
	  it is more aware of video info than allocator and pass a size array to
	  allocator import function.
	  Pointed out by Nicolas Dufresne <nicolas.dufresne@collabora.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=738013

2014-12-11 16:13:15 +0100  Philippe De Muyter <phdm@macqel.be>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: recognize and distinguish all bayer arrangements
	  Up to now, v4l2src recognized only "bggr" amongst the bayer arrangements.
	  Recognize now also the "rggb", "gbrg" and "grbg" arrangements.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742363

2015-01-15 16:11:53 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: set v4l2_buffer.field when queuing buffer in an output device
	  According to the current specification, application must set this field
	  for an output device.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743013

2015-02-24 05:57:24 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiocommon.h:
	  osxaudiosrc: iOS resampling causes stuttering
	  Fixes stuttering audio when iOS AU is resampling. To make AU resample,
	  one has to request a rate that differs from AVAudioSession's
	  sampleRate. The resampling itself is not the culprit, but rather our
	  API misuse.
	  AudioUnitRender modifies the mDataByteSize members with the
	  actual read bytes count. Therefore, they must be reinitialized
	  before each AudioUnitRender. (The buffers themselves can be
	  preallocated.)
	  The "stutter" was caused by one AudioUnitRender making the buffer
	  too small for other AudioUnitRender invocations, making them fail
	  with -50 (paramErr). By way of luck, when AU didn't resample, all
	  AudioUnitRender invocations read the same number of bytes.
	  (This patch addresses some non-interleaved audio concerns, but
	  at this moment the elements do not support non-interleaved audio
	  and non-interleaved is untested.)
	  https://bugzilla.gnome.org/show_bug.cgi?id=744922

2015-02-22 01:49:52 +0100  Krzysztof Kotlenga <pocek@users.sf.net>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: improve error message when unauthorized
	  Make use of NOT_AUTHORIZED error code instead of falling back to generic
	  READ error.
	  https://bugzilla.gnome.org/show_bug.cgi?id=601733

2015-02-23 20:06:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/ximage/ximageutil.c:
	  ximagesrc: remove pointless g_return_val_if_fail()
	  ximage won't ever be NULL here because the dispose
	  function is called via ximage->dispose().

2015-02-23 19:40:25 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: All segment resulting from a seek should have the same seqnum
	  https://bugzilla.gnome.org/show_bug.cgi?id=744983

2015-02-19 23:12:31 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: Enable copy when no known allocation params
	  When there is no allocation parameters in the query, enable copy
	  threshold. When this threshold is reached, the buffer pool will start
	  copying when the pool reaches a critical level. If the driver supports
	  CREATE_BUFS, this will be used instead.

2015-02-19 23:08:34 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Update allocator flags
	  When we hit emulated formats, we disable CREATE_BUFS since libv4l2
	  cope very badly with it. Also clear the allocator flags so we will
	  never try to allocate more buffers. This fixes failure when the copy
	  threshold is reached as we where calling CREATE_BUFS, which lead to
	  libv4l2 instability.

2015-02-19 23:07:23 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Use specific debug category
	  The pool has grown enough that it is now handy to seperate v4l2object
	  trace from v4l2bufferpool trace.

2015-02-19 14:29:02 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: default encoding name to VP8
	  https://bugzilla.gnome.org/show_bug.cgi?id=737810

2015-02-19 14:06:51 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: make caps writable before truncating them
	  https://bugzilla.gnome.org/show_bug.cgi?id=737810

2015-02-05 10:29:26 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: negotiate encoding name
	  Chrome uses a different one than gstreamer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737810

2015-02-19 12:35:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Send initial events on sync_rtcp pad when using RTP/RTCP muxing
	  Otherwise we will just send buffers on the pad without any events beforehand
	  and will get g_warnings() about that.

2015-02-19 11:20:51 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* ext/jack/gstjackaudiosrc.c:
	  jack: case missing break statement
	  commit b1098c2ea5eabea7af08ce51d22b867eaed2bbe2 added a new case in
	  gst_jack_audio_src_get_property() but forgot to add the break statement to it.

2015-02-18 19:18:00 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* sys/v4l2/v4l2_calls.c:
	  Revert "v4l2: fraction is reversed"
	  This reverts commit b91fe36644b15ae070d72b9e8a9c7087e82aef12.

2015-02-18 17:49:29 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: fraction is reversed
	  In the fraction 1 / 2. 1 is the numerator and 2 is the denominator.
	  The arguments of fraction gst_value_set_fractions() are value,
	  numerator and denominator.
	  Also, gst_value_set_fraction() fails if denominator is 0 for obvious
	  reasons.

2015-02-17 20:26:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2pool: Deactivate other pool
	  When importing buffers from a downstream pool, we need to deactivate
	  that pool to ensure it will be usable again later. Relying on the
	  refcount to reach zero does not work, since elements like xvimagesink
	  keeps a reference on their proposed pool.

2015-02-18 10:10:53 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	  qtmux: remove not needed condition
	  gst_buffer_replace can handle NULL inputs by itself

2015-02-18 09:40:14 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: prefer the tfdt timestamp over the buffer's that is less accurate
	  The tfdt should be more accurate as the buffer timestamp is provided
	  by the fragmented format manifest and it might just be an approximation.

2015-02-17 16:57:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: When resetting the jitterbuffer because of packet discont, don't flush sticky events
	  We will otherwise flush away STREAM_START, CAPS or SEGMENT events and will
	  confuse downstream with buffers that come before such events.

2015-02-17 12:20:57 +0100  hark <hark@puscii.nl>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: Add property port-pattern to specify which JACK ports to connect to
	  https://bugzilla.gnome.org/show_bug.cgi?id=690719

2015-02-17 12:31:06 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/gstisoff.c:
	* gst/isomp4/gstisoff.h:
	* gst/isomp4/qtdemux.c:
	  isomp4: Redefine gst_isoff_ symbols to gst_isoff_qt_
	  We need different symbol names, because these symbols are also present
	  in the fragmented plugin ... which will cause conflicts when doing
	  static linking

2015-02-16 14:31:05 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom2k1/lines.c:
	  goom2k1: use fractional part of float division

2015-02-16 13:59:14 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsin: remove dead code
	  Every instance of goto beach has buf_info equal NULL. Don't check
	  for a condition that never happens.
	  CID #1268399

2015-02-15 21:45:24 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* tests/check/elements/splitmux.c:
	  splitmux-test: Parse error message
	  The test had a function to print the error, but was not parsing it.
	  This was causing warning about dbg_info being used uninitialized. If
	  the test was testing any errors, this would have crashed.

2015-02-15 21:34:28 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	  spectrum: Fix min and max for bands property
	  The number of FFTs is calculated with the following formula:
	  guint nfft = 2 * bands - 2;
	  nfft is passed to gst_fft_f32_new() as the len argument and is of type
	  unsigned integer. This method required that len is at leas 1, then
	  maximum G_MAXINT, as other values would be negative. If we extrapolate
	  from the formula above it means we need "bands" to be between 2 and
	  ((guint)G_MAXINT + 2) / 2).
	  https://bugzilla.gnome.org/show_bug.cgi?id=744213

2015-02-15 15:51:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix freeing of shared memory
	  When memory (that has been shared using gst_memory_share()) are freed,
	  the memory (or the DMABUF FD) should not bee freed. These memories have
	  a parent. This also removes the extra _v4l2mem_free function and avoid
	  calling close twice on the DMABUF FD.
	  https://bugzilla.gnome.org/show_bug.cgi?id=744573

2015-02-14 11:11:30 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: do not use sparse streams in push-based seeking
	  Using the sparse streams can make the push-based seeking return
	  too far in the stream. It also can lead to issues as the
	  sparse streams will be ignored when restarting playback and,
	  if the sparse stream is the one that has the earliest sample,
	  it will confuse qtdemux's offsets as one stream will have
	  an earlier offset than the demuxer's one which might lead to
	  early EOS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742661

2015-02-13 19:43:16 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Enhance code readability in pulsesink_query
	  In pulsesink_query function, we use a switch for the query
	  type. In the CAPS case, there is no 'break', instead we
	  return right away. Use a break and return at the end of
	  the function instead for better code readability.
	  https://bugzilla.gnome.org/show_bug.cgi?id=744461

2015-02-13 20:40:48 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: flag as sink from the start

2015-02-11 15:30:44 +0100  Philippe Normand <philn@igalia.com>

	* gst/isomp4/Makefile.am:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstisoff.c:
	* gst/isomp4/gstisoff.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Initial 'sidx' atom parsing support
	  Parse the 'sidx' atom and update the total duration according to the
	  parser result. The isoff parser code is imported from
	  gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data()
	  function was factored out of the gst_isoff_sidx_parser_add_buffer()
	  function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743578

2015-02-11 05:06:45 +1100  Jan Schmidt <jan@centricular.com>

	* gst/flv/Makefile.am:
	* gst/flv/gstflvdemux.c:
	  flvdemux: Use gst_video_guess_framerate()
	  Use gst_video_guess_framerate() from libgstvideo to guess
	  sensible common framerates where possible from the
	  floating point fps in the stream.

2015-02-11 13:53:02 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/raw1394/gstdv1394src.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	* gst/interleave/interleave.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/videomixer/videomixer2.c:
	  Improve and fix LATENCY query handling
	  This now follows the design docs everywhere, especially the maximum latency
	  handling.
	  https://bugzilla.gnome.org/show_bug.cgi?id=744106

2015-02-11 10:29:55 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Handle first RTCP packet and early feedback correctly
	  According to RFC 4585 section 3.5.3 step 1 we are not allowed to send
	  an early RTCP packet for the very first one. It must be a regular one.
	  Also make sure to not use last_rtcp_send_time in any calculations until
	  we actually sent an RTCP packet already. In specific this means that we
	  must not use it for forward reconsideration of the current RTCP send time.
	  Instead we don't do any forward reconsideration for the first RTCP packet.

2015-02-10 18:53:53 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: fix compilation with gcc 5.0

2015-02-10 16:00:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: fix example pipeline properly
	  x264enc might not have a max-key-int property, but it
	  has a key-int-max property...

2015-02-10 14:57:55 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmux: fix typo

2015-02-10 14:56:23 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: update example pipeline
	  Element x264enc doesn't have a max-key-int property

2015-02-10 13:29:32 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: fix memory leak
	  If execution goes to the beach in line 981, buf_info goes out of scope without
	  the memory being free'd. Handle this case.
	  CID #1268403

2015-02-08 12:03:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix awkward if clause

2015-02-07 01:41:49 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxsink.c:
	* tests/check/elements/splitmux.c:
	  splitmux: Add unit test for file splitting
	  Add a unit test for file splitting, and fix the leaks in the
	  splitmuxsink it found

2015-02-06 14:43:22 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: fix which stop variable is used in assignment
	  Assignment is done to variable segment.stop when the intention was to assign to
	  local variable stop. Instead of overwriting it, the value is now clamped and
	  segment.stop is set to it soon after.
	  CID #1265773

2015-02-07 00:19:36 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxsrc.c:
	* tests/check/elements/splitmux.c:
	  splitmux: Fix memory leaks until the test valgrinds clean

2015-02-06 06:42:17 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmux: Handle early EOS during part preparation
	  Handle the case where a short file reaches EOS while we're still
	  waiting for no-more-pads, and make sure we continue to the internal
	  READY state for real playback to work properly later.

2015-02-06 05:03:19 +1100  Jan Schmidt <jan@centricular.com>

	* tests/files/splitvideo00.ogg:
	* tests/files/splitvideo01.ogg:
	* tests/files/splitvideo02.ogg:
	  tests: Change splitmux test video files
	  Avoid test failure by changing the stored video resolution
	  from 80x60 to 80x64, which needs bug 741030 to be fixed.

2014-08-01 00:07:53 +1000  Jan Schmidt <jan@centricular.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* gst/multifile/Makefile.am:
	* gst/multifile/gstmultifile.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	* gst/multifile/gstsplitmuxsrc.c:
	* gst/multifile/gstsplitmuxsrc.h:
	* gst/multifile/gstsplitutils.c:
	* gst/multifile/gstsplitutils.h:
	* gst/multifile/test-splitmuxpartreader.c:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/splitmux.c:
	* tests/files/splitvideo00.ogg:
	* tests/files/splitvideo01.ogg:
	* tests/files/splitvideo02.ogg:
	  splitmux: Implement new elements for splitting files at mux level.
	  Implement 2 new elements - splitmuxsink and splitmuxsrc.
	  splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
	  plus audio/subtitle streams, and starts a new file
	  whenever necessary to avoid overrunning a threshold of either bytes
	  or time. New files are started at a keyframe, and corresponding audio
	  and subtitle streams are split at packet boundaries to match
	  video GOP timestamps.
	  splitmuxsrc is a corresponding source element which handles
	  the splitmux:// URL and plays back all component files,
	  reconstructing the original elementary streams as it goes.

2015-02-04 16:32:14 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/souphttpsrc.c:
	* tests/files/test-cert.pem:
	* tests/files/test-key.pem:
	  tests: souphttpsrc: update ssl key/cert pair
	  Our ones were expired. The new ones were copied from libsoup's
	  tests files.
	  Also sets the property to use our own cert to validate the
	  server, otherwise the default system certs would be used
	  and it would fail.

2015-02-04 02:25:44 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: prevent trying to get 0 bytes from adapter
	  This causes an assertion and would lead to getting a NULL instead
	  of a buffer. Without proper checking this would easily lead to
	  a segfault
	  https://bugzilla.gnome.org/show_bug.cgi?id=737199

2015-02-04 21:50:51 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Simple implementation of GST_SEGMENT_FLAG_TRICKMODE_KEY_UNITS
	  When the trickmode key-units flag is set on the segment, simply skip
	  any sample on a video stream that isn't a keyframe

2015-02-03 17:35:52 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix container handling
	  We detect a container correctly now so we need to revert the weird
	  check there was before.
	  Use gst_rtspsrc_stream_push_event() to push the caps event on the
	  right pad.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=739391

2015-02-02 19:46:27 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: store and write stream tags
	  Separate global from stream tags storage and write them to the
	  appropriate tags entry in the output

2015-02-02 13:35:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: parse stream tags
	  Keep global and stream tags separately and parse the udta node
	  that can be found under the trak atom. The udta will contain
	  stream specific tags and will be pushed as such
	  https://bugzilla.gnome.org/show_bug.cgi?id=692473

2015-01-31 14:32:34 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: store stream and container tags separately
	  Tags received via events, when marked as stream tags, will
	  be stored on that stream's trak atom instead of being stored
	  in the main tags atom. This allows the resulting file to have
	  global and stream tags stored.
	  https://bugzilla.gnome.org/show_bug.cgi?id=692473

2015-01-31 13:14:44 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: refactor tags functions to accomodata UDTA at trak level
	  Refactor the functions that were bound to the 'moov' atom to
	  directly pass the desired 'udta' that should receive the tags.
	  This allows the tags to be written to 'udta' at the 'moov' or
	  the 'trak' level, creating tags that are for the container or
	  for a stream only.
	  https://bugzilla.gnome.org/show_bug.cgi?id=692473

2015-01-31 10:47:40 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: map application name to _swr tag
	  It refers to the application name and version used to create the
	  file
	  https://bugzilla.gnome.org/show_bug.cgi?id=692473

2015-01-31 02:30:40 +1100  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: Fix seeking past the end of the file in reverse mode.
	  Snap to the end of the file when seeking past the end in reverse mode,
	  and also fix GST_SEEK_TYPE_END and GST_SEEK_TYPE_NONE handling
	  for the stop position by always seeking on a segment in stream time

2015-01-30 18:22:31 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Fix signal name
	  This wasn't meant to be pushed at all yet, but now that it's there
	  already it won't hurt to make it correct at least.

2015-01-30 16:56:35 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpstats.h:
	  rtpstats: Fix typo in documentation

2015-01-30 16:50:36 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Add new on-receiving-rtcp signal
	  This will be emitted whenever an RTCP packet is received. Different to
	  on-feedback-rtcp, this signal gets every complete RTCP packet and not
	  just the individual feedback packets.

2015-01-28 14:02:15 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: simplify segment.base math
	  Remove a fix for heavily edited files added for fixing
	  https://bugzilla.gnome.org/show_bug.cgi?id=345830 to work
	  with seeks and proper gaps playback. The fix was replaced
	  for a more general solution that bases on using previous
	  segment's duration, just like it works for media segments
	  playback.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743518

2015-01-27 14:00:35 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/videomixer/videomixerorc-dist.c:
	  videomixer: update orc files

2015-01-26 17:08:12 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix data dropping for fragmented streams
	  For fragmented streams with extra data at the end of the mdat
	  qtdemux was not dropping those bytes and would try to use
	  that extra data as the beginning of a new atom, causing the
	  stream to fail.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743407

2015-01-25 17:30:33 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Deprecate rtcp-immediate-feedback-threshold property
	  It had no effect since quite some time and also is not needed in general,
	  especially not to switch between immediate feedback mode and early feedback
	  mode. The latest understanding of the RFC is that from the endpoint point of
	  view, both modes are exactly the same. RTCP is only allowed to use the
	  bandwidth as given by the RFC constraints, as such it is only ever possible
	  to schedule a RTCP packet early but it's against the RFC to schedule more RTCP
	  packets.
	  The difference between immediate feedback mode and early feedback mode is that
	  the former guarantees that an RTCP packet can be sent for every event
	  "immediately", which means that the bandwidth calculations from the RFC have
	  resulted in an RTCP scheduling interval that is small enough. Early feedback
	  mode on the other hand means that we can schedule some packets early to make
	  that happen, but it's not guaranteed at all that it's possible to schedule
	  an RTCP packet per event (i.e. they need to be accumulated or dropped).

2015-01-22 10:29:39 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Delay the next regular RTCP packet after early RTCP
	  This is required to not exceed the short term average RTCP bitrate when
	  using early feedback as compared to without early feedback.

2015-01-22 10:28:52 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Add new send-rtcp-full signal
	  This indicates with a boolean return value if scheduling a new RTCP packet
	  within the requested delay was possible. Otherwise it behaves exactly like
	  send-rtcp. The only reason for adding a new signal is ABI compatibility.

2015-01-20 00:32:00 +0000  Jimmy Ohn <yongjin.ohn@lge.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Free format_info in query_getcaps
	  If we can not create probe stream in query_getcaps function, it will appear
	  memory leakage from format info.
	  The following patch prevent memory leakage in pulsesink.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743178

2015-01-23 17:35:51 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: remove unnecessary check
	  No matter if gst_matroska_read_common_parse_index_cuetrack () returns that the
	  flow is OK or not, the check there will be a break from the switch. Removing the
	  check since the outcome is the same.
	  CID #1265762

2015-01-23 15:16:25 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Avoid using freed variable
	  the name variable might have been attributed to pad_name, make sure we
	  free it only *after* pad_name has been used.
	  Coverity CID : 1265774

2015-01-23 15:13:55 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavimux.c:
	  avimux: Avoid using freed variable
	  the name variable might have been attributed to pad_name, make sure we
	  free it only *after* pad_name has been used.
	  Coverity CID : 1265775

2014-11-14 12:59:31 +0100  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: reuse caps framerate if not overwritten by v4l2 device
	  Enables duration setting in v4l2src.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740403

2015-01-22 10:29:24 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Fix indention

2015-01-21 17:36:26 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux_dump: Bypass even more code if debugging is disabled
	  And avoid using variables that won't exist when debugging is disabled

2015-01-21 15:30:33 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux: Only traverse/dump nodes if guaranteed to be used
	  __gst_debug_min is the "global" lowest debug level set. There's no
	  guarantee the qtdemux debug category is actually set at that level.

2014-12-20 17:09:14 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/ebml-read.c:
	  matroska: Avoid debugging below category threshold
	  This part alone was what made the matroska thread take a full core
	  on an android phone ...

2015-01-21 09:56:41 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/twolame/gsttwolamemp2enc.c:
	  Constify some static arrays everywhere

2015-01-21 09:56:41 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/lame/gstlamemp3enc.c:
	  Constify some static arrays everywhere

2015-01-21 09:55:30 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/dv/gstsmptetimecode.c:
	* ext/mikmod/mikmod_types.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audiopanorama.c:
	* gst/effectv/gstradioac.c:
	* gst/isomp4/atoms.c:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/qtdemux.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/wavparse/gstwavparse.c:
	  Constify some static arrays everywhere

2015-01-19 17:49:54 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix deadlock seeking in files without seek entries
	  A mutex unlock was missing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739975

2015-01-19 12:34:25 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videomixer/blend.c:
	  videomixer: fix illegal memory access in blend function with negative ypos
	  https://bugzilla.gnome.org/show_bug.cgi?id=741115

2015-01-13 16:49:34 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Proxy getcaps
	  Replace the sink_query with new getcaps() virtual and use the proxy
	  helper with the probed caps. This allow upstream element taking decision
	  base on what is supported downstream.

2015-01-13 19:05:20 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Add support for v210

2015-01-13 18:58:01 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: v210 is v210, not UYVY and yuv2 is YUY2, not I420
	  Also add a few other raw video formats we support: v308, v216
	  and add comments for a few others we don't support yet.
	  https://developer.apple.com/library/mac/technotes/tn2162/

2015-01-12 15:56:29 +0100  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From f2c6b95 to bc76a8b

2015-01-10 15:51:16 +0100  Sebastian Dröge <sebastian@centricular.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: Disable hack for NSApp iteration with a special #define
	  The hack causes deadlocks and other interesting problems and it really
	  can only be fixed properly inside GLib. We will include a patch for
	  GLib in our builds for now that handles this, and hopefully at some
	  point GLib will also merge a proper solution.
	  A proper solution would first require to refactor the polling in
	  GMainContext to only provide a single fd, e.g. via epoll/kqueue
	  or a thread like the one added by our patch. Then this single
	  fd could be retrieved from the GMainContext and directly integrated
	  into a NSRunLoop.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741450
	  https://bugzilla.gnome.org/show_bug.cgi?id=704374

2015-01-08 21:07:05 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: uncork if needed upon commit
	  ... to provide for a running clock.

2015-01-09 16:59:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Prevent renegotiation
	  Renegotiation isn't supported, simply prevent it the way we do in
	  v4l2src.

2015-01-06 13:54:25 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Don't unlock the stream lock twice

2015-01-09 11:40:40 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix stream time conversion
	  Use the right macro to convert to the correct scale or the
	  segment information will be wrong
	  https://bugzilla.gnome.org/show_bug.cgi?id=742572

2015-01-07 18:48:58 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Add protection against driver bug
	  v4l2loopback driver has a this nasty bug that if the queue is larger
	  then 2 buffers, it returns random index on dqbuf. So far we assumed
	  that the index was always right, which would lead to memory being
	  unref twice, and eventually crash.

2015-01-07 17:58:05 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Don't use allocator size to iterate
	  As the buffer array is fixed size and small, it's safer to simply
	  use this static size to cleanup the buffers. This is also more
	  consistent with the rest. The associated method is no longer
	  required and can be dropped.

2015-01-07 17:55:14 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Don't clean buffer array in dispose
	  This should already have been done, plus this code is incorrect
	  and may lead to crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742074

2015-01-07 17:48:31 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Don't ref queued output buffer
	  This partly revert to the old 1.2 behavior. Instead of keeping a
	  reference to the output buffer queued, we simply release them but
	  don't forward it to GstBufferPool. This way, the buffer pool don't
	  need to be flushed to be stopped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742074

2015-01-08 11:37:23 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Never fail on streamoff
	  Failing streamoff prevents allocator from being disposed hence
	  lead to device FD leak. There is no known cases where streamoff
	  may fails for which we'd still be streaming. streamoff is known
	  to fail when a device is being unplugged (in which case errno
	  19/ENODEV is set).
	  https://bugzilla.gnome.org/show_bug.cgi?id=732734

2015-01-07 21:52:17 -0500  Brad Smith <brad@comstyle.com>

	* configure.ac:
	  v4l2: Add support for detecting the presence of V4L2 support on OpenBSD
	  https://bugzilla.gnome.org/review?bug=742503

2015-01-04 15:57:10 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: request at least 8 bytes to properly parse header
	  https://bugzilla.gnome.org/show_bug.cgi?id=742325

2015-01-07 16:20:03 -0800  Michael Smith <michael.smith@rdio.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: skip an additional uninteresting chunk type before the fmt chunk.

2015-01-07 18:16:12 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/audiofx/audiodynamic.c:
	  audiodynamic: assert func_index is inside bounds
	  Bringing back the check removed in the previous commit but have that check be a
	  g_assert. Changing the function to static void since return can never be False,
	  because audio format will never be unkown.

2015-01-07 17:31:39 +0000  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/audiofx/audiodynamic.c:
	  audiodynamic: remove always-true conditional
	  func_index is set by the sum of three ternary operators which add, 0:4, 0:2,
	  and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7.
	  The conditional checking if func_index is >= 0 and < 8 will always be true.
	  Removing it.
	  CID 1226442

2015-01-07 18:05:18 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: If we get a gap with a buffer without DTS, error out
	  We (currently?) can't really handle gaps between RTP packets if they're not
	  properly timestamped. The current code would go into calculations with
	  GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
	  better to error out cleanly instead.

2014-11-21 11:39:19 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: set PLAYING state after configuring caps
	  We set to PLAYING after we have configured the caps, otherwise we
	  might end up calling request_key (with SRTP) while caps are still
	  being configured, ending in a crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740505

2014-12-30 18:03:22 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/gdkpixbufoverlay-test.c:
	  tests: gdkpixbufoverlay-test: remove outdated FIXME

2014-12-30 17:19:42 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtpcollision.c:
	  tests: rtpcollision: use alawenc/dec in these tests instead of Speex
	  They should always be built, while the speex elements are not.
	  Need to check for a smaller number of buffers then (7->4) because
	  speexenc will add 3 header buffers while alawenc will just output
	  as many buffers as it receives as input.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742098

2014-12-30 16:36:02 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/pipelines/simple-launch-lines.c:
	  tests: simple-launch-lines: only run jpeg/png tests if elements are available

2014-12-30 16:26:58 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't return a buffer when returning not GST_FLOW_OK
	  basesrc assumes that we don't return a buffer if
	  something else than OK is returned. It will just
	  leak any buffer we might accidentially provide
	  here.
	  This can potentially happen during flushing.
	  Maybe fixes https://bugzilla.gnome.org/show_bug.cgi?id=741993

2014-12-30 14:52:42 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtpaux.c:
	  tests: rtpaux: use alawenc/dec in these tests instead of Speex
	  They should always be built, while the speex elements are not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742098

2014-12-29 15:35:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Improve detection of being stuck at the same offset
	  Only error out if we read from the same position again and got the
	  same length. Just the same position is not necessarily enough.

2014-12-29 15:00:02 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't get stuck at the same offset when searching for clusters
	  This could happen if there is an invalid cluster with size 0, and in that
	  case just error out instead of looping forever.

2014-12-25 21:32:40 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: fix ALAC muxing
	  Actually copy the codec data instead of copying nothing
	  and then bombing out because there's no data.
	  Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink
	  https://bugzilla.gnome.org/show_bug.cgi?id=741783

2014-12-25 15:48:04 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: just drop invalid rtp packets instead of erroring out
	  Apparently linphone sends an invalid RTP packet as very
	  first packet. We want to ignore that instead of erroring
	  out (same for any other invalid packets really).
	  https://bugzilla.gnome.org/show_bug.cgi?id=741398

2014-12-25 15:44:15 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: fix 0.10-ism in docs

2014-12-25 14:58:12 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/gdkpixbufoverlay-test.c:
	  tests: gdkpixbufoverlay-test: use absolute positioning to fix demo
	  https://bugzilla.gnome.org/show_bug.cgi?id=739566

2014-12-25 14:53:09 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	  gdkpixbufoverlay: add "positioning-mode" property to allow absolute positions
	  Set positioning-mode=pixels-absolute to allow positioning with
	  absolute coordinates, meaning negative x/y offsets will be
	  interpreted as being to the left/above the video frame instead
	  of being interpreted as relative to the right/bottom edge of
	  the video frame (which is a silly default, but that's how it is).
	  This means we can nicely slide images into and out of the frame,
	  see gdkpixbufoverlay-test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739566

2014-12-22 15:33:51 +0100  Sebastian Dröge <sebastian@centricular.com>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudio: Directly return the ringbuffer's caps if it is acquired

2014-12-22 12:56:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudio: Put all audio formats into the template caps
	  We report the proper caps later from the get_caps() vfunc implementation after
	  probing the selected device.

2014-12-22 12:56:05 +0100  Sebastian Dröge <sebastian@centricular.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	  osxaudio: Also set the big endian flag for floating point samples

2014-12-22 11:45:59 +0100  Sebastian Dröge <sebastian@centricular.com>

	* MAINTAINERS:
	  MAINTAINERS: Update my mail address

2014-12-22 10:23:01 +0100  Sebastian Dröge <sebastian@centricular.com>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudio: Fix deadlock and property change notification in device selection code
	  After creating the ringbuffer we have to set the device on the ringbuffer as
	  it defaults to kAudioDeviceUnknown. At this point it can't have changed to
	  anything else yet and we don't have to notify about changes to the sink/src
	  "device" property. It's also not a good idea because GstAudioBaseSrc has the
	  object lock taken while the ringbuffer is created, which might cause a
	  deadlock if something calls back into the element from "notify::device".
	  Once the base class is done with the NULL_TO_READY state change, it has opened
	  the device via the ringbuffer and this might have chosen a different device.
	  Especially if we initially used kAudioDeviceUnknown. Also notify about this
	  property change as initially intended by this code.

2014-12-19 12:30:03 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2pool: Update configuration size
	  We already update our copy of VideoInfo.size to proper size, now also
	  the configuration so the size matches on release.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741420

2014-12-19 10:57:29 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroska-demux: Cache upstream length
	  Instead of constantly querying upstream, just cache the last duration,
	  and in the unlikelyness we might have gone over query again before
	  deciding we are EOS.
	  Cut 15% cpu off matroskademux streaming thread (srsly...)

2014-12-17 17:36:18 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroska: mux/demux the OpusHead header
	  This is meant to be so (https://wiki.xiph.org/MatroskaOpus - while
	  it is marked as a draft, this part was confirmed to be correct on
	  IRC), and allows one to determine whether a demuxed stream is
	  multistream or not, and thus set the multistream caps field
	  accordingly. In turn, this means downstream does not have to guess.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740744

2014-12-18 11:50:33 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't dereference NULL if a suitable stream for the AUX element can't be found
	  CID 1258717

2014-12-18 10:53:39 +0100  Sebastian Dröge <sebastian@centricular.com>

	* common:
	  Automatic update of common submodule
	  From ef1ffdc to f2c6b95

2014-12-12 23:06:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  udpsink: allocate scratch space for render functions on the heap
	  and not the stack. Our allocations could get a bit too large
	  to be sure it's not going to cause trouble using the stack.

2014-06-24 01:16:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: re-use send_buffers() code path for render() function
	  It's like rendering a buffer list, just with one buffer.
	  Has the added advantage that if there are multiple clients
	  we can send the buffer to all the clients in one go.

2014-06-24 01:15:25 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: keep client list consistent during removals
	  We unlock and re-lock the client lock while emitting the
	  removed signal, which causes inconsistencies in the client
	  list vs. the client counts. Instead, remove the client from
	  the list already before emitting the signal and put it into
	  a temporary list of clients to be removed. That way things
	  look consistent to the streaming thread, but signal callbacks
	  can still do things like get stats from removed clients.

2014-06-24 00:56:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: fix client count after removal

2014-06-23 18:43:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: keep client list sorted by socket family
	  We make use of in the send_buffers() function if we
	  need to use different sockets to send to IPv4 and
	  IPv6 destinations.

2014-06-20 11:36:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: add sendmmsg-ready render_list function prototype
	  Add prototype for a render_list() function that can use a
	  sendmmsg-style g_socket_send_messages() function once it lands
	  in GLib. We can use this infrastructure to send multiple buffers
	  made up by multiple memories to multiple clients in one go, which
	  drastically reduces the number of syscalls made when sending
	  high-bitrate video streams.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732152

2014-06-19 19:16:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: make udp client structure refcounted
	  Use the refcount for memory management and keep track
	  of the number of duplicate clients in a separate
	  variable. This will be useful later, and means we
	  don't have to hold the OBJECT_LOCK all the time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732866

2014-06-19 18:31:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: keep count of number of unique and non-unique IPv4 and IPv6 clients
	  This will come in handy later.

2014-12-16 15:00:22 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Disable create_buf with libv4l2
	  Libv4l2 does not work with CREATE_BUFS. Instead of failing on random
	  error caused by libv4l2, disable CREATE_BUFS when an emulated format is
	  detected.

2014-12-09 17:39:12 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Add protection against broken libv4l2
	  It looks like libv4l2 support for CREATE_BUF is incomplete. That
	  combine with existing bugs may lead to crash in GStreamer. These
	  check will make it robust by:
	  - Checking create buf index isn't an already in used index
	  - Checking that the index out of QUERYBUF matches the requested
	  index

2014-12-16 16:37:24 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Add something to the debug logs if an RTX AUX element can't be added
	  ... because the application already has a signal handler set up here.

2014-11-21 14:13:34 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add retransmission support according to RFC4588
	  Based on the client-rtpaux example

2014-12-16 13:25:01 +0100  Wim Taymans <wtaymans@redhat.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: clear rectangle structures before use

2014-12-09 15:09:56 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Always set format
	  Right now we try to be clever by detecting if device format have
	  changed or not, and skip setting format in this case. This is valid
	  behaviour with V4L2, but it's also very error prone. The rational
	  for not setting these all the time is for speed, though I can't
	  measure any noticeable gain on any HW I own. Also, until recently,
	  we where doing get/set on the format for each format we where
	  probing, making it near to impossible that the format would match.
	  This also fixes bug where we where skipping frame-rate setting if
	  format didn't change.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740636

2014-12-15 18:30:01 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: Remove todo about caps filter
	  The filter is already interected.

2014-12-15 18:19:05 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	  videocrop: Make sure new crop is applied
	  Since "basetransform: Fix caps equality check" commit a7f357,
	  set_info() will not be called anymore if crop didn't change
	  the caps. This is fixed by setting "need_update" boolean when
	  cropping properties has been changed, and then applying these
	  if they where not applied before rendering the next frame. This
	  patch also fixed the locking, dropping un-needed custom lock,
	  and no holding needless lock while doing the operation as we
	  already hold the streaming lock.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740787

2014-12-12 18:10:35 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: Prefer filter caps order while getting caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-12-09 13:38:26 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: Add some error handling around channel layout parsing
	  For now we just spit a warning and ignore the channel layout if we can't
	  support it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-12-08 22:38:22 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudio: Take lock around sink/source before accessing the ringbuffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-12-01 21:06:27 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudiosrc: Probe channel layout too
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-12-01 20:32:04 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: Only fix up channels/layout for PCM caps while probing
	  It's unlikely that setting a channel layout will do much for AC3/DTS
	  streams. If we find at some point that it does make sense, we can
	  perform the structure copying unconditionally (i.e., the current code is
	  wrong, since AC3/DTS will get two structures now - one with the channel
	  layout, one without).
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-12-01 19:41:35 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxaudiosrc.h:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudiosrc: Implement caps probing
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-12-01 19:29:57 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: Bind audio device to audio unit early
	  We want to bind the device during open so that subsequent format queries
	  on the audio unit are as specific as possible from that point onwards.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-11-29 23:16:30 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: Fix up caps querying a bit
	  This should make caps queries correct in PAUSED and higher as well.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-11-28 22:32:36 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: Move osxaudiosrc-specific code out of the generic path
	  Avoids one layering violation (GstCoreAudio referring to
	  GstOsxAudioSrc).
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-11-28 22:23:17 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxaudioringbuffer.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: Clean up a GstCoreAudio -> GstOsxAudioSrc/Sink reference
	  Now that device selection has no sink/source-specific bits, we can have
	  generic device selection for this path. We do need to now track state
	  changes so we can look up the final device_id once the device is open,
	  though.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-11-28 19:40:52 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: Move device caps probing to get_caps()
	  This should be preferred to running the probe at device open time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-11-28 18:37:02 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: Make some debug code compile conditionally
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-11-28 15:06:35 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxaudioringbuffer.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudio: Move device selection to ringbuffer->open_device()
	  This is conceptually the right thing to do, and allows us to correctly
	  catch errors in device selection as well, which we could not do while
	  creating the ringbuffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-11-28 14:34:34 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: Consolidate input and output code paths a bit
	  https://bugzilla.gnome.org/show_bug.cgi?id=740987

2014-11-21 11:54:18 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/deinterlace/gstdeinterlace.c:
	  Deinterlace: in query_caps return only supported formats if filter is interlaced
	  In some cases the currently set GstVideoInfo is not interlaced, but
	  upstream caps are interlaced and the info is passed in the filter,
	  we should take that info into account and make sure that we do not
	  consider that case as a "pass through" case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741407

2014-12-12 11:06:17 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix debug statement
	  It was using the non-increasing offset variable, which made that statement
	  not so useful :)

2014-12-12 11:03:15 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add macros for the various timescale conversions
	  This helps make the code more readable and avoid future bad usage of
	  scaling function argument order.

2014-12-11 10:16:06 +0100  Patrick Radizi <patrickr@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: fix potential crash when shutting down
	  A race condition in the state change function may cause buffers
	  to be unreffed while they are still used by the streaming thread
	  in gst_rtp_h264_pay_send_sps_pps() resulting in a crash. Chain
	  up to the parent class first in the state change function to
	  make sure streaming has stopped and only then free those buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741381

2014-12-12 00:42:06 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Copy flags of the overall segment to output segments
	  Preserve the segment flags of the overall demux segment on the output
	  segments for each pad.

2014-12-09 02:43:00 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: use 64bit chunk_offset
	  https://bugzilla.gnome.org/show_bug.cgi?id=741279

2014-12-10 17:39:17 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix rounding errors in duration update
	  Make sure we store updated segment stop/duration with the same
	  granularity as the duration timescale.
	  And add more debug

2014-12-10 16:55:44 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Update duration when we get more information
	  When dealing with fragmented files, we will get more accurate duration
	  information via the mfra and moof atoms.
	  In order for playback to not stop at the initial duration (from the
	  moov atom), we need to check and update the various duration variables
	  when we find more information.
	  Fixes playback of fragmented files in pull mode

2014-12-10 15:08:40 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Remove variable assignments never read
	  As detected by clang/scan-build

2014-12-10 14:56:06 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Use GstClockTime for nanosecond-based time variables/fields
	  Avoids confusion with timescaled-based variables and bytes (offset)
	  variables.
	  And use GST_CLOCK_TIME_NONE where applicable

2014-12-03 14:47:05 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gstpushfilesrc.h:
	  pushfilesrc: Add TIME SEGMENT capability
	  Adds a new set of properties to make pushfilesrc output a TIME SEGMENT
	  (instead of the filesrc BYTE SEGMENT).
	  When time-segment is set to True the following will happen:
	  * Seeks are refused (data starts from the beginning of the file)
	  * The BYTE segment will be replaced by a TIME segment with the values
	  specified in the various properties
	  * The first outgoing buffer will have a timestamp set on it (by default
	  it has a value of GST_CLOCK_TIME_NONE)

2014-12-10 11:35:29 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Also only unref caps if they're not NULL

2014-12-10 11:34:42 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: gst_pad_get_allowed_caps() will return NULL if there is no peer

2014-12-09 16:38:38 +0100  Thibault Saunier <tsaunier@gnome.org>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vpXenc: CLOCK_TIME_NONE is not a valid min_latency value
	  We should just use 0 if we do not have the information

2014-12-03 17:26:56 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Use an empty iterator in iterate_internal_link when no links
	  And not a NULL Iterator, so it is consistent with the way it usually
	  works and avoid user to need a different code paths to handle that.

2014-12-09 14:01:50 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: set buffer interlace flags when field is V4L2_FIELD_INTERLACED
	  If v4l2_buffer.field is V4L2_FIELD_INTERLACED, we set corresponding
	  GstVideoBuffer flags depending on the video standard.
	  According to V4L2 specification, M/NTSC transmits the bottom field
	  first, all other standards the top field first.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737603

2014-12-08 21:26:18 +0100  Patrick Radizi <patrickr@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Fixes buffer leak when using SPS/PPS
	  Fixes a buffer leak that would occurr if the pipeline was shutdown
	  while a SPS/PPS header was being created.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741271

2014-12-09 04:43:29 +0100  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/effectv/gstaging.c:
	  agingtv: fix memcpy when no color aging requested.
	  video_size is the size in pixels, actual size of the memcpy
	  has to be stride * height.

2014-12-07 17:33:51 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Workaround libv4l2 RW emulation bug
	  When libv4l2 emulates RW mode on top of MMAP devices, the queues are
	  only initialized on first read. The problem is that poll() will fail
	  if called before the queues are initialized and streaming. Workaround
	  this by doing a zero size read when pool is started in that IO mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740633

2014-12-07 17:27:37 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Fix RW io mode
	  In RW, allocator can be null, max_buffers can be zero, and we need not
	  to wait while the queue is empty since there is no queue.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740633

2014-12-03 16:40:49 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Cleanup uneeded check and cases
	  There is nothing in between the break and the "done:" anymore, plus
	  USERPTR and DMABUF_IMPORT case is exactly the same.

2014-12-03 17:07:49 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2pool: Fix CREATE_BUFS support for capture
	  This patch fixes CREATE_BUFS support for capture devices. Initially we
	  would only try and allocate more buffers when the copy threshold
	  is reached. When the threshold was not set (needed) it would never
	  happen. Another problem is that on capture side, acquire returns
	  filled buffer, hence need to pool. We need to set a special flag to
	  force allocation to happen.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741134

2014-12-03 16:27:59 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix CREATE_BUF probing
	  Current for every memory type we where probing MMAP CREATE_BUFS ioct.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741134

2014-11-18 16:52:40 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: set framerate 0/1 when duration is not known
	  https://bugzilla.gnome.org/show_bug.cgi?id=740130

2014-12-04 17:25:55 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: More fixes for reverse playback
	  When seeking or finding the previous keyframe, do
	  comparisons against targets and segments using composition time
	  to correctly decide which sample times match.

2014-12-03 11:12:55 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Use an empty iterator in iterate_internal_link when no links
	  We used to setup an iterator with 1 GValue set with a NULL object
	  pointer which is not the normal way to do that. Instead we should make
	  sure that the first call to gst_iterator_next returns GST_ITERATOR_DONE.

2014-12-03 13:20:57 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Handle seeks past EOS as a seek to the end
	  Fix reverse playback of every frame by making seeks past/to EOS
	  find the last segment and start there.

2014-12-02 15:33:25 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpmpadepay.c:
	  rtpmpadepay: Relax caps to allow any clock-rate
	  Some Wowza setups seem to send an invalid non-90000 clock-rate.

2014-12-01 21:04:02 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't use GST_CLOCK_TIME_NONE in non GstClockTime variables
	  Use -1 instead as those are gint64/guint64 variables and not GstClockTime

2014-11-07 17:06:49 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2allocator.h:
	  v4l2allocator: fix gst_v4l2_allocator_stop prototype
	  gst_v4l2_allocator_stop returns a GstV4l2Return, not a gboolean.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739792

2014-11-07 16:41:52 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: unref pool when v4l2_allocator_new() fails
	  https://bugzilla.gnome.org/show_bug.cgi?id=739791

2014-11-30 17:52:47 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/v4l2_calls.h:
	  v4l2: Remove last include to linux/videodev2.h
	  We now use and update our internal copy so we no longer have to ifdef
	  the entire code for features and defines that where added over the
	  years.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740905

2014-08-24 13:38:08 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: implement seeking in fragmented mp4 files in pull mode based on the mfra table

2014-11-29 15:25:51 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: use track fragment decoding time (tfdt) in parse_trun() for interpolation
	  As fallback if we don't have any existing samples
	  as reference point yet.
	  Based on patch by David Corvoysier <david.corvoysier@orange.com>

2014-11-29 14:37:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: parse mfra random access box for fragmented mp4 files
	  If it's present, and we operate in pull mode.

2014-08-15 14:58:26 +0200  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: stop parsing headers for fragmented mp4s at the first moof
	  Currently during header parsing, we scan through the entire file
	  and skip every moof+mdat chunk for fragmented mp4s, which makes
	  start-up incredibly slow. Instead, just stop at the first moof
	  chunk when have a moov, and start exposing the streams, so we
	  can go and start handling the moofs for real.

2014-11-29 13:59:35 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/.gitignore:
	* tests/icles/Makefile.am:
	* tests/icles/gdkpixbufoverlay-test.c:
	  tests: add interactive gdkpixbufoverlay test
	  Just need to fix the coordinate system now so
	  that negative offsets are actually negative
	  and not flipped to position things from the
	  opposite border.

2014-11-29 13:53:03 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	  gdkpixbufoverlay: add "pixbuf" property
	  So we can set a GdkPixbuf directly instead of
	  reading it from an image file on the file system.

2014-11-29 13:23:50 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/gdk_pixbuf/pixbufscale.h:
	  gdkpixbuf: remove pixbufscale code that was never ported
	  Don't think we'll need this again.

2014-11-29 18:35:42 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtprtxreceive: Use offset when copying header
	  The header is not always at the start of the packet, so we need to compute
	  the offset first.

2014-11-28 13:12:46 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/taglib/gstapev2mux.cc:
	  apev2mux: write APE tags at end for wavpack files
	  http://www.wavpack.com/file_format.txt:
	  "Both the APEv2 tags and/or ID3v1 tags must come at the end of the
	  WavPack file, with the ID3v1 coming last if both are present."
	  WavPack files that contain APEv2 tags at the beginning of the files
	  are unplayable on players that use FFmpeg (like VLC) and most other
	  software (except Banshee). Players that use libwavpack directly can
	  play the files because it skips the tags, but does not recognize the
	  tag data at that location.
	  https://bugzilla.gnome.org/show_bug.cgi?id=711437

2014-11-28 10:41:55 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/.gitignore:
	* tests/icles/Makefile.am:
	* tests/icles/test-segment-seeks.c:
	  tests: add interactive test for gapless playback using SEGMENT seeks
	  Not working too well yet, there are glitches even with WAV or FLAC.
	  https://bugzilla.gnome.org/show_bug.cgi?id=692368

2014-11-26 10:33:09 +0300  Andrei Sarakeev <sarakusha@gmail.com>

	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstaspectratiocrop.h:
	  aspectratiocrop: Handle resolution changes properly
	  When an caps-event is received, we must immediately change the crop
	  to videocrop correctly changed caps-event dimension, otherwise the
	  videocrop will first use the previous value of the crop that when
	  resizing video to a smaller resolution may cause an error.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740671

2014-11-27 17:10:53 +0100  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From 7bb2bce to ef1ffdc

2014-11-27 11:20:36 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/test-accurate-seek.c:
	  test: use gst_util_uint64_scale_round() for timestamp to sample calculation

2014-11-27 11:16:35 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/icles/.gitignore:
	* tests/icles/Makefile.am:
	* tests/icles/test-accurate-seek.c:
	  tests: add interactive test for accurate seeking
	  For some audio formats.
	  https://bugzilla.gnome.org/show_bug.cgi?id=655276

2014-11-26 16:04:26 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: Check presence of mfhd in moof
	  The 'mfhd' atom is mandatory in 'moof'. We can later on check whether
	  the fragment number properly increases

2014-11-26 15:59:36 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux_dump.c:
	  isomp4: Fix mfro and tfra atom dumping
	  mfro was skipping the version/flags
	  tfra had wrong byte_reader return value checks

2014-11-26 15:58:26 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_types.c:
	  isomp4: Add mfhd atom dumping

2014-11-27 00:15:02 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Handle empty segments when seeking in reverse play.
	  Empty segments in an edit list have a media_start time of -1,
	  as they don't actually play any media. Allow for that when
	  aligning to the reference stream in reverse play.

2014-11-24 10:36:54 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	  Revert "v4l2allocator: Remove unused variable"
	  This reverts commit ad4480d53408a4d97ab531174ef37f258f3253c0.

2014-11-24 10:36:30 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  Revert "v4l2: move vb_queue probing from allocator to v4l2object"
	  This reverts commit ec6b8b84af719d828ddd91c724e715c0b4a556bc.

2014-11-24 10:33:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  Revert "v4l2object: allow to automatic selection of dmabuf"
	  This reverts commit e6c2ad5571e5dedb212287efe238eb450032cd4f.

2014-11-23 16:34:15 +0000  Tim-Philipp Müller <tim@centricular.com>

	* REQUIREMENTS:
	  REQUIREMENTS: update a little
	  People actually look at that it seems.

2014-11-23 16:22:12 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/icydemux/Makefile.am:
	  icydemux: does not need to link against zlib

2014-11-22 21:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* ext/speex/gstspeexdec.h:
	* ext/speex/gstspeexenc.h:
	  speex: remove support for ancient speex versions

2014-11-21 11:21:18 +0100  Branislav Katreniak <bkatreniak@nuvotechnologies.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: log connection events at info level
	  https://bugzilla.gnome.org/show_bug.cgi?id=739305

2014-10-20 13:00:37 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: ensure rtx_retry_period >= 0
	  https://bugzilla.gnome.org/show_bug.cgi?id=739344

2014-11-21 11:44:24 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Remove unused variable
	  this was introduced by commit ec6b8b
	  https://bugzilla.gnome.org/show_bug.cgi?id=699382

2014-11-16 12:34:17 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Handle corrupted buffer with empty payload
	  This allow skipping buffer flagged with ERROR that has no payload.
	  This is typical behaviour when a recovererable error occured during
	  capture in the driver, but that no valid data was ever written into that
	  buffer. This patch also translate V4L2_BUF_FLAG_ERROR into
	  GST_BUFFER_FLAG_CORRUPTED. Hence decoding error produce
	  by decoder due to missing frames will now be correctly marked. Finally,
	  this fixes a buffer leak when EOS is reached.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740040

2014-11-21 16:36:15 +0100  Benjamin Gaignard <benjamin.gaignard@linaro.org>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2object: allow to automatic selection of dmabuf
	  If the v4l2 queue support dmabuf select this buffer pool mode
	  and update the query with allocator.
	  This patch only concern exporting dmabuf and not importing dmabuf
	  fd from downstream element.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699382

2014-11-21 16:13:05 +0100  Benjamin Gaignard <benjamin.gaignard@linaro.org>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: move vb_queue probing from allocator to v4l2object
	  The goal is to make those information available in v4l2_object
	  to be able later to select the best allocation method for the pool
	  https://bugzilla.gnome.org/show_bug.cgi?id=699382

2014-11-20 22:42:59 +0530  Arun Raghavan <git@arunraghavan.net>

	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: Fix up new_jitterbuffer signal prototype

2014-11-20 20:19:25 +0530  Arun Raghavan <git@arunraghavan.net>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Document how to control per-SSRC retransmission

2014-11-20 20:18:45 +0530  Arun Raghavan <git@arunraghavan.net>

	* docs/design/design-rtpretransmission.txt:
	  doc: Trivial spelling and consistency update

2014-11-20 13:14:14 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: put 0-byte at the end of events
	  Put a 0-byte at the end of the event string. Does not break ABI because
	  old depayloaders will skip the 0 byte (which is included in the length).
	  Expect a 0-byte at the end of the event string or a ; for old
	  payloaders.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=737591

2014-11-20 12:40:28 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: avoid buffer overread.
	  Check that a caps event string is 0 terminated and the event string is
	  terminated with a ; to avoid buffer overreads.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737591

2014-11-20 10:45:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: don't limit max video resolution to 4096x4096
	  MAX isn't entirely correct as upper limit either,
	  it should really be MAXUINT32, but it's unlikely
	  to be a problem in the near future.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740407

2014-11-19 15:06:00 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix leak for mikey base64 decoded key-mgmt
	  https://bugzilla.gnome.org/show_bug.cgi?id=740392

2014-11-20 09:01:38 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: fix unhandled format in passthrough
	  In passthrough we can handle all formats.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740387

2014-11-19 16:12:38 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Restrict resyncing to TS regressions
	  The behavior of resyncing video and audio indepen-
	  dently can cause A/V desyncs. Lets restrict resyncs
	  to jumps backward for now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736397

2014-11-17 23:16:03 +1100  Matthew Waters <matthew@centricular.com>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: fix up QoS handling for live sources
	  Only attempt adaptive drop when we are not live
	  https://bugzilla.gnome.org/show_bug.cgi?id=739996

2014-11-10 22:34:39 +0100  Henning Heinold <henning@itconsulting-heinold.de>

	* tests/examples/rtp/client-PCMA.py:
	* tests/examples/rtp/server-alsasrc-PCMA.py:
	  examples: port python rtp PCMA client/server tests to 1.0
	  https://bugzilla.gnome.org/show_bug.cgi?id=739930

2014-06-04 12:11:10 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: set the channel positions using the appropriate API
	  This avoids _set_format setting the unpositioned flag when passed
	  NULL as channel positions, as it would not be cleared when setting
	  actual channel positions later.

2014-11-01 22:39:41 +0100  Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vpx: mark arnr-type properties as deprecated and set them to no-op
	  ARNR type control in libvpx has been deprecated so this commit mark the
	  vp8enc and vp9enc associated properties as deprecated and change their
	  behavior to just display a warning message.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739476

2014-11-10 13:16:01 +0530  Arun Raghavan <git@arunraghavan.net>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpmanager: Trivial typo fix

2014-11-09 11:04:33 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Use G_DEFINE_TYPE() to register the pad instead of manually registering it

2014-11-06 15:37:28 +0100  Göran Jönsson <goranjn@axis.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: make GstMatroskamuxPad get_type() function thread-safe
	  https://bugzilla.gnome.org/show_bug.cgi?id=739722

2014-11-07 16:11:24 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: fix error message if allocator is already active
	  https://bugzilla.gnome.org/show_bug.cgi?id=739789

2014-11-06 21:21:40 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Improve buffer validation
	  Improve buffer validation by making sure each memory are the right
	  one and that each memory is writable. This fixes tearing issues in
	  case downstream uses gst_buffer_make_writable() or other type
	  of GstBuffer copy where memory are only reffed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739754

2014-11-06 21:38:43 +0100  Josep Torra <n770galaxy@gmail.com>

	* gst/rtsp/Makefile.am:
	  rtsp: fix build in gst-uninstalled setup

2014-10-29 18:44:43 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: Handle seqnums
	  https://bugzilla.gnome.org/show_bug.cgi?id=739366

2014-11-04 08:18:41 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngdec.h:
	  pngdec: change parse logic
	  Right now in parse logic the signature is checked every time the parse function
	  is called, and the whole data is the scanned each and every time, even though the
	  data is scanned in the previous instance. Changing the logic such that, we skip
	  the bytes which are already scanned in the previous instances of parse. This
	  helps in avoiding multiple scan of already scanned data/signature.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737708

2014-11-03 15:26:06 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/videomixer/videomixer2.c:
	  videomixer2: reverse order of params for converter

2014-11-03 11:44:28 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: fix typo in flags
	  https://bugzilla.gnome.org/show_bug.cgi?id=739549

2014-11-02 23:33:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: fix a couple of minor leaks

2014-11-02 19:42:03 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	  goom2k1: post QoS messages when dropping frames due to QoS

2014-11-02 19:29:52 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/goom/gstgoom.c:
	* gst/goom/gstgoom.h:
	  goom: post QoS messages when dropping frames due to QoS

2014-11-02 19:02:35 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: tweak writing app tag string a little

2014-11-02 16:51:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	* gst/isomp4/gstqtmux.c:
	* gst/level/gstlevel.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	  Sprinkle some G_PARAM_DEPRECATED and #ifndef GST_REMOVE_DEPRECATED

2014-11-02 16:58:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/level.c:
	  tests: don't use deprecated property in level unit test

2014-11-02 13:06:33 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: implement get/set for new rtx-min-retry-timeout property
	  Properties are so much more useful if you can actually set
	  and get their values.

2014-10-30 17:41:19 +0000  Simon Farnsworth <simon.farnsworth@onelan.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Clean up interlace support
	  Rather than try and guess interlace support as part of checking supported
	  sizes, look for interlace support specifically in its own function.
	  As a cleanup, use V4L2_FIELD_ANY when probing sizes, which should result in
	  the driver doing the right thing.
	  With my capture setup, this gets me the following sample caps:
	  For 1080i resolution:
	  video/x-raw, format=(string)YUY2, width=(int)1920, height=(int)1080, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)interleaved, framerate=(fraction){ 25/1, 30/1 }
	  For 720p resolution:
	  video/x-raw, format=(string)YUY2, width=(int)1280, height=(int)720, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, framerate=(fraction){ 50/1, 60/1 }
	  For 576i/p resolution (both possible at the point of query):
	  video/x-raw, format=(string)YUY2, width=(int)720, height=(int)576, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string){ progressive, interleaved }, framerate=(fraction){ 25/1, 50/1 }
	  This, in turn, makes 576i work correctly; with the old code,
	  the caps would be interlace-mode=progressive for interlaced video.
	  https://bugzilla.gnome.org/show_bug.cgi?id=726194

2014-11-01 12:18:02 +0100  Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>

	* ext/vpx/gstvp8utils.h:
	  vpx: remove compatibility defines
	  We are guaranteed to have VPX_IMG_FMT_I420, VPX_PLANE_Y,
	  VPX_PLANE_U and VPX_PLANE_V as we require libvpx > 1.1.0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739476

2014-11-01 15:33:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* ext/wavpack/gstwavpackcommon.c:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	  wavpack: remove support for ancient API version

2014-11-01 10:14:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8: Use VP8 encoding name
	  Both Firefox and Chrome uses VP8 as the encoding in their SDP.
	  Adding this now defacto standard name removes the need for special
	  case in SDP parsing code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737810

2014-11-01 11:59:26 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpmp2tpay.c:
	  rtpmp2tpay: fix up template caps so we can output the default pt 33
	  Add fixed payload type for mp2t to template caps as well, so
	  our output caps match the advertised default pt. Fixes a
	  regression from 1.2.
	  There's still something wrong with caps negotiation though,
	  rtpmp2tpay payload=96 ! fakesink will not output caps with
	  payload=96.

2014-10-30 15:37:36 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: mikey related memory leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=739430

2014-06-10 10:04:07 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	* ext/speex/gstspeexenc.h:
	  speexenc: update output segment stop time to match clipped samples
	  This will let oggmux generate a granpos on the last page that properly
	  represents the clipped samples at the end of the stream.

2014-06-10 10:59:13 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flacenc: update output segment stop time to match clipped samples
	  This will let oggmux generate a granpos on the last page that properly
	  represents the clipped samples at the end of the stream.

2014-10-07 15:29:33 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: cleanly handle streamon failure for output device
	  On streamon failure, the queued buffer is not released from the
	  bufferpool class point of view because it is queued to the driver and
	  the flush logic is not performed since we are not in streaming state.
	  It causes the v4l2 bufferpool to always return that stop method failed
	  and to leak v4l2 objects and buffers.
	  This commit solve this by performing the flush logic in error case, ie
	  flushing the allocator and restoring queued buffer state to non-queued.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738102

2014-10-08 10:31:21 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: implement dispose method
	  Unref objects in dispose method rather than in finalize in order to
	  prevent circular reference.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738102

2014-10-08 10:35:14 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: check that allocator is non null when stopping pool
	  Otherwise, we could dereference NULL allocator when the stop method is
	  called by the GstBufferPool's finalize method.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738102

2014-10-09 12:15:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Implement unlock/unlock_stop
	  This will prevent deadlocks, but will also properly flush the pool and allocator
	  when going to READY state. It should also fix issues reported on mailing list
	  when seeking is performed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738152

2014-10-28 21:32:06 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/pulse/pulsedeviceprovider.h:
	* sys/v4l2/gstv4l2deviceprovider.h:
	* sys/v4l2/gstv4l2tuner.h:
	  pulse, v4l2: add missing G_END_DECLS in some places

2014-10-27 17:57:20 +0100  Sebastian Dröge <sebastian@centricular.com>

	* common:
	  Automatic update of common submodule
	  From 84d06cd to 7bb2bce

2014-10-27 11:08:20 +0100  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/aacparse.c:
	  aacparse: Fix unit test now that we always have profile/level in the caps

2014-10-26 14:55:49 +0000  Tim-Philipp Müller <tim@centricular.com>

	* Makefile.am:
	  Parallelise 'make check-valgrind'
	  Some of the RTP unit tests are very flaky and will
	  fail more often with the CPU maxed out fully. Those
	  tests need to be fixed in any case though, they also
	  fail on slower machines and also occasionally with
	  normal 'make check'.

2014-10-26 11:47:25 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Always set profile/level on the caps
	  We have the information already, so why not use it?

2014-10-25 12:36:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix crash on some 32-bit systems
	  Make sure to pass right number of bits to gst_structure_new()
	  which is a vararg function.
	  Fixes elements/rtpaux unit test on ppc32.

2014-10-25 00:56:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rgvolume.c:
	  tests: fix rgvolume test on big-endian systems

2014-10-25 00:53:39 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/mulawdec.c:
	* tests/check/elements/mulawenc.c:
	  tests: fix mulawdec/mulawenc test for big endian systems

2014-10-24 23:48:30 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/interleave/interleave.c:
	  interleave: intersect result with filter caps in caps query
	  Fixes crash in audiotestsrc because of an unsupported format
	  getting negotiated on big-endian systems with
	  audiotestsrc ! interleave ! audioconvert ! wavenc

2014-10-23 15:46:13 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/pulse/pulsedeviceprovider.c:
	* ext/pulse/pulsedeviceprovider.h:
	  pulse: remove some unused typedefs

2014-10-22 15:28:44 +0200  Ananda <ananda@latelier23.com>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	  speex: Fix segfault when resetting the codecs multiple times
	  https://bugzilla.gnome.org/show_bug.cgi?id=738793

2014-10-22 22:50:54 +0530  Arun Raghavan <arun@accosted.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: Temporarily disable stream status posting
	  We need a mechanism in PulseAudio to allow running code outside the
	  mainloop lock. Then we'd be able to post to the bus (taking the
	  GST_OBJECT_LOCK), without worrying about locking order with the mainloop
	  lock, which is the current cause of deadlocks while trying to post the
	  stream status messages.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736071

2014-10-22 15:04:24 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: limit the retry frequency
	  When the RTT and jitter are very low (such as on a local network), the
	  calculated retransmission timeout is very small. Set some sensible lower
	  boundary to the timeout by adding a new property. We use the packet
	  spacing as a lower boundary by default.

2014-10-22 13:40:58 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  gstrtpjitterbuffer: add "rtx-min-delay" property
	  This property is useful to set a min time to wait before sending a
	  retransmission event.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=735378

2014-10-22 13:29:48 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Refactor code
	  Refactor some code dealing with calculating various timeouts.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=735378

2014-10-10 19:50:06 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: fix Early Feedback Transmission
	  In early retransmission we are allowed to schedule 1 regular RTCP packet
	  at an earlier time. When we do that, we need to set allow_early to FALSE
	  and ignore/drop (or merge) all future requests for early transmission.
	  We now first check if we can schedule an early RTCP and if we can,
	  actually prepare the data for the next RTCP interval.
	  After we send the next regular RTCP after the early RTCP, we set
	  allow_early to TRUE again to allow more early requests.
	  Remove the condition for the immediate feedback for now.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319

2014-10-21 13:01:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* common:
	  Automatic update of common submodule
	  From a8c8939 to 84d06cd

2014-10-21 13:10:24 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: make debug line less confusing

2014-10-21 12:58:13 +0200  Stefan Sauer <ensonic@users.sf.net>

	* README:
	* common:
	  Automatic update of common submodule
	  From 36388a1 to a8c8939

2014-07-02 17:50:35 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: rework resync handling
	  Add a need-resync state, this is when we need to try to lock on to a
	  time/RTPtime pair.
	  Always check the RTP timestamps and if they go backwards, mark ourselves
	  as need-resync.
	  Only resync when need-resync is TRUE and we have a valid time. Otherwise
	  we keep the old values. This avoids locking on to an invalid time and
	  causing us to timestamp everything with -1.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417

2014-10-03 17:28:06 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: set full stream caps on internal src TCP pads
	  Set the complete stream caps on the TCP internal src pads. Otherwise,
	  ptdemux will not properly detect the caps change.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737868

2014-10-17 22:23:27 +0200  Sjoerd Simons <sjoerd@luon.net>

	* gst/rtpmanager/gstrtpmux.c:
	* tests/check/elements/rtpmux.c:
	  rtpmux: Don't set PROXY_CAPS flag on the src pad
	  rtpmux behaves like a funnel in that it forwards whatever upstream is
	  sending buffers. So setting proxy caps doesn't make sense as the
	  upstream don't have to have compatible caps, thus resulting in an empty
	  caps set as a result of a caps query. Instead set fixed caps just
	  as funnel does.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738722

2014-10-20 11:57:38 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* gst/videobox/gstvideobox.c:
	  videobox: critical error when element properties set as max/min
	  left, right, top, bottom can be set from range of -2147483648 to 2147483647
	  when i launch the videobox element with that values, it gives a critical error
	  (gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed
	  This happens because min cannot be equal to max.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738838

2014-10-15 17:45:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	  Revert "rtp: add h265 RTP payloader + depayloader"
	  This reverts commit d06ba9051f904a7eb482c07a97a1827169158663.
	  This breaks the build, as it depends on parser API in -bad.

2014-10-15 17:34:50 +0200  Jurgen Slowack <jurgen.slowack@barco.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtph265depay.c:
	* gst/rtp/gstrtph265depay.h:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtph265pay.h:
	  rtp: add h265 RTP payloader + depayloader

2014-10-05 21:24:27 +0200  Peter G. Baum <peter@dr-baum.net>

	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	  wavenc: Support RF64 format
	  https://bugzilla.gnome.org/show_bug.cgi?id=725145

2014-10-11 11:18:42 +1100  David Sansome <me@davidsansome.com>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Don't call iirequalizer's transform_ip in passthrough mode
	  It tries to map the read-only buffer with GST_MAP_READWRITE and crashes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737886

2014-10-10 18:30:07 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsource: Rename seqnum-base to seqnum-offset in caps
	  This was modified back in 1.0 in GstRtpBasePayload

2014-10-10 18:11:19 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/dtmf/gstrtpdtmfsrc.c:
	* tests/check/elements/dtmf.c:
	  rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
	  These were renamed in GstRTPBasePayload in 1.0

2014-10-10 17:30:24 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* tests/check/elements/rtpmux.c:
	  rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
	  These were renamed in GstRTPBasePayload in 1.0

2014-10-06 14:23:22 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom2k1/filters.c:
	  goom2k1: removing block of code that does nothing
	  The loop in zoomFilterSetResolution is meant to change the values in the
	  zf->firedec[] array. Each iteration writes the value of decc onto the arrya,
	  but no conditions that change the value of decc are ever met and the array is
	  filled with zero for each element. Which is the initial state of the
	  array before the loop begins.
	  The loop does nothing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=728353

2014-10-04 17:17:13 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: don't log all clock_rate changes as warnings.
	  We never initialize clock_rate explicitly, therefore it is 0 by default. The
	  parameter is a uint32 and the only caller ensure that it is >0, therefore it
	  won't become -1 ever.

2014-10-02 14:26:08 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Fix lifetime of stream headers and queued buffers
	  Stream headers are updated whenever ::set_caps is called, so we can't assume
	  they'll be valid before the message body is written out. We *can* assume that
	  for queued buffers, but SOUP_MEMORY_STATIC is still wrong for those.
	  Also, add some debug logging for stream header interactions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737771

2014-10-02 03:26:22 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix memory leak when prepending ADTS headers
	  https://bugzilla.gnome.org/show_bug.cgi?id=737761

2014-09-23 10:48:09 +0200  Antonio Ospite <ao2@ao2.it>

	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	  interleave: interleave samples following the Default Channel Ordering
	  In order to have a full mapping between channel positions in the audio
	  stream and loudspeaker positions, the channel-mask alone is not enough:
	  the channels must be interleaved following some Default Channel Ordering
	  as mentioned in the WAVEFORMATEXTENSIBLE[1] specification.
	  As a Default Channel Ordering use the one implied by
	  GstAudioChannelPosition which follows the ordering defined in SMPTE
	  2036-2-2008[2].
	  NOTE that the relative order in the Top Layer is not exactly the same as
	  the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users
	  using so may channels are already aware of such discrepancies.
	  [1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx
	  [2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf
	  Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127

2014-10-02 10:10:11 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Send CAPS event after the pad was activated
	  Otherwise the CAPS event will be dropped and we never configure any caps at
	  all, leading to weird behaviour in many situations. Especially header
	  rewriting is not going to work if a capsfilter is after wavenc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737735

2014-10-01 23:12:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Add some more useful debug logging

2014-10-01 23:05:03 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Free queued buffers in ::reset
	  ::render sets a new callback for writing out new buffers only if there aren't
	  already buffers queued for writing with a previously-scheduled callback.
	  However, if the previously-scheduled callback is interrupted by a state change
	  (either manually or due to an error) and there are still buffers in the queue,
	  restarting the pipeline will result in buffers being queued forever, and no
	  callbacks will ever be scheduled, and no buffers will be written out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737739

2014-10-01 17:29:29 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Actually use the correct GstVideoInfo for conversion

2014-10-01 17:24:59 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Revert the last commit and handle resolutions differences properly
	  This is about converting the format, not about converting any widths and
	  heights. Subclasses are expected to handler different resolutions themselves,
	  like the videomixers already do properly.

2014-10-01 17:12:59 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: GstVideoConverter currently can't rescale and will assert
	  Leads to ugly assertions instead of properly erroring out:
	  CRITICAL **: gst_video_converter_new: assertion 'in_info->width == out_info->width' failed

2014-09-30 11:35:12 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vp8enc/vp9enc: Protect the encoder with a mutex in all situations

2014-09-30 11:31:43 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp9enc.c:
	  vp9enc: Allow caps renegotiation
	  https://bugzilla.gnome.org/show_bug.cgi?id=726329

2014-09-30 11:28:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: finish() and drain() should return a GstFlowReturn

2014-03-14 12:59:02 +0100  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Allow caps renegotiation
	  https://bugzilla.gnome.org/show_bug.cgi?id=726329

2014-09-29 11:49:45 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: set colorspace for output devices
	  When the v4l2 device is an output device, the application shall set the
	  colorspace. So map GStreamer colorimetry info to V4L2 colorspace and set
	  on set_format. In case we have no colorimetry information, we try to
	  guess it according to pixel format and video size.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737579

2014-09-29 22:48:16 +0530  Arun Raghavan <arun@accosted.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulse: Add some documentation about threading and synchronisation
	  This gives a quick introduction to how the pulsesink/pulsesrc code
	  interacts with the pa_threaded_mainloop that we start up to communicate
	  with the server.

2014-09-29 20:18:08 +0530  Arun Raghavan <arun@accosted.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: Make emitting stream status messages synchronous
	  The stream status messages are emitted in the PA mainloop thread, which
	  means the mainloop lock is taken, followed by the Gst object lock (by
	  gst_element_post_message()). In all other locations, the order of
	  locking is reversed (this is unavoidable in a bunch of cases where the
	  object lock is taken by GstBaseSink or GstAudioBaseSink, and then we get
	  control to take the mainloop lock).
	  The only way to guarantee that the defer callback for stream status
	  messages doesn't deadlock is to either stop posting those messages, or
	  make sure that the message emission is completed before we proceed to
	  any point that might take the object lock before the mainloop lock
	  (which is what we do after this patch).
	  https://bugzilla.gnome.org/show_bug.cgi?id=736071

2014-09-16 12:12:49 +0200  Antonio Ospite <ao2@ao2.it>

	* gst/wavenc/gstwavenc.c:
	  wavenc: print channel masks in hexadecimal

2014-09-27 16:01:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2deviceprovider.h:
	  v4l2: remove redundant struct declaration

2014-09-26 13:46:16 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix compiler warnings
	  gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
	  'GstRTSPResult' [-Werror,-Wenum-conversion]
	  res = gst_sdp_message_new (&sdp);
	  ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
	  gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
	  'GstRTSPResult' [-Werror,-Wenum-conversion]
	  res = gst_sdp_message_parse_uri (uri, sdp);
	  ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

2014-09-25 15:01:14 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: make demuxer reusable
	  Remove pads from flow combiner and reset last
	  flow return to FLOW_OK by resetting the flow combiner.
	  This prevents FLOW_FLUSHING when trying to re-use the
	  demuxer after setting it back to NULL/READY state.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737359

2014-09-24 16:46:36 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/gstcms.c:
	* gst/videomixer/gstcms.h:
	* gst/videomixer/videoconvert.c:
	* gst/videomixer/videoconvert.h:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2pad.h:
	* gst/videomixer/videomixerorc-dist.c:
	* gst/videomixer/videomixerorc-dist.h:
	* gst/videomixer/videomixerorc.orc:
	  videomixer: use video library code instead of copy

2014-09-18 16:39:19 +0530  Sanjay NM <sanjay.nm@samsung.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparsers: Added index check before using the index
	  https://bugzilla.gnome.org/show_bug.cgi?id=736878

2014-09-23 23:33:37 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Do not infer DTS on buffers from sparse streams.
	  DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later.
	  This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap)
	  https://bugzilla.gnome.org/show_bug.cgi?id=737095

2014-09-18 17:08:37 +0530  Sanjay NM <sanjay.nm@samsung.com>

	* gst/goom/ifs.c:
	  goom: Clarified precedence between % and ?
	  https://bugzilla.gnome.org/show_bug.cgi?id=736887

2014-09-18 17:59:31 +0530  Sanjay NM <sanjay.nm@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: clarify expression so operator precedence is clear
	  https://bugzilla.gnome.org/show_bug.cgi?id=736903

2014-09-18 16:04:03 +0530  Sanjay NM <sanjay.nm@samsung.com>

	* ext/libpng/gstpngdec.c:
	* gst/alpha/gstalpha.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/gstscaletempo.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-mux.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/rtpsession.c:
	  Miscellaneous minor cleanups
	  Fix redundant variables and assignments,
	  and unreachable breaks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736875
	  https://bugzilla.gnome.org/show_bug.cgi?id=736876
	  https://bugzilla.gnome.org/show_bug.cgi?id=736879
	  https://bugzilla.gnome.org/show_bug.cgi?id=736880
	  https://bugzilla.gnome.org/show_bug.cgi?id=736881
	  https://bugzilla.gnome.org/show_bug.cgi?id=736888
	  https://bugzilla.gnome.org/show_bug.cgi?id=736890
	  https://bugzilla.gnome.org/show_bug.cgi?id=736892
	  https://bugzilla.gnome.org/show_bug.cgi?id=736893
	  https://bugzilla.gnome.org/show_bug.cgi?id=736894

2014-09-24 00:12:14 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/videobox/gstvideobox.c:
	  videobox: remove duplicate assignments
	  https://bugzilla.gnome.org/show_bug.cgi?id=736897

2014-09-23 22:55:48 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Only calculate with durations != -1

2014-09-23 19:08:48 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: collect pad for sparse stream should be created with lock set to false
	  Avoids waiting for buffers from sparse streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=737095

2014-09-23 19:07:25 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: fix subtitle buffer duration and strip null termination
	  Strip the \0 off the subtitle as we already know the size and also remember
	  to set the duration as buffer copying doesn't do it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737095

2014-09-23 19:06:18 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/atoms.c:
	  qtmux: move subtitle layer above video and set alternate group
	  layer -1 is above video, that is 0
	  And having all subtitles in alternate group 2 means that only one
	  should be selected at a time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=737095

2014-09-23 09:47:31 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/souphttpsrc.c:
	  check/soup: Temporarily disable G_ENABLE_DIAGNOSTIC
	  The SOUP_SERVER_PORT property has been deprecated in recent libsoup
	  versions.

2014-09-23 09:43:05 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/souphttpsrc.c:
	  check/soup: Define minimum version required
	  To avoid deprecation warnings

2014-09-19 19:14:28 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Handle mp4a without ESDS atom
	  https://bugzilla.gnome.org/show_bug.cgi?id=736986

2014-09-22 16:15:27 +0200  Linus Svensson <linussn@axis.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Fix build problem without XFIXES

2014-09-19 14:34:13 +0530  Sanjay NM <sanjay.nm@samsung.com>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  dtmf: Removed unused structure members
	  https://bugzilla.gnome.org/show_bug.cgi?id=736883

2014-09-11 13:48:44 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  isomp4: fix wrong DAR calculation for PAR <= 1
	  CID #1226452
	  https://bugzilla.gnome.org/show_bug.cgi?id=736396

2014-09-18 16:59:52 +0530  Sanjay NM <sanjay.nm@samsung.com>

	* gst/flv/gstflvdemux.c:
	  flv: Removed unreachable break statements
	  https://bugzilla.gnome.org/show_bug.cgi?id=736884

2014-09-17 16:37:11 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: do not leak encsink pad in error case
	  https://bugzilla.gnome.org/show_bug.cgi?id=736807

2014-09-17 16:23:21 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: do not leak new stream event
	  https://bugzilla.gnome.org/show_bug.cgi?id=736805

2014-09-15 09:08:18 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  y4menc: port y4menc to use GstVideoEncoder base class
	  https://bugzilla.gnome.org/show_bug.cgi?id=735085

2014-09-17 13:55:18 +0300  Sebastian Dröge <sebastian@centricular.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: OSStatus is not a fourcc, so don't print it as one...

2014-09-16 14:26:08 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: do not leak uid after parsing TOC event
	  https://bugzilla.gnome.org/show_bug.cgi?id=736739

2014-09-16 22:47:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtpvrawdepay: Declare some more required caps fields in the sink template caps
	  Now only missing are width and height, which are expressed as strings
	  for RTP... so we can't put them into the template caps.

2014-09-16 16:46:07 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.h:
	  gdkpixbufdec: modify wrong packetized mode logic
	  packetized mode is being set when framerate is being set
	  which is not correct. Changing the same by checking the
	  input segement format. If input segment is in TIME it is
	  Packetized, and if it is in BYTES it is not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736252

2014-09-16 11:26:22 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Remove unused variable and use correct decoder variable name

2014-09-16 11:25:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/libpng/gstpngdec.c:
	  pngdec: Remove unused variable

2014-09-16 13:24:15 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* ext/jpeg/gstjpegdec.c:
	  jpeggdec: modify wrong packetized mode logic
	  packetized mode is being set when framerate is being set
	  which is not correct. Changing the same by checking the
	  input segement format. If input segment is in TIME it is
	  Packetized, and if it is in BYTES it is not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736252

2014-09-16 13:23:16 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* ext/libpng/gstpngdec.c:
	  pngdec: modify wrong packetized mode logic
	  packetized mode is being set when framerate is being set
	  which is not correct. Changing the same by checking the
	  input segement format. If input segment is in TIME it is
	  Packetized, and if it is in BYTES it is not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736252

2014-09-15 14:39:41 +0200  Antonio Ospite <ao2@ao2.it>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	* sys/ximage/ximageutil.c:
	* sys/ximage/ximageutil.h:
	  ximagesrc: Remove unused screen-num property
	  The screen number can be still specified as part of the display-name
	  property (e.g. for screen 1 of display 0 use display-name=":0.1").
	  https://bugzilla.gnome.org/show_bug.cgi?id=736122

2014-09-04 16:10:51 +0200  Antonio Ospite <ao2@ao2.it>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Draw the cursor only when it is active in the capturing region
	  Use XQueryPointer to check that the pointer is actually active inside
	  the capturing region.
	  This prevents drawing the cursor when the pointer is partially outside
	  of the captured region but not active inside the region; in particular
	  this avoids drawing the "window resize" cursor shapes to the captured
	  image when the mouse pointer crosses a window border.
	  NOTE that this is not only an optimization, this also happen to fix
	  a serious problem in multi-screen setups.
	  Because XFixes gives no information of what screen the pointer is on,
	  ximagesrc was always drawing the cursor on the captured screen even if
	  the mouse pointer was on another screen.
	  For example, when capturing from screen 1 (i.e. display-name=":0.1") the
	  cursor was drawn in the captured image even when the mouse pointer was
	  actually on screen 0, which is wrong and visually confusing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690646

2014-09-05 11:33:31 +0200  Antonio Ospite <ao2@ao2.it>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Fix drawing the cursor when it is outside the capturing region
	  When the cursor is partially or totally out of the capturing region on
	  the top side or on the left side, it gets drawn fully inside of the
	  region with its coordinates rounded up to the left or to the top border.
	  This is immediately noticeable when using the xid property to capture
	  a specific window.
	  To fix the issue, allow negative cx and cx coordinates when checking the
	  boundaries before drawing the cursor.
	  NOTE that the boundaries checking calculations still allows the cursor
	  to be drawn when it is only partially outside of the capturing region,
	  but this makes sense and gives a more pleasing visual behaviour.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690646

2014-09-05 00:15:30 +0200  Antonio Ospite <ao2@ao2.it>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	  ximagesrc: Fix the destination coordinates of the cursor
	  XFixes provides the cursor coordinates relative to the root window, this
	  is not taken into account when using the xid property to capture
	  a specific window, the result is that the cursor gets drawn at the wrong
	  position.
	  In order to fix this consider the window location when calculating the
	  cursor position in the destination image.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690646

2014-09-15 14:51:24 +0200  Peter Korsgaard <peter@korsgaard.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: O_CLOEXEC needs _GNU_SOURCE
	  Similar to 94f3d6fc / bz 709423
	  On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only
	  defined when _GNU_SOURCE is specified, so do so.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736670

2014-09-15 18:11:37 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/debugutils/gstcapssetter.c:
	  capssetter: update to 1.0 transform_caps sematics
	  In 1.0, we pass the complete caps to transform_caps to allow for better
	  optimizations. Make this function actually work on non-simple caps
	  instead of just ignoring the configured filter caps.

2014-09-08 14:06:00 +0200  Peter G. Baum <peter@dr-baum.net>

	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	  wavenc: use WAVE_FORMAT_EXTENSIBLE for more than 2 channels
	  https://bugzilla.gnome.org/show_bug.cgi?id=733444

2014-09-12 15:06:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix parsing of adtl chunks
	  We have to skip 12 bytes of data for the chunk, and the data size
	  passed to the sub-chunk parsing functions should have 4 bytes less
	  than the data size.
	  Also when parsing the sub-chunks, check if we actually have enough
	  data to read instead of just crashing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736266

2014-09-12 10:55:23 +0530  Sanjay NM <sanjay.nm@samsung.com>

	* gst/udp/gstudpsrc.c:
	  udp: include string.h for memcmp and memset
	  https://bugzilla.gnome.org//show_bug.cgi?id=736528

2014-09-12 13:36:18 +0530  Anuj Jaiswal <anuj.jaiswal@samsung.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: don't bitwise OR the same flag twice
	  https://bugzilla.gnome.org//show_bug.cgi?id=736543

2014-09-12 10:35:36 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: handle real audio 28_8
	  Fixes duplicate check for 14_4.
	  https://bugzilla.gnome.org//show_bug.cgi?id=736543

2014-09-11 14:46:09 +0530  Anuj Jaiswal <anuj.jaiswal@samsung.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: don't OR the same flag twice
	  https://bugzilla.gnome.org/show_bug.cgi?id=736462

2014-09-11 12:52:11 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: If the server reports "Accept-Ranges: none" don't try range requests

2014-09-10 09:50:45 +0200  Ognyan Tonchev <ognyan@axis.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Unref pool after usage
	  https://bugzilla.gnome.org/show_bug.cgi?id=736384

2014-09-09 19:03:50 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Don't rank it for now
	  This will prevent the converter to be picked automatically in case
	  someone implement dynamic converter selection support. I'd like this
	  to be ranked only for known device, as it's hard to be sure a device is
	  a converter suited for general purpose. Re-negotiation is also needed
	  before we can rank it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733607

2014-09-05 08:29:20 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2: Detect bad drivers timestamps
	  Even though the UVC driver do a great deal of effort to prevent bad
	  timestamp to be sent to userspace, there still exist UVC hardware that
	  are so buggy that the timestamp endup nearly random. This code detect
	  and ignore timestamp from these drivers, making these camera usable.
	  This has been tested on both invalid and valid cameras, making sure it
	  does not trigger for valid cameras.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732910

2014-08-29 17:09:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Workaround driver that don't support REQBUFS(0)
	  There is still around 18 drivers not yet ported to videobuf2. These driver
	  don't support freeing buffetrs through REQBUFS(0) hence for these the
	  memory type probing fails. In order to gain back our previous behaviour in
	  presence of these, we implement a workaround that assuming MMAP is
	  supported. Note that an allocator is only created for device with
	  STREAMING support in the device capabilities. In such case one of MMAP,
	  USERPTR and DMABUF is required. Though DMABUF came afterward, so is
	  not an option and in practice none of these drivers will only do USERPTR.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735660
	  Also-by: Hans de Goede <hdegoede@redhat.com>

2014-09-04 15:11:40 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Merge min_buffers_for* variable into one
	  Reuse the same min_buffers variable for both capture and output, this
	  reduce the length of lines and make the code more readable.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736072

2014-09-04 18:35:46 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: set min_latency for output device according to required minimum number of buffers
	  Since we can get the minimum number of buffers needed by an output
	  device to work, use it to set min_latency which will determine how many
	  buffers are queued.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736072

2014-09-09 16:10:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/udpsrc.c:
	  tests: udpsrc: add check to make sure multiple memory chunks are used

2014-09-09 15:55:18 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/udpsrc.c:
	  tests: udpsrc: wait for buffers with GCond instead of sleeping
	  Avoids half-second sleep for no reason.

2014-09-09 15:31:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/udpsrc.c:
	  tests: udpsrc: split out socket setup

2014-09-09 13:46:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: more efficient memory handling
	  Drop use of g_socket_get_available_bytes() which is
	  not useful on all systems (where it returns the size
	  of the entire buffer not that of the next pending
	  packet), and is yet another syscall and apparently
	  very inefficient on Windows in the UDP case.
	  Instead, when reading UDP packets, use the more featureful
	  g_socket_receive_message() call that allows to read into
	  scattered memory, and allocate one memory chunk which is
	  likely to be large enough for a packet, while also providing
	  a larger allocated memory chunk just in case the packet
	  is larger than expected. If the received data fits into the
	  first chunk, we'll just add that to the buffer we return
	  and re-use the fallback buffer for next time, otherwise we
	  add both chunks to the buffer.
	  This reduces memory waste more reliably on systems where
	  get_available_bytes() doesn't work properly.
	  In a multimedia streaming scenario, incoming UDP packets
	  are almost never fragmented and thus almost always smaller
	  than the MTU size, which is also why we don't try to do
	  something smarter with more fallback memory chunks of
	  different sizes. The fallback scenario is just for when
	  someone built a broken sender pipeline (not using a
	  payloader or somesuch)
	  https://bugzilla.gnome.org/show_bug.cgi?id=610364

2014-09-09 12:15:43 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: rework memory allocation bits and ensure we always have two chunks of memories to read into
	  First chunk is the likely/expected buffer size, second is as
	  fallback in case the packet is larger in the end.
	  Next step: actually use these.

2014-09-09 09:42:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: track max packet size and save allocator negotiated by GstBaseSrc

2014-09-08 16:15:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audiofx/audioecho.c:
	  audioecho: fix example command line

2014-09-07 12:46:08 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix crash with certain videos
	  This is a regression from 1.2 caused by the port
	  to the pad flow combiner.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736192

2014-09-04 16:21:20 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-read-common.h:
	  matroska-demux: Don't handle parse errors at the end of file as an error
	  But only if they happen after the Matroska segment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735833

2014-09-04 12:14:11 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Include redirection target in error messages
	  Just giving the original URI can give the false impression that e.g.
	  that one failed host name resolution, while actually the redirection target
	  did.

2014-09-02 11:13:44 +0400  Andrei Sarakeev <sarakusha@gmail.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Fix synchronization if dynamically changing the FPS
	  https://bugzilla.gnome.org/show_bug.cgi?id=735859

2014-09-02 13:52:43 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/smpte/gstsmpte.c:
	  smpte: Check if input caps are the same and create output caps from video info
	  This makes sure that also properties like the pixel-aspect-ratio are the same
	  between both streams and that the output caps contain all fields necessary for
	  complete video caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735804

2014-09-02 17:22:07 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: replace with gst_buffer_copy
	  gst_buffer_ref and gst_buffer_writable is being used to create a writable copy of source buffer.
	  replacing the same with gst_buffer_copy as the functionality is same.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735880

2014-09-03 23:06:53 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: mark jpeg and png as parsed so avdec_mjpeg can be used too
	  https://bugzilla.gnome.org/show_bug.cgi?id=735971

2014-09-03 11:46:13 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	  gdkpixbufdec: free query after use
	  In gst_gdk_pixbuf_dec_setup_pool(), query is being allocated using
	  gst_query_new_allocation(), but the same is not unreferenced
	  hence calling gst_query_unref() after usage of query.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735950

2014-09-03 23:46:34 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Silence some warnings for normal file contents

2014-09-01 09:56:02 +0200  Nicolas Huet <nicolas.huet@parrot.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Fix parsing issue when the buffer does not have a complete ADTS/LOAS frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=735520

2014-09-02 09:09:49 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp9dec.c:
	  vp9dec: Get input width/height from the codec instead of the input caps
	  They are reported properly by libvpx if the correct struct members are used.
	  This also fixes handling of resolution changes without input caps changes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719359

2013-10-22 18:49:22 +0100  Tom Greenwood <tcdgreenwood@hotmail.com>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Fix for handling resolution changes when decoding VP8
	  If the resolution changes in the bitstream without the input caps changing we
	  would previously output corrupted video or crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719359

2014-09-02 00:55:17 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/vpx/gstvp9dec.c:
	  vp9dec: Fix segfault when a new caps is received
	  Remember to unref the output caps when a new caps event is received
	  as it should generate a new one based on the new caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=734266

2014-09-02 00:54:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/vp8dec.c:
	  tests: vp8dec: add test for caps renegotiation
	  Check that vp8dec can properly accept a new caps when upstream
	  changes it
	  https://bugzilla.gnome.org/show_bug.cgi?id=734266

2014-08-05 10:34:39 +0200  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Reset output and input states when changing format
	  https://bugzilla.gnome.org/show_bug.cgi?id=734266

2014-09-01 16:39:23 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Don't call gst_caps_unref() on template caps when already unreferenced
	  Adding an extra condition while calling gst_caps_unref (templ)
	  and replacing gst_caps_make_writable (gst_caps_ref (caps)) with
	  gst_caps_copy (caps) in line 177, since the functionality is same.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735795

2014-08-29 12:01:27 +0200  Hans de Goede <hdegoede@redhat.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: get_nearest_size: Fix "Unsupported field type" errors
	  Most V4L2 ioctls like try_fmt will adjust input fields to match what the
	  hardware can do rather then returning -EINVAL. As is docmented here:
	  http://linuxtv.org/downloads/v4l-dvb-apis/vidioc-g-fmt.html
	  EINVAL is only returned if the buffer type field is invalid or not supported.
	  So upon requesting V4L2_FIELD_NONE devices which can only do interlaced
	  mode will change the field value to e.g. V4L2_FIELD_BOTTOM as only returning
	  half the lines is the closest they can do to progressive modes.
	  In essence this means that we've failed to get a (usable) progessive mode
	  and should fall back to interlaced mode.
	  This commit adds a check for having gotten a usable field value after the first
	  try_fmt, to force fallback to interlaced mode even if the try_fmt succeeded,
	  thereby fixing get_nearest_size failing on these devices.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735660

2014-08-29 10:57:20 +0200  Hans de Goede <hdegoede@redhat.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: get_nearest_size: Always reinit all struct fields on retry
	  They may have been modified by the ioctl even if it failed. This also makes
	  the S_FMT fallback path try progressive first, making it consistent with the
	  preferred TRY_FMT path.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735660

2014-08-29 11:55:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Store size of data tag in a 64 bit integer locally too
	  Otherwise we will clip the DS64 value of RF64 files to 32 bits again.

2014-08-29 11:53:23 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Use 64 bit scaling functions now that fact is a 64 bit integer

2014-08-27 18:55:18 +0200  Peter G. Baum <peter@dr-baum.net>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: support rf64 format
	  https://bugzilla.gnome.org/show_bug.cgi?id=735627

2014-08-28 13:48:50 -0600  Jason Litzinger <jlitzinger@control4.com>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: Ensure caps before pad added.
	  This stores the stream-start, sets caps, and then adds the pad,
	  which ensures that the caps are set for the "pad-added" callback.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735626

2014-08-28 15:03:50 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Fallback to PTS if DTS is missing
	  Fixing a regression introduce when fixing:
	  https://bugzilla.gnome.org/show_bug.cgi?id=731352

2014-08-28 16:13:29 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Remove impossible error condition
	  We return EOS after the first buffer, and GstPad will make sure now that we
	  won't get any other buffer afterwards until a flush happens. No need to check
	  for it ourselves.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735581

2014-08-28 13:53:23 +0530  Vineeth T M <vineeth.tm@samsung.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	  gdkpixbufdec: EOS and NOT_LINKED are no errors in general
	  Don't post an error message for them but let upstream handle
	  anything accordingly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=735564

2014-08-27 21:07:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Correctly offset timestamp
	  The previous method would break AV sync in the case audio or video
	  didn't start at the same point in running time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731352

2014-08-27 20:56:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Save dts from buffer
	  We no longer set dts in muxed buffer. This would lead to encoding tags
	  with timestamp 0 instead of the timestamp of previous buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731352

2014-07-28 20:58:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Ensure Timestamp starts at 0
	  FLV documentation stipulates that timestamp must start at zero.
	  In order to respect this rule, keep the first timestamp around
	  and offset the timestamp from this value. This allow for longer
	  recording time in presence of timestamp that does not start
	  at 0 already.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731352

2014-06-06 23:17:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvmux.c:
	  flv: Tag timestamp are DTS not PTS
	  The tags in FLV are DTS. In audio cases, and for many video format this makes
	  no difference, but for AVC with B-Frames, PTS need to be computed from
	  composition timestamp CTS, with PTS = DTS + CTS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731352

2014-08-07 21:58:14 -0400  Youness Alaoui <kakaroto@kakaroto.homelinux.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Allow rtp caps without clock-rate
	  The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.
	  https://bugzilla.gnome.org/show_bug.cgi?id=734322

2014-08-18 14:05:52 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid crashing on dash streams
	  DASH/fragmented moov might have no samples as those are carried
	  in moof fragments. Avoid crashing or failing the stream because
	  of that.

2014-08-18 10:33:48 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* tests/examples/equalizer/demo.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  examples: use 'post-messages' property instead of deprecated 'message' property
	  https://bugzilla.gnome.org/show_bug.cgi?id=734979

2014-08-18 11:45:54 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst/udp/gstudpsrc.c:
	  udp: fix udpsrc documentation
	  udpsrc gtk-doc documentation refers to sockfd and closefd properties which has
	  been removed. This patch replaces those references to socket and close-socket
	  respectively.
	  https://bugzilla.gnome.org/show_bug.cgi?id=734987

2014-08-15 10:09:56 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Make the default timescale 1/1800 second
	  The old default timescale of 1 millisecond produces irrational
	  numbers for a lot of framerate/audio-packet-duration multiples.
	  1/1800 is a nicer number, as it tends to produce better fractions
	  and therefore slightly higher accuracy overall

2014-08-15 01:17:27 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska: Use gst_video_guess_framerate() function
	  Remove local framerate guessing function in favour of
	  the new gst_video_guess_framerate() function.

2014-08-15 01:12:20 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/Makefile.am:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Improve framerate calculation/guessing
	  Change the way the output framerate is calculated
	  to ignore the first sample (which is sometimes truncated
	  in my testing) and use the new gst_video_guess_framerate()
	  function to recognise common standard framerates better.
	  Remove the code that was sorting the first 20 sample
	  durations and then ignoring the result.

2014-08-14 16:36:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Use the best width/height/etc if downstream can handle that
	  Before it was always using whatever downstream preferred, while
	  the code and documentation claimed something different.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727180

2014-08-14 11:29:00 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Avoid double free of VideoConvert
	  https://bugzilla.gnome.org/show_bug.cgi?id=734764

2014-08-13 11:58:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix indentation

2014-08-13 11:54:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: un-break duration querying
	  Commit 2b9493b5 broke this in two ways: a) we should only
	  pass duration queries in TIME format upstream (or at least
	  not those in DEFAULT or BYTE format), and b) we mustn't
	  overwrite the default value of 'res' from TRUE to FALSE
	  and not set it again later. This led to bogus durations
	  being reported for FLV playback from file, because TIME
	  queries would fail (as 'res' had been set to FALSE) and
	  parsers then do a BYTE query as fallback and try to
	  guesstimate something in return, which of course goes
	  horribly wrong since the BYTE size returned is for the
	  muxed file.

2014-08-13 13:23:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Allow any raw caps in passthrough mode, not just the ones we handle

2014-08-13 13:04:21 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Allow ANY capsfeatures, but only in passthrough mode
	  When changing the properties to not be in passthrough mode anymore,
	  we will only accept caps we can process ourselves, potentially causing
	  a not-negotiated error.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720345

2014-08-12 11:34:30 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update for git

2014-08-12 11:33:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	  configure: build ximagesrc again when checks succeed
	  Third time lucky, hopefully.

2014-08-11 09:26:17 +0100  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	  configure: fix x11 checks to be non-fatal again
	  Must pass an action-if-not-found argument to
	  PKG_CHECK_MODULES or it will error out when
	  it can't find the module requested. Also fix
	  AC_CHECK_LIB usage, extra libs argument was
	  in the wrong place.

2014-08-07 17:12:38 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: forward DISCONT from upstream to the output streams
	  This makes sense in DASH reverse playback, where the upstream dashdemux
	  will download DASH segments in reverse order, but push their buffers
	  forward to qtdemux and mark each segment start as DISCONT. This needs
	  to be forwarded downstream to the parser/decoder, otherwise it won't work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=734443

2014-08-10 18:55:07 +0100  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	  configure: use pkg-config to detect x11 and simplify checks
	  AC_PATH_XTRA macro unnecessarily pulls in libSM and libICE.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731047

2014-08-10 12:30:07 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: adjust test data to avoid NAL chopping
	  ... and correspondingly unexpected buffer sizes.

2014-08-09 14:22:42 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* ext/speex/gstspeexenc.c:
	  speexenc: Improve annotation of internal function
	  https://bugzilla.gnome.org/show_bug.cgi?id=734542

2014-08-08 12:54:30 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/shapewipe/gstshapewipe.c:
	* tests/examples/shapewipe/shapewipe-example.c:
	  shapewipe: Unref caps and element after usage
	  https://bugzilla.gnome.org/show_bug.cgi?id=734478

2014-08-09 20:47:30 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: improve debug logging of fourccs
	  If we can't show ASCII, at least show them
	  in big endian order.

2014-08-09 20:46:04 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for 'wma ' mapping as found in some ismv files
	  e.g. To_The_Limit_720_2962.ismv

2014-08-09 18:31:20 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for 'vc-1' mapping as found in some ismv files
	  e.g. To_The_Limit_720_2962.ismv

2014-08-07 16:34:36 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Unref pad template caps after use
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734435

2014-08-08 12:36:01 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Unref allowed caps after usage
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734474

2014-08-08 12:40:49 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Unref pad template caps after usage
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734475

2014-08-08 12:44:09 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/debugutils/gstnavseek.c:
	  navseek: Unref peer pad after usage
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734476

2014-08-08 12:29:52 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Unref pad template caps after usage
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734473

2014-08-05 11:47:39 +0200  Srimanta Panda <srimanta@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: append packetization mode parameter to SDP
	  Append packetization-mode parameter to SDP description.
	  Packetization mode signals the properties of an RTP payload type.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733556

2014-08-08 03:58:14 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	  isomp4/qtmux: Write correct file duration when gaps exist.
	  When writing out a trak with an edit list, make sure the
	  overall file duration is also updated to reflect the
	  lengthening of the stream.
	  Add some more debug to qtdemux to warn about streams that
	  are longer than the file and get truncated.

2014-08-04 15:39:17 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Push the correct segment in TCP mode when seeking

2014-08-03 12:33:32 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: unbreak au aligned byte-stream payloading

2014-07-22 13:24:09 +0200  Srimanta Panda <srimanta@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: append profile-level-id to SDP
	  Append profile-level-id to SDP if available.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733539

2014-07-31 18:47:49 +0200  Edward Hervey <edward@collabora.com>

	* Makefile.am:
	* common:
	  Makefile: Add usage of build-checks step
	  Allows building checks without running them

2014-07-31 09:53:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/ximage/ximageutil.c:
	  ximagesrc: Fix warning about missing return value

2014-07-24 15:28:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/ximageutil.c:
	* sys/ximage/ximageutil.h:
	  ximagesrc: Add missing return value to Buffer dispose function
	  Depending ont he build, the method could return FALSE, hence never
	  free the buffers, or already TRUE and lead to a crash:
	  Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=733695

2014-07-28 16:49:16 +0200  Philippe Normand <philn@igalia.com>

	* gst/interleave/interleave.c:
	* tests/check/elements/interleave.c:
	  interleave: set output caps layout to interleaved
	  Set output caps layout independently from input caps layout which can
	  be either non-interleaved or interleaved.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733866

2014-07-26 12:06:39 -0300  Thiago Santos <ts.santos@osg.sisa.samsung.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: clear gcond

2014-07-25 14:30:33 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  Revert "v4l2bufferpool: Workaround elements not requesting any buffers"
	  This was a tempory workaround, we should fix the encoders that do not
	  negotatiate the amount of buffers they need.
	  This reverts commit d03bcba3db15d06dbdea6b776a6f28ed2f03272a.

2014-07-08 14:31:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't share own pool if min exceed V4L2 capacity
	  If the minimum required buffer exceed V4L2 capacity, don't share down
	  pool. This allow support very high latency, like with x264enc default
	  encoding settings.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732288

2014-07-25 17:42:20 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: query minimum required buffers for output
	  Some v4l2 devices could require a minimum buffers different from default
	  values. Rather than blindly propose a pool with min-buffers set to the
	  default value, it ask the device using control ioctl.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733750

2014-07-23 18:40:10 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: use directly 'obj' instead of 'v4l2sink->v4l2object'
	  https://bugzilla.gnome.org/show_bug.cgi?id=733616

2014-07-23 18:39:50 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2: set debug messages according to device type and IO mode
	  https://bugzilla.gnome.org/show_bug.cgi?id=733616

2014-05-24 19:02:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Remove is_active checks
	  These checks are no longer required with recent change to the bufferpool. This
	  should allow changing the configuartion, hence the way forward renegotiation
	  support.
	  https://bugzilla.gnome.org/show_bug.cgi?id=728268

2014-07-21 18:11:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_lang.c:
	  qtdemux: fix language code parsing for 3-letter codes starting with 'a'
	  And handle special value for 'unspecified' explicitly.
	  https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFChap4/qtff4.html

2014-07-08 02:18:27 +0200  Nicola Murino <nicola.murino@gmail.com>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Add support for encoding from NV21 and NV12
	  https://bugzilla.gnome.org/show_bug.cgi?id=732870

2014-07-19 18:04:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.4.0 ===

2014-07-19 17:20:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.4.0

2014-07-19 16:35:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2014-07-19 12:32:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: Update translations

2014-07-19 11:30:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/videobox/gstvideobox.c:
	  videobox: Don't overwrite the first component with the alpha value for BGRx
	  Instead leave the x component unset when filling the borders.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733380

2014-07-16 17:18:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Properly report in the CAPS query that we can convert ADTS<->RAW
	  https://bugzilla.gnome.org/show_bug.cgi?id=733190

2014-07-13 16:05:56 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/replaygain/gstrgvolume.c:
	  rgvolume: Avoid taking unnecessary refs
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122

2014-07-13 16:04:23 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: Avoid taking an unnecessary ref
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122

2014-07-15 16:59:06 +0200  Piotr Drąg <piotrdrag@gmail.com>

	* po/POTFILES.in:
	  po: update POTFILES
	  https://bugzilla.gnome.org/show_bug.cgi?id=733208

2014-07-11 13:35:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Fix copy threshold implementation
	  We cannot allocate new buffer in acquire, otherwise the base class
	  is not aware and get confused. Instead, copy in _process(). This leads
	  to crash on finalize.
	  Fixes regression, see https://bugzilla.gnome.org/show_bug.cgi?id=732912

=== release 1.3.91 ===

2014-07-11 11:38:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.3.91

2014-07-11 10:58:08 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2014-07-10 18:11:20 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2allocator: Use qdata instead of parenting to DmabufMemory
	  Parenting V4l2Memory to DmabufMemory was in conflict with recent
	  optimization in DmabufMemory to avoid dup(), and didn't work with
	  memory sharing. Instead, use a qdata and it's destroy notify.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730441

2014-07-11 08:52:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/da.po:
	* po/de.po:
	* po/hu.po:
	* po/id.po:
	* po/pl.po:
	* po/ru.po:
	* po/uk.po:
	* po/vi.po:
	  po: Update translations

2014-07-08 17:50:47 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Workaround elements not requesting any buffers
	  This is a workaround for element that don't request buffers when
	  they should.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732288

2014-07-06 11:27:36 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* tests/icles/videocrop-test.c:
	  tests: fix pipeline leak in videocrop test
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732976

2014-07-06 11:26:46 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* tests/examples/rtp/client-rtpaux.c:
	  examples: client-rtpaux: Release reference to parent when done
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732976

2014-07-10 17:19:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix query leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=733003

2014-07-10 12:10:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Return not-negotiated if we got no caps or caps negotiation failed
	  And do it always, not inside a g_return_val_if_fail().
	  See https://bugzilla.gnome.org/show_bug.cgi?id=732939

2014-07-08 13:34:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Ensure internal pool activation
	  Before we would hit an assertion "'gst_buffer_pool_is_active (bpool)' failed"
	  if the internal pool was not used to push buffer downstrea, hence not
	  given to the baseclass.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732912

2014-07-04 20:22:10 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: fix double unlock in segment seek segment code path
	  We only want to unlock if we push an event downstream and
	  jump to done_unlock label afterwards. We would also unlock
	  in case of a segment seek and then unlock again later, and
	  nothing good can come of that.
	  (This code looks a bit dodgy anyway though, shouldn't it
	  also bail out with FLOW_EOS here in case of a segment seek
	  scenario, just without the event?)

2014-07-04 19:45:55 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: suppress glib criticals caused by testing deprecated dts methods

2014-07-04 03:21:30 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/avi/gstavidemux.c:
	* gst/wavparse/gstwavparse.c:
	  avidemux, wavparse: Print invalid fourcc in hex
	  Previously this was printed as characters which caused later processing
	  of the error message to sometimes warn about non-UTF-8 characters.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732714

2014-07-03 15:21:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Pool might be NULL in decide allocation
	  If special stride is needed and downstream don't support VideoMeta,
	  pool might be NULL in order to let the baseclass create a generic
	  pool­. This would lead to assertion with on Exynos with:
	  gst-launch-1.0 -v filesrc location=mov ! qtdemux ! h264parse ! \
	  v4l2video8dec ! fakesink
	  https://bugzilla.gnome.org/show_bug.cgi?id=732707

2014-07-03 15:29:54 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Handle FD error during poll
	  This will ensure we fail earlier if something unrecoverable
	  happens.

2014-07-03 15:28:45 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Wait before polling if queue is empty
	  In kernel before 3.17, polling during queue underrun would unblock right
	  away and trigger POLLERR. As we are not handling POLLERR, we would endup
	  blocking in DQBUF call, which won't be unblocked correctly when going
	  to NULL state. A deadlock at start caused by locking error in libv4l2 was
	  also seen before this patch. Instead, we wait until the queue is no longer
	  empty before polling.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731015

2014-07-02 16:01:47 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix for mikey api change

2014-06-30 10:29:54 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix probing and enumeration of stepwise frame sizes
	  The code enumerating STEPWISE framesizes would start from
	  (min_w, min_h) and then add (step_w, step_h) to get the
	  next framesize. However, it should really allow any width
	  from min_w to max_w with step_w and same for heights.
	  Secondly, we would add and probe each individual stepped
	  frame size to the caps as separate structure, which would
	  lead to hundreds if not thousands of structs ending up in
	  the probed caps. Use integer ranges with steps instead.
	  This was particularly noticable with the Raspberry Pi Cam.
	  https://bugzilla.gnome.org/show_bug.cgi?id=724521
	  https://bugzilla.gnome.org/show_bug.cgi?id=732458
	  https://bugzilla.gnome.org/show_bug.cgi?id=726521

2014-06-27 11:33:06 +0100  Daniel Drake <drake@endlessm.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: drop workaround for misbehaving TRY_FMT
	  This workaround from 2011 was causing 25 S_FMT ioctls to be sent
	  to my UVC webcam from under gst_v4l2_object_get_caps as it probes
	  all the formats. In total, this adds up to about 5 seconds of
	  execution time, or a 10 second delay while starting up cheese.
	  These ioctls come from a workaround from 2011 where TRY_FMT might
	  make changes to hardware settings, so S_FMT was used to restore
	  the original config:
	  https://bugzilla.gnome.org/show_bug.cgi?id=649067
	  The driver bug is now assumed fixed. Remove the workaround to fix the
	  long startup delay.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732326

2014-07-01 12:50:31 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: reset QoS on segment event
	  https://bugzilla.gnome.org/show_bug.cgi?id=732540

2014-07-01 15:14:34 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: send gap events instead of segment tricks
	  This fixes missing frames from being time skipped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=732372

2014-06-30 00:00:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: Fix memory leaks in unit test

2014-06-29 23:55:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Don't leak caps

2014-06-29 20:02:14 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Fix compiler warning when compiling with G_DISABLE_ASSERT

2014-06-29 19:59:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Fix compiler warning when compiling with G_DISABLE_ASSERT

2014-06-29 19:57:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Fix compiler warnings when compiling with G_DISABLE_ASSERT

2014-06-29 19:54:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: Fix compiler warnings when compiling with G_DISABLE_ASSERT

2014-06-29 17:05:13 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/pulse/pulsedeviceprovider.c:
	  pulse: fix compiler warnings when compiling with -DG_DISABLE_ASSERT
	  Compiler complains about uninitialised variables in the impossible
	  'default' code path in device provider source/sink switch-case.

2014-06-29 17:03:17 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2: fix compiler warnings when compiling with -DG_DISABLE_ASSERT
	  Compiler complains about uninitialised variables in the impossible
	  'default' code path in device provider source/sink switch-case.

2014-06-28 17:40:45 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/matroskaparse.c:
	  tests: matroskaparse: fail on errors and disable pull mode test
	  Actually look for error messages on the bus and fail if there
	  is one before the EOS message. Disable pull mode test which is
	  pointless as long as matroskaparse only supports push mode
	  (pull mode support has not been ported over to 1.0).

2014-06-28 17:37:23 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: don't error out if there's not enough data in the adapter
	  gst_matroska_parse_take() would return FLOW_ERROR instead of
	  FLOW_EOS in case there's less data in the adapter than requested,
	  because buffer is NULL in that case which triggers the error
	  code path. This made the unit test fail (occasionally at least,
	  because of a bug in the unit test there's a race and it would
	  happen only sporadically).

2014-06-28 16:53:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/videomixerorc-dist.c:
	* gst/videomixer/videomixerorc-dist.h:
	  videomixer: Update dist generated ORC files

2014-06-28 16:48:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/gstcms.c:
	* gst/videomixer/gstcms.h:
	* gst/videomixer/videoconvert.c:
	* gst/videomixer/videoconvert.h:
	* gst/videomixer/videomixerorc.orc:
	  videomixer: Update videoconvert code from -base
	  And also rename the remaining symbols to prevent conflicts
	  during static linking.
	  https://bugzilla.gnome.org/show_bug.cgi?id=728443

2014-06-28 13:01:46 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/autodetect/gstautovideosrc.c:
	  autovideosrc: use videotestsrc as fallback element instead of fakesrc
	  fakesrc doesn't announce video caps, so most video pipelines will
	  just error out with not-negotiated if a fallback element is created.

2014-06-28 12:44:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautodetect.c:
	* gst/autodetect/gstautodetect.h:
	  autoaudiosrc: use audiotestsrc as fallback element instead of fakesrc
	  fakesrc doesn't announce audio caps, so most audio pipelines will
	  just error out with not-negotiated if a fallback element is created.

=== release 1.3.90 ===

2014-06-28 11:21:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.3.90

2014-06-28 11:08:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2014-06-26 14:52:57 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/Makefile.am:
	* ext/pulse/plugin.c:
	* ext/pulse/pulsedeviceprovider.c:
	* ext/pulse/pulsedeviceprovider.h:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/gstv4l2deviceprovider.h:
	  Rename GstDeviceMonitor to GstDeviceProvider

2014-06-24 09:14:40 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/videobox.c:
	  videobox: Add unit test
	  https://bugzilla.gnome.org/show_bug.cgi?id=732144

2014-06-16 11:35:39 +0200  Thibault Saunier <tsaunier@gnome.org>

	* gst/videomixer/videomixer2.c:
	  videomixer: Declare as Compositor in 'klass'

2014-06-26 13:50:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix speex caps
	  Decoder complains about "notification: Invalid mode encountered.
	  The stream is corrupted" though, even if it works, so there's
	  probably something wrong with the generated codec headers.

2014-06-26 13:43:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: fix speex in FLV
	  Speex in FLV is always mono @ 16kHz, see
	  http://download.macromedia.com/f4v/video_file_format_spec_v10_1.pdf
	  section E.4.2.1: "If the SoundFormat indicates Speex, the audio is
	  compressed mono sampled at 16 kHz, the SoundRate shall be 0, the
	  SoundSize shall be 1, and the SoundType shall be 0"
	  Also see https://bugzilla.gnome.org/show_bug.cgi?id=683622

2014-06-26 05:19:57 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: Add object type id and fourcc for DTS/DTS-HD
	  Enables playback for files with DTS audio tracks.
	  Also add an extra AC-3 variant fourcc from Nero

2014-03-13 10:35:30 +0100  David Fernandez <d.fernandezlop@gmail.com>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Solve segmentation fault when src caps are configured
	  Change function pointers to NULL while holding the lock to avoid
	  race conditions
	  https://bugzilla.gnome.org/show_bug.cgi?id=701110

2014-06-25 14:34:21 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve SR packet handling
	  Implement 3 different cases for handling the SR:
	  1) we don't have enough timing information to handle the SR packet and
	  we need to wait a little for more RTP packets. In that case we keep
	  the SR packet around and retry when we get an RTP packet in the
	  chain function.
	  2) the SR packet has a too old timestamp and should be discarded. It is
	  labeled invalid and the last_sr is cleared.
	  3) the SR packet is ok and there is enough timing information, proceed
	  with processing the SR packet.
	  Before this patch, case 2) and 1) were handled in the same way,
	  resulting that SR packets with too old timestamps were checked over and
	  over again for each RTP packet.

2014-06-24 10:47:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/udpsink.c:
	  tests: add udpsink test to check client add/remove

2014-06-23 16:13:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/udpsink.c:
	  tests: port udpsink tests to 1.0
	  They all seem a bit pointless though.

2014-06-23 19:55:29 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/avi/gstavimux.c:
	  avimux: Add UYVY format

2014-06-06 11:20:21 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  gstrtpssrcdemux: manage ssrc of RTCP RR packets
	  https://bugzilla.gnome.org/show_bug.cgi?id=731324

2014-06-23 20:53:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Update offset after parsing adtl chunk
	  Otherwise we will parse it over and over again without ever
	  getting past it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731533

2013-07-07 20:18:27 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: remove legacy code for passing a window handle
	  "have-ns-view" and the "embed" property was kept in 0.10 for
	  backwards compatibility but it's no longer used in favor of
	  the GstVideoOverlay interface
	  https://bugzilla.gnome.org/show_bug.cgi?id=703753

2014-06-22 19:36:14 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

2014-06-22 19:26:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: Don't call GST_DEBUG_OBJECT() and other macros with non-GObject objects
	  It will crash with latest GLib GIT and was never supposed to work before
	  either.

=== release 1.3.3 ===

2014-06-22 18:08:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.3.3

2014-06-22 17:36:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2014-06-22 14:24:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: Update translations

2014-06-21 01:32:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/pulse/pulsedevicemonitor.c:
	* sys/v4l2/gstv4l2devicemonitor.c:
	  pulse, v4l2: update for device "klass" -> "device-class" rename

2014-06-20 12:21:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: optimisation: avoid unnecessary memory ref/unrefs
	  We know the buffer will stay valid and we will also not
	  modify the buffer, we just want to send out the data.

2014-06-19 14:59:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: avoid some unnecessary run-time type checks

2014-06-19 16:17:23 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: pass the stream id when asking for crypto params
	  This way the app can choose different parameters for each stream.

2014-05-20 14:58:07 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add support for key length parameters
	  This patch adds supports for the incoming key management parameters for
	  encryption and authentication key lengths.
	  It also adds a new signal request-rtcp-key that allows the user to
	  provide the crypto parameters and key for the RTCP stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730473

2014-06-19 15:25:01 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtp/gstrtpvp8depay.c:
	  vp8depay: fix header size checking
	  Use a different variable name to make it clear that we are calculating
	  the header size.
	  Correctly check that we have enough bytes to read the header bits. We
	  were checking if there were 5 bytes available in the header while we
	  only needed 3, causing the packet to be discarded as too small.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723595

2014-05-20 12:39:31 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: propagate the GST_BUFFER_FLAG_DISCONT flag
	  Similarly to what we did with the DELTA_UNIT flag, this patch
	  propagates the DISCONT flag to the first RTP packet being used to transfer a
	  DISCONT buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730563

2014-05-06 17:42:14 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: propagate the GST_BUFFER_FLAG_DELTA_UNIT flag
	  Downstream elements may be interested knowing if a RTP packet is the start
	  of a key frame (to implement a RTP extension as defined in the
	  ONVIF Streaming Spec for example).
	  We do this by checking the GST_BUFFER_FLAG_DELTA_UNIT flag we receive from
	  upstream and propagate it to the *first* RTP packet outputted to transfer this
	  buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730563

2014-05-20 13:58:20 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4gpay.h:
	  gstrtpmp4gpay: propagate the GST_BUFFER_FLAG_DISCONT flag
	  Propagate the DISCONT flag to the first RTP packet being used to transfer
	  a DISCONT buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730563

2014-05-20 13:58:20 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: propagate the GST_BUFFER_FLAG_DISCONT flag
	  Propagate the DISCONT flag to the first RTP packet being used to transfer
	  a DISCONT buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730563

2014-06-18 15:03:25 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: don't leak flow combiner

2014-06-18 14:38:55 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpjp2kpay: pre-allocate buffer-list of the right size

2014-06-18 14:34:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: pre-allocate buffer list of the right size

2014-06-18 14:19:28 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpmp4vpay.c:
	  rtpmp4vpay: pre-allocate buffer list of the right size

2014-06-18 13:44:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: allocate bitreader on the stack

2014-06-18 13:29:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: post error message on bus on error and don't use g_message()

2014-06-18 13:20:44 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: couple of minor optimisations
	  Pre-allocate buffer list of the right size to avoid re-allocs.
	  Avoid plenty of double runtime cast checks and re-doing the
	  same calculation over and over again in rtp_vp8_calc_payload_len().
	  Only call gst_buffer_get_size() once.

2014-06-18 08:10:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: pre-allocate buffer list of the right size
	  To avoid re-allocs.

2014-06-18 07:52:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: pre-allocate bufferlist of the right size
	  To avoid unnecessary re-allocs.

2014-06-16 20:15:43 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264pay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtph264pay: push single buffer directly, no need to wrap it in a bufferlist
	  No point in a buffer list if we just have one single
	  buffer to push. Fix up unit test to handle that case
	  as well.

2014-06-16 15:35:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtp/gstrtpvrawpay.h:
	  rtpvrawpay: make chunks per frame configurable
	  Bit of a misnomer because it's really chunks per field
	  and not per frame, but we're going to ignore that for
	  the time being.

2014-06-16 14:52:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtp/gstrtpvrawpay.h:
	  rtpvrawpay: remove unused variables

2014-06-16 14:44:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: pre-allocate buffer lists of sufficient size
	  Avoids unnecessary reallocs when appending buffers
	  to the bufferlist.

2014-06-16 13:51:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: micro-optimise variable access in inner loop
	  Store some values that don't change during the execution
	  of the inner loops locally, so the compiler knows that too.

2014-06-16 13:38:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: use buffer lists
	  Collect buffers to send out in buffer lists instead of
	  pushing out single buffers one at a time. For HD video
	  each frame might easily add up to a couple of thousand
	  packets, multiply that by the frame rate and that's a
	  lot of push() and sendmsg() calls per second.
	  A good reason to push out buffers as early as possible is
	  latency, so we don't accumulate the whole frame in a single
	  buffer list, but instead push it out in a few chunks, which
	  is hopefully a reasonable compromise.

2014-06-16 16:40:07 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	  udp: improve element descriptions for dynudpsink and multiudpsink

2014-06-16 16:17:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	  udp: remove suppression of compiler warnings for deprecated GLib API
	  Not needed any more.

2014-06-17 13:16:27 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix caps negotiation issue
	  Make sure that if AYUV is received it will detect that it can produce
	  both RGB and YUV formats
	  Signed-off-by: Ravi Kiran K N <ravi.kiran@samsung.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=725248

2014-06-16 12:02:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: fix double frees
	  Fix double-frees introduced to fix another coverity report.
	  CID 1223053

2014-06-13 10:12:07 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstdynudpsink.c:
	  dynudpsink: return FLUSHING when sendto got canceled, not an error

2014-06-13 09:52:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/oss/gstosshelper.c:
	  oss: simplify probed caps before returning them
	  Exposes all formats in the first structure if the
	  rest is the same for all of them.

2014-06-13 09:45:28 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/oss/gstosshelper.c:
	  oss: make sure 16-bit formats are before 8-bit formats in probed caps
	  Probe supported formats in order of desirability rather than in
	  what order they may happen to be in the formats bitmask. Fixes
	  accidentally exposure of 8-bit formats in caps before 16-bit formats
	  (in case where U16 was not supported S8 might be listed before S16).
	  https://bugzilla.gnome.org/show_bug.cgi?id=706884

2014-06-12 16:36:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Cleanly handle v4l2_allocator_new failure

2014-06-12 11:24:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheordepay: fix leaks
	  Coverity 1212163

2014-06-12 11:16:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: leak fixes
	  Coverity 1212159

2014-06-12 11:11:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: fix leak
	  Coverity 1212157

2014-06-12 10:43:53 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: fix leaks
	  Coverity 1212149

2014-06-12 10:31:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpdvpay.c:
	  rtpdvpay: catch failures to map buffer
	  Coverity 1139741

2014-06-11 17:43:42 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: guard against having no MIME type
	  The code would previously crash trying to insert a NULL string
	  into a hash table.
	  It does seem a little broken that indexing is done by MIME type
	  and not by index though, unless the spec says there cannot be
	  two parts with the same MIME type.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659573

2014-06-10 15:42:14 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: Send stream-start event
	  This event was not sent. Send it before caps, this requires the pad to
	  be parented. This removes warning like: "Got data flow before
	  stream-start event".
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731475

2014-06-10 15:33:33 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid looping indefinitely in broken svq3 files
	  Abort if an atom with size 0 is read from within the svq3 stsd
	  atoms
	  https://bugzilla.gnome.org/show_bug.cgi?id=726512

2014-06-10 10:52:23 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: add const where appropriate

2014-06-09 10:39:20 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/speex/gstspeexenc.c:
	  speexenc: add missing va_end in variadic function
	  Coverity 1139944

2014-06-09 10:04:38 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Attempt upstream seek first
	  If we have an upstream element that can handle the seek (such as
	  rtmpsrc), try to do that first before attempting it ourself.

2014-06-04 11:34:27 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: do not include codec_data on raw audio caps
	  If the wav header contains an extended chunk, we want to keep
	  the codec_data field, but not for raw audio.
	  This fixes some elements (such as adder) from failing to intersect
	  raw audio caps which would otherwise be intersectable.

2014-06-05 09:38:29 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Query duration upstream first
	  Upstream elements (like rtmpsrc) might be able to provide the duration
	  more accurately than flvdemux. Especially with index-less vod files

2014-05-30 19:37:57 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Cleanup poll method and retry on EINTR/EAGAIN
	  https://bugzilla.gnome.org/show_bug.cgi?id=731015

2014-03-06 16:37:51 +0100  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: set RESYNC buffer flag when bridging large PTS gaps
	  So downstream gets notified when this happens.
	  https://bugzilla.gnome.org/show_bug.cgi?id=725903

2014-06-03 17:59:32 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtprtx.c:
	  rtprtx: Reset state on each iteration
	  Otherwise it didn't wait for the test to finish before checking the results.
	  https://bugzilla.gnome.org/show_bug.cgi?id=728501

2014-05-09 14:22:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: don't leak doctype string in error code path
	  CID 1212145.

2014-05-20 08:20:42 +0200  Edward Hervey <edward@collabora.com>

	* ext/vpx/gstvp9enc.c:
	  vp9enc: Don't dereference NULL checks
	  CID #1197703

2014-05-20 08:23:06 +0200  Edward Hervey <edward@collabora.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Don't dereference NULL variable
	  CID #1139838

2014-05-30 14:32:42 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: upstream handles seek if fragmented and on time segment
	  Otherwise we can reject seeks on local files that contain fragmented-like
	  atoms like 'mvex'. Also improve a message log
	  https://bugzilla.gnome.org/show_bug.cgi?id=730722

2014-05-30 16:43:44 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: make sure we call handle_nal for each NAL
	  Call handle_nal for each NAL in the STAP-A RTP packet. This makes
	  sure we correctly extract the SPS and PPS.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730999

2014-05-07 14:09:06 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Add custom sticky event to contain the HTTP request and response headers
	  This can be useful to e.g. get cookie information downstream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=729707

2014-05-26 19:47:39 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: remove stream last flow return
	  GstPad already stores that information
	  https://bugzilla.gnome.org/show_bug.cgi?id=709224

2014-05-26 19:37:46 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: remove last flow return from stream struct
	  It is already stored on GstPad on core
	  https://bugzilla.gnome.org/show_bug.cgi?id=709224

2014-05-26 19:19:45 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Use GstFlowCombiner
	  Use the flow combiner to have the standard combination results and avoid
	  repeating the same code
	  https://bugzilla.gnome.org/show_bug.cgi?id=709224

2014-05-26 13:21:25 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	  matroskademux: use GstFlowCombiner
	  Use the flow combiner to have the standard combination results and avoid
	  repeating the same code
	  https://bugzilla.gnome.org/show_bug.cgi?id=709224

2014-05-26 13:04:10 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: use GstFlowCombiner
	  Removes flow return combination code to use the newly added GstFlowCombiner

2014-05-23 17:53:00 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: use GstFlowCombiner
	  Removes the common code to combining flow returns to let it be
	  handled by core gstutils' GstFlowCombiner
	  https://bugzilla.gnome.org/show_bug.cgi?id=709224

2014-05-26 10:59:55 -0400  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: implement gstvideosink.show_frame instead of gstbasesink.render
	  It allows to show preroll frame. Especially it allows to update the
	  frame when seeking in PAUSED state.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722303

2014-05-26 10:59:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Cleanup old pad alloc declaration

2014-05-26 12:34:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2bufferpool: Copy already queued buffer
	  This is required as during preroll we pass the first buffer twice, hence already
	  queued. It is also useful, to allow filters replaying a previous rendered buffers.
	  This will require 1 more buffer in sink if last-sample is enabled, since the last
	  sample will not be the same as the currently queued buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722303

2014-05-24 20:20:07 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2bufferpool: Port to bufferpool flush_start/stop method
	  Port the buffer pool to use the new flush_start/flush_stop virtual
	  methods added to GstBufferPool.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727611

2014-05-25 17:40:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update

2014-05-25 16:54:18 +0200  Piotr Drąg <piotrdrag@gmail.com>

	* po/POTFILES.in:
	  po: update POTFILES
	  https://bugzilla.gnome.org/show_bug.cgi?id=726556

2014-05-24 23:51:58 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Don't queue all the buffers before dequeueing first
	  For output device, we where queuing all the buffers, and then we would
	  dequeue one. This means we only have 1 buffer for the pipeline, no matter
	  the size of the queue. Instead, start dequeued when min_latency is reached.
	  Eventually, this the min_latency should also be affected by control
	  MIN_BUFFERS_FOR_OUTPUT (use by encoders).

2014-05-24 23:49:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Simply read back the config to update the query
	  It's easy to get the min/max outdate when hacking decide allocation. In
	  order to avoid this, simply read back the choosen value from the config.

2014-05-24 23:31:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Cleanup and fix calculation of latency
	  Calculation of num_buffers (the max latency in buffers) was
	  up-side-down.  If we can allcoate, then our maximum latency match
	  pool maximum number of buffers. Also renamed it to max latency. Finally
	  introduced a min_latency for clarity.

2014-05-24 20:00:14 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/v4l2_calls.c:
	  Revert "v4l2bufferpool: Port to bufferpool flush_start/stop method"
	  This reverts commit 2e0fb42e868fc9f6d98b028def80a3e953527307.
	  Conflicts:
	  sys/v4l2/gstv4l2allocator.c
	  sys/v4l2/gstv4l2bufferpool.c
	  sys/v4l2/gstv4l2videodec.c

2014-05-24 18:56:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix configuration of other_pool and importation case
	  Fix the choice of min/max, don't override the min/max with own pool selected
	  size, correct other_pool is_active check, start from other_pool config when
	  configuring the other pool and finally validate the configuration.

2014-05-24 18:45:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Use proposed allocator as default

2014-05-24 18:43:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Fix USERPTR map flags
	  We need to map READ only for output and write only for capture, we where
	  doing the opposite. This fixing USERPTR with glimagesink
	  https://bugzilla.gnome.org/show_bug.cgi?id=730698

2014-05-24 11:16:35 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: parse tkhd transformation matrix and add tags if appropriate
	  Handle the transformation matrix cases where there are only simple rotations
	  (90, 180 or 270 degrees) and use a tag for those cases. This is a common scenario
	  when recording with mobile devices
	  https://bugzilla.gnome.org/show_bug.cgi?id=679522

2014-05-23 19:10:21 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Prevent num_queued from going negative

2014-05-23 18:25:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: don't stop if loop returned FLUSHING
	  The decodeing thread returning flushing isn't an error, we should simply
	  try starting the task again. If it's actually flushing, it will stop again by itself.

2014-05-23 17:54:20 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Handle early task stop

2014-05-23 17:28:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Handle gst_pad_start_task() failure

2014-05-23 17:19:07 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Add trace for FLUSH_START/STOP handling

2014-05-23 17:18:16 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Fix use of atomic value

2014-05-23 17:01:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Improve debugging
	  No need to use obj->element, the pool now have a significant name. Also don't
	  warn if flushing.

2014-05-23 17:01:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Fix handle_frame error handling

2014-05-23 15:56:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Add a trace when _start() is called

2014-05-23 15:56:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Add debug assert to detect calls in the wrong state

2014-05-23 15:55:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Reset count when stopped

2014-05-23 15:55:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2allocator: Return a GstFlowReturn instead of boolean in alloc

2014-05-23 15:17:27 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't leak config structure

2014-05-23 14:12:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2bufferpool: Port to bufferpool flush_start/stop method

2014-05-23 03:00:50 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: add tag mappings for _swr, _mak and _mod tags
	  swr -> Application name
	  mak -> device manufacturer
	  mod -> device model

2014-05-20 17:37:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Fix ximage leaks when buffer has more then one ximage
	  From time to time, when the image_pool list has more then 1 element
	  and I suppose at start, all but 1 pooled ximage are leaked. This is
	  due to broken algorithm in gst_ximagesink_src_ximage_get(). There was
	  also a risk of use after free for the case where the ximage size has
	  changed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=728502

2014-05-21 13:23:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.3.2 ===

2014-05-21 13:06:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* common:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect-build.stamp:
	* docs/plugins/inspect.stamp:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.3.2

2014-05-21 12:19:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2014-05-21 10:51:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* common:
	  Automatic update of common submodule
	  From 211fa5f to 1f5d3c3

2014-05-20 08:23:06 +0200  Edward Hervey <edward@collabora.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Don't dereference NULL variable
	  CID #1139838

2014-05-20 08:20:42 +0200  Edward Hervey <edward@collabora.com>

	* ext/vpx/gstvp9enc.c:
	  vp9enc: Don't dereference NULL checks
	  CID #1197703

2014-05-19 11:26:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Explicitly cast enum "subtype" to its "supertype"
	  gstv4l2bufferpool.c:608:18: error: implicit conversion from enumeration type
	  'enum _GstV4l2BufferPoolAcquireFlags' to different enumeration type
	  'GstBufferPoolAcquireFlags' [-Werror,-Wenum-conversion]
	  params.flags = GST_V4L2_POOL_ACQUIRE_FLAG_RESURECT;
	  ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

2014-05-19 11:24:06 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/goom/tentacle3d.c:
	  goom: Use fabs() instead of abs() to calculate the floating point absolute value
	  tentacle3d.c:268:7: error: using integer absolute value function 'abs' when
	  argument is of floating point type [-Werror,-Wabsolute-value]
	  if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot + 2.0 * G_PI))) {
	  ^

2014-05-19 11:21:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/debugutils/tests.c:
	  debugutils: Properly calculate the difference with unsigned types
	  tests.c:161:16: error: taking the absolute value of unsigned type
	  'unsigned long' has no effect [-Werror,-Wabsolute-value]
	  t->diff += labs (GST_BUFFER_TIMESTAMP (buffer) - t->expected);

2014-05-16 17:46:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Handle flush while in start_streaming
	  We need to handle the case where a flush occure while the streaming
	  thread is being brought up. In this case, the flushing state of the poll
	  object is cleared. To solve this, we simply set the capture poll to flushing
	  again, this way we know the thread will exit. The decoder streamlock
	  is used to synchronize with handle frame.

2014-05-16 16:44:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Don't trace twice the same message

2014-05-15 11:25:50 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: always use a random ssrc for the internal session
	  Use a random SSRC different than 0 for the internal session SSRC.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730212

2014-05-16 16:52:25 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: update last_activity when sending RTP
	  Also update last_activity when doing something with the internal
	  source to make sure don't timeout early.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=730217

2014-05-15 18:08:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Cleanup M2M properties
	  M2M devices were sharing the same properties as src and sink. Most of
	  these made no sense. This patch reduces the number of propeties and
	  makes io-mode clearer by having capture-io-mode and output-io-mode. This
	  also accidently fixed a bug in gstv4l2transform io-mode code, where the
	  capture io-mode could not be set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=729591

2014-05-15 17:39:39 +0200  Benjamin Gaignard <benjamin.gaignard@linaro.org>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Update pool limit with hardware requiremenst
	  If the driver need more buffers than requested by the config,
	  update the pool min/max values. The minimum value for the pool
	  could be provided either by the driver or by the pool. This is
	  best effort for drivers that don't support
	  CID V4L2_CID_MIN_BUFFERS_FOR_CAPTURE.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730200

2014-05-15 10:44:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Handle start_streaming error
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-15 10:39:40 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Print the flow return causing the loop to leave
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-15 10:31:40 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Don't lock the decoder when stopping task
	  That src pad task may need to take the lock when being pulled
	  down. takeing that lock can lead to a deadlock.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-14 17:18:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Don't leak pool if activation failed
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-14 17:18:35 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Split flush in start/stop_streaming
	  This allow calling start streaming later for capture device. Currently it breaks
	  in dmabuf-import because downstream is holding a buffer that will only be
	  released after stream-start.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-14 15:12:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Flush buffer pools on flush stop
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-14 13:28:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix use of atomic active marker
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-14 13:05:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Don't deactivate otherpool
	  We should not stop the otherpool unless we also stop our own
	  pool, otherwise it will never get restarted.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-14 12:33:58 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Also update num_buffers for import cases
	  https://bugzilla.gnome.org/show_bug.cgi?id=730207

2014-05-14 13:42:25 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: update rtp encoder/decoder docs
	  Use %u in RTP encoder/decoder pads to match other rtpbin pads.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730146

2013-12-27 11:55:18 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtpsession.c:
	  tests/check: rtpsession: test internal sources timing out

2013-12-26 17:30:42 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: remove unused if branch
	  1) sources that have sent BYE in the past cannot be senders, since
	  they would have timed out to being receivers in the meantime...
	  2) sources that have sent BYE are now being removed earlier inside
	  this function

2013-12-26 17:29:42 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: cleanup sources that have sent BYE

2013-12-26 17:24:51 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: unify nested if clauses

2013-12-26 17:21:44 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: timeout internal sources that are inactive for a long time and send BYE

2014-05-13 12:25:04 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: don't stop looping if event found in the queue
	  If we are inserting a packet into the jitter queue we need to keep
	  looping through the items until the right position is found. Currently,
	  the code stops as soon as an event is found in the queue.
	  Regarding events, we should only move packets before an event if there
	  is another packet before the event that has a larger seqnum.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730078

2014-04-17 13:04:00 +0000  Adrien SCH <adrien.schwartzentruber@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix the memory leak of language attribute
	  https://bugzilla.gnome.org/show_bug.cgi?id=728418

2014-05-13 13:44:20 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix regression in offset extrapolation
	  When extrapolating the offset, we need to use the extrapolate
	  stride rather then the base stride. This should fix support for format
	  with more then two planes (I420, Y42B, etc).

2014-05-12 18:03:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Use default VideoInfo for frame operation
	  When doing frame operation, we need to use the default VideoInfo
	  and let the frame API read the video meta in order to get the stride
	  and offset right. Currently we where using the specialized VideoInfo
	  which reflects what the HW is setup to.

2014-05-12 17:23:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: minor GValue handling optimisation in probing code

2014-05-12 17:20:14 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: avoid lists with one single framerate in probed caps
	  Simplify framerate field if possible, so we don't end up with
	  e.g. framerate = (fraction) { 30/1 }. Maybe the helper function
	  should be moved to core, but we can do this later.

2014-05-12 16:56:35 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix leak of palette_data in error cases
	  CID #1212151

2014-05-12 16:53:32 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Free node_header in error cases
	  CID #1212134

2014-05-12 13:46:01 +0200  Edward Hervey <edward@collabora.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't use WARNING for not-linked flow return
	  Pollutes debug logs for no reason. It's only an error if all pads
	  return not-linked

2014-05-12 13:45:06 +0200  Edward Hervey <edward@collabora.com>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Skip unknown tags in push-mode
	  We add a new mode (SKIP) in push-mode to skip tags that we don't known about
	  Partially fixes https://bugzilla.gnome.org/show_bug.cgi?id=670712

2014-05-10 09:14:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Add support for variable block size files and remove dead code
	  This dead code wasn't used since the 1.0 port and would need to
	  be modified heavily for variable block size support.
	  https://bugzilla.gnome.org/show_bug.cgi?id=729894

2014-05-09 12:14:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Fix NULL check copy paste error
	  CID 1212129

2014-05-09 12:11:54 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Fix potential deadlock due to missing break
	  CID 1212131

2014-05-09 18:01:28 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: increment accepted packets after loss
	  When we detect a lost packet, expect packets with higher
	  seqnum on the input.
	  Also update the unit test.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729524

2014-05-04 11:12:54 -0600  Jason Litzinger <jlitzingerdev@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  Add new test case.

2014-05-09 16:14:21 +0200  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/shapewipe.c:
	  shapewipe: no need to activate pads
	  Activation will happen in the state change

2014-05-09 12:10:04 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't leak config structure
	  this fixes a leak of the config structure and take care of making sure
	  caps can't reach ref 0 before we are done doing our check.
	  CID 1212144

2014-05-09 12:08:11 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Remove uneeded cast for code clarity

2014-05-09 11:56:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2pool: Fix leak of config structure in error case
	  CIDs 1212167 and  1212167

2014-05-09 11:51:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix use of unitilized pool pointer
	  CID #1212173

2014-05-09 16:48:58 +0200  Eric Trousset <etrousset@awox.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't respond to a position query in BYTE format with a TIME position
	  https://bugzilla.gnome.org/show_bug.cgi?id=729553

2014-05-09 14:22:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: don't leak doctype string in error code path
	  CID 1212145.

2014-05-06 13:37:47 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Readback pool config if used within the baseclass

2014-05-06 12:58:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Replace miss-use of crop meta in favour of proper offset
	  This moves away from copying information and store everything inside
	  the GstVideoInfo structure. The alignement exposed by v4l2 api
	  is now handled using proper offset.

2014-05-06 12:55:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Style fix

2014-05-05 12:38:33 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Reset imported buffer size with expected size
	  This ensure that the buffer pool won't always discard buffer with these
	  memory when they are released.

2014-05-05 12:37:43 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Reset flushed group
	  This ensure that a flushed group memory are the same size as when they
	  where originally allocated / imported.

2014-05-05 12:07:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Get number of allocated buffers from allocator
	  The value of num_allocated buffer would get confused when
	  buffer are being discarded.

2014-05-05 12:06:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	  v4l2allocator: Add a method to read number of allocated group

2014-05-04 20:23:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Improve debugging

2014-05-04 19:51:48 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Ensure we don't re-enqueue buffer during flush

2014-05-04 19:13:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Initilialize debug category

2014-05-04 16:11:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Fix libv4l2 support
	  Need to include config.h, otherwise we endup directly using the
	  ioct/mmap/munmap calls and need to vall v4l2_munmap.

2014-05-01 13:04:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Set the flags on the object
	  We where not setting the probed flags on the allocator, which mean even if
	  CREATE_BUFS was supported on some driver, it would endup being ignored.

2014-04-29 16:49:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Re-enqueue buffer at stream start

2014-04-29 16:06:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: There is not group on error

2014-04-29 14:56:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Handle FLUSH_STOP event

2014-04-29 13:05:41 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2bufferpool: Acquire cannot return a buffer from another pool
	  Return a buffer from an otherpool has unwanted side effects that lead to leaks and
	  prevents deactivating the pool. Instead, we change the _process() API so it can
	  replace the internal buffer with the buffer from the downstream pool. This implied
	  moving from _fill() to _create() method in the src.

2014-04-29 13:00:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Remove unreached acquire code
	  The acquire is done in _prepare now.

2014-04-29 12:57:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Sanetize buffer refount handling
	  Buffer refcounting is a bit hard, because of the duality between CAPTURE and
	  OUTPUT mode. In the long term, we should consider having two seperate pool
	  instead of this mess. At least state should be better kept this way.

2014-04-29 12:48:04 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Add more traces

2014-04-28 08:48:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	  v4l2-allocator: Add S to REQBUFS/CREATE_BUFS enum
	  All enum that has REQBUFS and CREATE_BUFS where missing S, which was
	  confusing since they are supposed to match with associcated ioctl name. This
	  also fixes the yet unused CAN_REQUEST flag check.

2014-04-18 17:51:07 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Enabled QoS

2014-04-18 17:02:50 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Fixup USERPTR/DMABUF capture support

2014-04-18 14:45:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Improve selecton of min/max in decide allocation

2014-04-18 13:09:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Update config if meta is missing
	  Rather then hard failure, we should update the config with the meta option we
	  need and return false.

2014-04-11 17:10:11 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Add DMABUF and USERPTR importation

2014-04-17 21:45:58 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Valid FD are bigger or equal to zero

2014-04-16 17:04:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't leak downstream pool in propose_allocation
	  parse_nth_allocation_pool() give a ref on the pool, we need to unref it
	  when done.

2014-04-14 12:19:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Introduce DMABUF_IMPORT IO mode

2014-04-10 16:26:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Add dmabuf export support
	  This can be enabled sing io-mode=dmabuf. This will enabled mmap base
	  drivers to export the buffers as dmabuf.

2014-04-16 15:51:03 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2allocator.c:
	  v4l2allocator: Guaranty queued state integrety
	  Because of the buf in videobuf2, dqbuf may leave the DONE flag being,
	  which would implied that the buffer is queued. As this has been broken
	  for 4 years, simply guaranty the state flags integrity when doing
	  qbuf/dqbuf.
	  See https://patchwork.linuxtv.org/patch/23641/

2014-04-15 17:31:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Implement open/close

2014-04-15 16:43:41 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Ensure output pool is configured

2014-04-15 16:43:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2transform.h:
	  v4l2transform: Check if caps have changes before asserting
	  In set_caps, now checks if caps actually changed and succeed if they didn't
	  change.

2014-04-15 16:41:46 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Ensure pool is configured

2014-04-08 18:54:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Always set a size when deciding allocation

2014-04-08 18:20:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Improved decide allocation
	  Improve decide allocation so it properly configure both local and downstream
	  buffer pools. Also read back the pool config if it was changed to to driver
	  limitations.

2014-04-15 13:30:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Do not pre-configure the pool
	  Pre-configuring the pool is error prone, since it may hide a configuration failure and
	  endup with a pool that is not configured the way it should (e.g. no video meta, wrong
	  queue size, etc.)

2014-04-15 13:23:33 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Preserve downstream minimum even in RW

2014-04-15 13:20:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Turn cropmeta into a custom option
	  Turn crop meta into a custom option and make sure it's there is needed.

2014-04-09 12:53:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2bufferpool: Early catch short allocation
	  Catch short allocation after saving the format. This is not a catch all, but should catch
	  most of the miss-behaving drivers when doing S_FMT/G_FMT and avoid potential crash.

2014-04-04 22:46:40 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: Port to use GstV4l2Allocator

2014-04-04 22:35:48 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/v4l2_calls.h:
	  Implement V4l2 Allocator
	  This goal of this allocator is mainly to allow tracking the memory.
	  Currently, when a buffer memory has been modified, the buffer and it's
	  memory is disposed and lost until the stream is restarted.

2014-04-16 16:35:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't advertise crop meta
	  Currently we advertise crop meta, but not element handle support this meta.

2014-04-08 18:18:57 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Setup pool already send element error

2014-04-08 18:17:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Workaround decoder that set num_planes to 0 in the format
	  Some well known decoder wrongly set num_planes to 0 in their format instead of
	  one. In this case we would endup with no size when deciding buffer allocation.

2014-04-08 17:34:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Ensure size before configuring the pool

2014-04-04 22:38:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Set minimum buffers to 2
	  All the element requires at least two buffers. This is not used for RW mode.

2014-04-04 22:37:14 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Remove unused MAX_BUFFERS define

2014-04-04 22:36:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't hardcode min/max use default instead

2014-04-10 17:49:41 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Install PROP_CAPTURE_IO_MODE with right ID

2014-04-08 18:54:50 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: decide_allocation returns a boolean

2014-04-10 17:49:29 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Install PROP_CAPTURE_IO_MODE with right ID

2014-03-27 13:21:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Add propose_allocation
	  This should remove 1 copy between the decoder and the transform.

2014-03-27 13:20:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2: Move propose allocation to v4l2object

2014-03-20 17:26:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Fixup caps query

2014-03-20 15:31:22 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Setup cropping if needed

2014-03-19 17:25:16 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2.c:
	  v4l2transform: Expose BGRA and ARGB formats

2014-03-18 17:33:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Ensure output pool is activated
	  That pool may be different then the internal pool.

2014-03-16 19:11:16 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Ensure internal buffer pools actication

2014-03-16 11:36:19 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Move subinstance subclass init near other init

2014-03-15 18:56:51 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: Stop stream before closing the devices.

2014-03-15 16:53:54 +0000  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: copy metdata

2014-03-04 18:31:27 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2transform.h:
	  Implement GstV4l2Transform
	  Implement a v4l2 element that wraps HW video converters.

2014-03-27 18:41:07 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Probe for CREATE_BUFS in order to correctly set pool min/max
	  In order to correctly set the pool min/max, we need to probe for CREATE_BUFS
	  ioctl. This can be done as soon as the format has been negotiated using a
	  count of 0.

2014-03-25 15:21:03 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Move capture eos handling in _process()
	  Now that we might be copying out buffer (e.g. downstream don't support video
	  meta bug we need it) we need to move the EOS handling inside the process
	  method.

2014-03-25 10:49:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix support for planar format in 1 v4l2 mplane
	  So far we where only setting saving the first plane stride in the meta. This was
	  leading to wrong values in GstVideoMeta.

2014-03-19 17:52:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Cleanly fail if set_format is never called

2014-03-19 17:00:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: Expose RGB32 formats with and without alpha
	  As soon a the alpha component can be set, we can expose the RGB32 and BGR32
	  format as ARGB and BGRA as long we can deterministically set the alpha padding
	  value.

2014-03-18 15:49:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: Correctly check if video meta is needed
	  Correctly check if video meta is needed. In buffer pool, trust need_video_meta
	  flag in order to decide if configuration should succeed.

2014-03-18 15:45:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix tiled stride request
	  Fix stride request for tiled format and improve logging.

2014-03-18 11:53:57 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Ensure video and crop meta are enabled if needed
	  In certain cases we cannot live without video meta and/or crop meta
	  being enabled in our internal buffer pool. Ensure this is always the case,
	  regardless of having support for allocation query.

2014-03-16 18:39:32 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Ensure internal pool are activated

2014-03-16 17:01:10 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Check that pool where allocated before flushing them
	  Upon error, the pools might not have been allocated yet, hence we should not
	  try and flush them (even though we still want to make sure the processing thread
	  is fully stopped).

2014-03-16 16:55:43 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2bufferpool: Enforce activation outside of process
	  Enforce pool being activate from before calling pool process. This should
	  help catching basic errors in the usage of buffer pool.

2014-03-16 12:44:14 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: don't use own pool if downstream don't support video meta

2014-03-14 00:31:32 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Use obj->n_v4l2_planes for correct number of planes
	  Buffer pool was guessing wrongly the number of planes rather
	  then reading the value from obj->n_v4l2_planes. This was causing
	  format YU12 (I420) to fail upon check.

2014-03-07 16:39:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix handling of contiuous vs non-contiguous support
	  The complex mechanic to try and choose the right thing did not work. Instead,
	  simply probe the non-contiguous format first and then the contiguous one.
	  This is in fact very low overhead, as there is a relatively small number of
	  pixel format supported by each devices.

2014-04-15 15:07:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2videodec.h:
	  v4l2: Add initial support for alignment and cropping

2014-03-13 19:24:51 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2object: Rename setup_format() method into acquire_format()
	  The setup_format() was confusing since it does not set anything, in fact
	  it reads the setup from the driver and save it.

2014-03-13 18:21:41 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Move type declaration to the top

2014-03-12 18:07:38 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Protect NULL pool while going to READY
	  When the pipeline fails early, the pool might be unset before the processing
	  thread has run once. Add protection against that.

2014-03-12 18:01:09 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fail cleanly if pixel format is unkown or not raw video
	  Certain decoder has been found to not choose a format automatically. Running
	  v4l2videodec on these would assert. This patch will make it fail cleanly
	  instead.

2014-03-12 17:56:18 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Clear the input state pointer after unref
	  If caps are set again, we have a risk od returning from set_format with a
	  input_state pointing to dead memory. Clearing the pointer after unref fix
	  this issue.

2014-03-12 17:11:16 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: handle stop being called without flush
	  Uppon certain downstream error, stop() is called without a flush(). This mean that
	  the streaming thread may still be running even though unlock has been called.
	  Now calling flush to reset the decoder state if we are processing.

2014-03-06 18:13:14 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Default to template in caps query

2014-03-11 14:23:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Ensure processing thread has stopped when draining

2014-03-11 14:01:27 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Don't drain if processing thread is inactive

2014-05-08 09:49:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Clean up all pending operations from libsoup before unreffing our context
	  When we cancel connection attempts and similar things, there are still
	  some operations pending on our main context from the GCancellables. We
	  should let them all run before unreffing our context, otherwise we leak
	  file descriptors.
	  Unfortunately this requires libsoup 2.47.0 or newer as earlier versions
	  steal our main context from us and we can't use it for cleanup later
	  without assertions and funny crashes.
	  Based on a patch by Dmitry Shatrov <shatrov@gmail.com>.
	  https://bugzilla.gnome.org/show_bug.cgi?id=663944

2014-05-07 15:49:39 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: fix compilation of souphttpsrc test for libsoup 2.40 for real
	  https://bugzilla.gnome.org/show_bug.cgi?id=727329

2014-05-07 13:23:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: fix compilation of souphttpsrc test for libsoup 2.40
	  SOUP_CHECK_VERSION was only added in 2.41, but we only
	  depend on 2.40.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727329

2014-05-07 00:58:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: skip PICTURE headers without any image data
	  Fixes warning if the image length is 0.

2014-05-06 09:22:18 +0000  Руслан Ижбулатов <lrn1986@gmail.com>

	* configure.ac:
	  configure: use X11 detection macro from common
	  https://bugzilla.gnome.org/show_bug.cgi?id=729621

2014-04-30 11:13:12 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/rtp/README:
	  rtp/README: update pipelines to work with 1.0
	  - Use gst-libav encoders/decoders instead of gst-ffmpeg
	  - gstrtpjitterbuffer -> rtpjitterbuffer
	  - gst-launch-0.10 -> gst-launch-1.0
	  - Add 'videoconvert' element
	  - xvimagesink -> autovideosink
	  https://bugzilla.gnome.org/show_bug.cgi?id=729247

2014-05-05 14:41:05 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroska: rejig test to avoid undefined shift behavior
	  Coverity 1195121, 1195120

2014-05-05 14:33:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vpx/gstvp9enc.c:
	  vp9enc: do not dereference NULL pointer
	  Coverity 1197703

2014-05-05 14:32:06 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: ensure we don't dereference a NULL pointer
	  while working out the codec ID.
	  Coverity 1195148

2014-05-05 12:07:25 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2.c:
	  v4l2: minor fix for closing the fd
	  The fd returned by open() could theoretically be 0 as well.
	  Coverity CID 1211823.

2014-05-04 20:23:29 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* tests/check/elements/rtpaux.c:
	* tests/check/elements/rtprtx.c:
	  rtpaux/rtprtx: Make tests non-racy
	  Fix the raciness by iterating on a condition instead of using the gmainloop.
	  Don't use the EOS as the target, otherwise the retransmission of the last
	  packets are lost. Also count the retranmissions requests that are dropped.
	  Check the condition before blocking on the GCond
	  https://bugzilla.gnome.org/show_bug.cgi?id=728501

2014-05-04 22:32:54 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxreceive.h:
	  rtprtxreceive: Wait until timeout to clear association requests
	  If two streams request a retranmission for the same SSRC, ignore the second
	  one if the first oen is less than one second old, otherwise time out the first
	  one and ignore the second.

2014-05-04 18:59:33 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/rtpmanager/gstrtpmux.c:
	* tests/check/elements/rtpmux.c:
	  rtpmux: Always let upstream chose the ssrc if it wishes

2014-05-04 13:37:46 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: avoid stall by corrupted seqnum accounting

2014-05-04 01:14:33 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* ext/pulse/pulsedevicemonitor.c:
	* ext/pulse/pulsedevicemonitor.h:
	  pulsedevicemonitor: Index are per facility, not global
	  So need to keep the type of device in the device object

2014-05-04 01:13:24 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* ext/pulse/pulsedevicemonitor.c:
	  pulsedevicemonitor: pa_subscription_event_t are enums, not flags
	  Coverity 1195132

2014-05-02 22:42:54 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2devicemonitor.c:
	  v4l2devicemonitor: Port to use GstV4l2Iterator
	  https://bugzilla.gnome.org/show_bug.cgi?id=727925

2014-05-02 21:38:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videodec.h:
	  v4l2: Use single pass iterator for M2M probe
	  Instead of having each M2M class do their own probing, use the
	  GstV4l2Iterator and probe all devices in a single pass.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727925

2014-05-02 16:55:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/v4l2-utils.c:
	* sys/v4l2/v4l2-utils.h:
	  v4l2: Add a common device enumerator
	  This will allow removing code duplication (hence bugs duplication).
	  https://bugzilla.gnome.org/show_bug.cgi?id=727925

2014-03-16 11:38:07 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videodec.h:
	  v4l2videodec: Simplify sub-instanciation mechanism
	  Simplify sub-instanciation by defining an absract type and using subtype
	  class and instance init callback. This also fixes a bug where the template
	  pads get initialized too late.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727925

2014-05-02 18:18:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2.c:
	  v4l2: Cleanup plugin registration
	  There is no plan to introduce special sources for jpeg, te v4l2src works fine
	  for this.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727925

2014-05-03 18:30:20 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* tests/check/elements/rtpcollision.c:
	  rtpsession: Keep local conflicting addresses in the session
	  As we now replace the local RTPSource on a conflict, it's no longer possible
	  to keep local conflicts in the RTPSource, so they instead need to be kept
	  in the RTPSession.
	  Also fix the rtpcollision test to generate multiple collisions instead of
	  one by change the address, as otherwise we detected that it was a single one.

2014-05-03 20:48:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

=== release 1.3.1 ===

2014-05-03 18:02:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* gst/audiofx/audiopanoramaorc-dist.c:
	* gst/deinterlace/tvtime-dist.c:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videomixer/videomixerorc-dist.c:
	* win32/common/config.h:
	  Release 1.3.1

2014-05-03 18:02:01 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2014-05-03 17:22:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/hu.po:
	* po/id.po:
	* po/lv.po:
	* po/nb.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sr.po:
	* po/zh_CN.po:
	  po: Update translations

2014-05-03 11:43:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/shapewipe.c:
	  shapewipe: Send initial events after setting the elements to PLAYING
	  Otherwise we send them too early, and setting the elements to PLAYING
	  afterwards will drop all the events again.

2014-05-03 10:15:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* common:
	  Automatic update of common submodule
	  From bcb1518 to 211fa5f

2014-05-02 17:12:29 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Set segment position to the stop position of the buffer

2014-05-02 17:10:18 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Properly report errors before stopping the srcpad task

2014-05-02 17:02:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Error out if we have no caps yet

2014-05-02 14:49:27 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: avoid dividing by a 0 blockalign
	  This can be 0. In that case, do not try to cut off the last few
	  bytes from the last buffer.
	  Coverity 1146971

2014-05-02 14:25:01 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: do not use uinitialized clut on error
	  If we're missing part of the clut, do not try to use it. It seems
	  very likely the break was meant to break out of the switch rather
	  than from the loop.
	  Coverity 1139878

2014-05-02 14:18:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flx/gstflxdec.c:
	  flxdec: fix integer overflow
	  Coverity 1139859

2014-05-02 14:09:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpqdmdepay.c:
	  rtpqdmdepay: remove pointless check
	  Besides, the pointer was dereferenced earlier anyway.
	  Coverity 1139853

2014-05-02 14:06:25 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove duplicate test
	  item was dereference previously.
	  While there, reorder some test for faster early out.
	  Coverity 1139844

2014-05-02 14:02:52 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: guard against NULL pointer dereference
	  Coverity 1139838

2014-05-02 13:59:07 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix theoretical integer overflow
	  This code isn't actually used at the moment, unsure if I should
	  just remove it or not...
	  Coverity 1139811

2014-05-02 13:33:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroska: blindly fix writing variable length negative values
	  Spotted while fixing something else in the area.
	  Nothing calls this with a negative value.

2014-05-02 13:29:33 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroska: do not lose the top bits when writing a > 32 bit value
	  Coverity 1139806

2014-05-02 12:10:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: add missing break in switch
	  Coverity 1139755

2014-05-02 11:39:39 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	  matroska: do not try to call gst_pad_query_default on a NULL pad
	  gst_matroska_parse_query can be called explicitely with a NULL pad.
	  If we reach this point with a NULL pad, fail the query.
	  Coverity 1139715

2014-05-02 11:28:01 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	  matroska: do not return GST_FLOW_OK if we did not get a buffer
	  Coverity 1139714 (which will likely come back in another guise,
	  as the _read_init call can have a failing _map)

2014-05-02 11:20:33 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroska: catch failure to map buffer
	  Avoids dereferencing NULL.
	  Coverity 1139712

2014-05-02 10:52:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: refuse caps with invalid framerate
	  Coverity 1139701

2014-05-02 10:21:09 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: handle 0 size packets without dividing by 0
	  Coverity 1139691

2014-05-02 09:49:32 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: guard against invalid frame size to avoid division by 0
	  Coverity 1139690

2014-05-02 09:49:17 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: trivial typo fix

2014-05-02 09:43:54 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: remove dead code
	  fpp can never equal 0 here, or the loop would not execute at all.
	  Zero fpp was possible before as the loop condition was allowing
	  it specifically, but no more.
	  Coverity 1139681

2014-05-02 09:41:19 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/oss4/oss4-property-probe.c:
	  oss4: remove dead mixer code
	  This was partly removed in the port to 0.11. If still needed,
	  it's still there in the history.
	  Coverity 1139687

2014-05-02 09:33:51 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/oss4/oss4-property-probe.c:
	  oss4: fix a missing unlock and a return-only-when-assertions-enabled
	  Spotted on the side while looking at another issue.

2014-03-07 17:31:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Correctly map RGB32 format
	  In v4l2 specification, RGB32 has the alpha, or pading, first, not last.
	  See http://linuxtv.org/downloads/v4l-dvb-apis/packed-rgb.html .
	  https://bugzilla.gnome.org/show_bug.cgi?id=540941

2014-04-30 18:06:40 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: remove dead code
	  For 8 bit width, we always have depth==gdepth==width==8.
	  Coverity 1139678

2014-04-30 17:48:53 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: remove dead code
	  A stricer check is already done earlier, and integer overflows
	  do not seem possible here.
	  Coverity 1139675

2014-04-30 14:50:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: guard against pathological "no space" condition
	  Even if one woul hope one pixel can fit in a MTU, ensure we do not
	  overwrite a buffer if this is not the case.
	  Spotted while looking at Coverity 1208786

2014-04-30 11:52:10 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	  rtpjpegdepay: sanity check for NULL qtable
	  Can happen (at least in crafted stream)
	  Coverity 1208778

2014-04-30 01:08:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: pass on tags from upstream if there are any
	  Don't just ignore upstream tags from e.g. an ID3 tag before
	  the .wav data, pass them on downstream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=729223

2014-04-29 16:26:53 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: optimize timer update
	  When we are not doing retransmission, we just need to find the current
	  seqnum so we can stop when we found it.

2014-04-29 16:21:44 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpjitterbuffer.h:
	  rtpjitterbuffer: small optimizations
	  Small optimizations where we can.
	  Add some more debug.

2014-04-29 16:16:17 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: signal when next_seqnum changed
	  Signal the pushing thread when the next_seqnum changed and we might be
	  able to push a buffer now.

2014-04-29 16:12:29 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: only signal event when head changed
	  After adding a buffer, only signal the pushing thread when the head
	  buffer changed or else we cause a useless wakeup.

2014-04-29 15:29:31 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: rework packet insert
	  Rework the packet queue so that the most common action (insert a packet
	  at the tail of the queue) goes very fast.
	  Report if a packet was inserted at the head instead of the tail so that
	  we can know when to retry _pop or _peek.

2014-04-29 16:38:55 +1000  Matthew Waters <ystreet00@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	* tests/examples/gtk/gstgtk.c:
	  gl/examples: move to -bad
	  - fix all the compiler errors
	  - give them their own gl directory

2014-04-28 14:41:10 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtpvraw: use plane pointers when needed
	  Pack/unpack planar formats to/from the first plane.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729058

2014-04-28 09:47:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Remember if a redirect is permanent or not and store it in the query

2014-04-27 21:57:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/goom/config_param.c:
	  goom: Remove french comment saying to prefix functions
	  All non-static function in this file are already prefixed with goom_.

2014-04-28 00:20:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/goom/filters.c:
	  goom: fix compilation on ios-arm7-10.9 and osx-x86_64
	  uint is not a standard type, and the rest of the code uses
	  Uint which is locally typedefed to unsigned int.
	  https://bugzilla.gnome.org/show_bug.cgi?id=729067

2014-04-27 18:29:11 -0400  Luis de Bethencourt <luis@debethencourt.com>

	* gst/goom/filters.c:
	  goom: fix undefined behaviour of left-shift
	  Don't left-shift into the sign bit, the result is undefined and potentially
	  an overflow could flip the sign.

2014-04-26 20:51:36 -0400  Luis de Bethencourt <luis@debethencourt.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check return from qt_demux_video_caps
	  Now qtdemux_video_caps() can return NULL. We need to check this return before
	  using it's value.
	  https://bugzilla.gnome.org/show_bug.cgi?id=728987

2014-04-26 23:35:17 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/speex/gstspeexdec.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/spectrum/gstspectrum.c:
	* gst/udp/gstudpsrc.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/wavparse/gstwavparse.c:
	* sys/osxaudio/gstosxaudiosink.c:
	  docs: remove outdated and pointless 'Last reviewed' lines from docs
	  They are very confusing for people, and more often than not
	  also just not very accurate. Seeing 'last reviewed: 2005' in
	  your docs is not very confidence-inspiring. Let's just remove
	  those comments.

2014-04-25 17:58:42 -0400  Luis de Bethencourt <luis@debethencourt.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: initialize caps pointer to null
	  Make sure the caps pointer returns initialized when using it in
	  qtdemux_parse_tree ().
	  https://bugzilla.gnome.org/show_bug.cgi?id=728987

2014-04-22 17:07:38 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Clear last_pt on flush-stop.
	  Otherwise, we don't recheck the buffer caps for clock-rate
	  properly on the next chain.

2014-04-22 17:29:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix compiler warning
	  gstdeinterlace.c: In function 'gst_deinterlace_output_frame':
	  gstdeinterlace.c:1537:57: error: 'pattern.length' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  This actually is always initialized before it is used there, but
	  let's just silence gcc here.

2014-04-21 15:58:45 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: fix buffer list drop check
	  While porting to 0.11, the check was mistakenly made constant,
	  instead of testing for the return value of process_buffer_locked.
	  Coverity 1139663

2014-04-21 13:44:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  matroska: fix content encoding scope validity check
	  It's 3 bits, and http://matroska.org/technical/specs/index.html
	  says it can't be 0.
	  Coverity 1139660

2014-04-21 13:34:37 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix PAR fraction sanity check
	  It was checking par_num twice, and never par_denum.
	  Coverity 1139634

2014-04-21 13:32:40 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  multiidpsink: warn when setsockopt fails
	  This doesn't seem to be fatal, but it's good to let the user know
	  in the logs.
	  Coverity 1139630

2014-04-21 13:27:24 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/interleave/deinterleave.c:
	  interlace: catch failure to create audio info from caps
	  Coverity 1139627, 1139628

2014-03-13 09:37:48 +0100  Göran Jönsson <goranjn@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  gstrtph264pay: Reset sps pps variable when state change.
	  Reset last_spspps and sps/pps arrays  when state transition
	  GST_STATE_CHANGE_PAUSED_TO_READY.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726015

2014-04-18 11:11:14 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: improve EOS handling
	  Make a new method to disable the jitterbuffer buffering.
	  Rework the update_estimated_eos() method. Calculate how much time
	  there is left to play. If we have less than the delay of the
	  jitterbuffer, we disabled buffering because we might never be able to
	  fill the complete jitterbuffer again.
	  If we receive an EOS event, disable buffering. We will drain the
	  buffer and eventually push the EOS event out.
	  When we reach the estimated NPT timeout and we didn't receive an EOS
	  event, make one and queue it so that it can be pushed.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017

2014-04-18 10:21:27 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: send reconfigure when internal-ssrc changes
	  When the internal-ssrc property changes, we want to send a reconfigure
	  upstream to make payloaders use the new suggested ssrc.
	  Using the internal-ssrc property to change the SSRC of a stream is not a
	  good idea and doesn't work when there are multiple senders, we want to
	  set the SSRC directly on the payloaders. Therefore, deprecate this
	  property.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725361

2014-04-18 04:23:26 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: assume a full buffer when eos
	  Rework the logic to make buffering messages a little, make sure we
	  don't make the same message multiple times.
	  Consider the buffer full when EOS was received.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017

2014-04-17 18:07:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtprtx.c:
	  rtprtx: Don't forget to unmap rtp buffer in the test

2014-04-17 17:58:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: Require clock-rate in the caps and handle no ssrc in the caps properly

2014-04-17 17:43:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtprtx.c:
	  rtprtx: Provide an ssrc in the test
	  And increase timeout to allow all tests to run in valgrind.

2014-04-17 17:33:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtpsession.c:
	  rtpsession: Fix memory leaks in test

2014-04-17 17:26:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix hundreds of memory leaks in the test

2014-04-17 17:00:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Unref clock id when waiting for the clock is interrupted

2014-04-17 16:39:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtpcollision.c:
	  rtpcollision: Fix memory leaks in unit test

2014-04-16 21:40:45 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: name collectpads object based on videomixer name
	  Makes it easier to track things in debug logs when there
	  are multiple mixers and muxers.

2014-04-16 21:37:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: better logging of incoming events
	  The pad and parent names are already logged as part of logging
	  the object. Instead log the full event details.

2014-04-16 19:03:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/videomixer.c:
	  videomixer: Fix memory leak in unit test

2014-04-16 18:49:43 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/level/gstlevel.c:
	  level: Use the correct number of samples to iterate over the input array
	  Fixes invalid memory accesses and accesses to uninitialised data.

2014-04-16 18:00:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/icydemux/gsticydemux.c:
	  icydemux: Unref dropped events

2014-04-16 17:29:30 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	  matroska: fix check for amount of data to read
	  History shows length==0 should set data to NULL and return,
	  so we do that too instead of trying to read nothing.
	  Coverity 206205

2014-04-16 17:25:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix sign comparison
	  history_count is unsigned, so the whole comparison will be made
	  as unsigned, and fail to reject what it was meant to.
	  Coverity 206204

2014-04-16 17:04:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: remove dead code
	  sub may not be NULL in this switch, there is a bail out just
	  before it if so.
	  Coverity 206098

2014-04-16 16:59:43 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: remove dead code
	  The block_size == 0 was shortcut earlier, and the variable is not
	  modified in the meantime.
	  Coverity 206097

2014-04-16 16:56:54 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videomixer/videoconvert.c:
	  videomixer: remove dead code
	  While it seems to keep a compile time selection, I traced it
	  to some code copied from videoconvert, where it was removed,
	  with the following comment:
	  Also remove the high-quality I420 to BGRA fast-path as it needs
	  the same fix, which causes an additional instruction, which causes
	  orc to emit more than 96 variables, which then just crashes.
	  This can only be fixed in orc by breaking ABI and allowing more
	  variables.
	  Thus, I remove it here as well.
	  Coverity 206064

2014-04-16 16:50:30 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  isomp4: fix incorrect masking for multiple tags
	  Coverity 206058

2014-04-16 16:45:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/atoms.c:
	  isomp4: fix wrong atom flags set when adding samples
	  Coverity 206057

2014-04-16 16:40:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofx: fix comparison of delta time to a threshold
	  Coverity 206055

2014-04-16 16:32:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: do not rely on call failure keeping return data unmodified
	  This is clearer this way too.
	  Coverity 206029

2014-04-16 16:28:49 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/atomsrecovery.c:
	  isomp4: catch fseek error
	  Coverity 206028

2014-04-16 16:25:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/atoms.c:
	  isomp4: report failures to caller
	  Coverity 206027

2014-04-16 18:05:46 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refuse serialied query when buffering
	  When we are buffering, we can't block and wait for the serialized query
	  to complete because the jitterbuffer will not try to forward the query
	  while buffering. Instead, just refuse the query.

2014-04-16 16:51:15 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: don't free the serialized query
	  We should never free a serialized query in the queue, it is the upstream
	  caller that will free it.

2014-04-16 17:35:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/aacparse.c:
	  aacparse: Fix memory leak in the test

2014-04-16 17:33:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Create hashtable only when we actually use it
	  In error cases we previously returned without freeing it.

2014-04-16 17:30:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Chain up to the parent class' dispose function

2014-04-16 17:23:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Initialise ioctl struct with zeroes before passing it to ioctl()

2014-04-16 13:47:43 +0200  Marc Leeman <marc.leeman@gmail.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: correct LOG msg for -1
	  Signed-off-by: Marc Leeman <marc.leeman@gmail.com>

2014-04-15 21:36:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/interleave/interleave.c:
	  interleave: Fix negotiation to work at all again
	  The caps query handling function for the sinkpads was called for
	  the srcpad, and the sinkpads had none. This commit moves it to the
	  right pad, but nonetheless the negotiation still looks wrong.
	  This makes the test pass again after the recent coverity fix
	  and also allows interleave to work again, but someone should
	  really review the negotiation code and fix it.

2014-04-13 09:03:41 +0200  Edward Hervey <edward@collabora.com>

	* sys/oss4/oss4-audio.c:
	  oss4: Maximum number of channels support is 8
	  Avoids doing potential overwrites in ch_layout (which only has 8
	  fields).
	  CID #1139826

2014-04-12 22:16:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: Set rank to MARGINAL
	  If available we prefer using glimagesink over osxvideosink. It supports
	  more formats and in general has more features than osxvideosink.

2014-04-11 18:19:49 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: only guess AU boundaries when aren't indicated by marker
	  The marker bit isn't mandatory and we had in place code to guess AU
	  boundaries by detecting a new picture start. This guessing code
	  didn't work with interlaced content that has proper marker bits
	  to indicate the AU boundaries. It was leaking the first field buffer
	  and producing a corrupted output.
	  fixes: https://bugzilla.gnome.org/show_bug.cgi?id=728041

2014-04-10 10:38:19 -0300  Rafał Mużyło <galtgendo@o2.pl>

	* ext/libpng/gstpngdec.c:
	  pngdec: enable libpng interlaced picture handling
	  Makes libpng deinterlace Adam7 interlaced pictures
	  by default. It is the only interlaced format available
	  and if the picture isn't interlaced the code should behave
	  as before.
	  https://bugzilla.gnome.org/show_bug.cgi?id=726161

2014-04-11 13:27:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Only keep-alive the connection in stop() if we have finished all previous messages
	  After cancelling a request we need to create a new connection.

2014-04-11 11:54:12 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/dv/gstdvdec.c:
	  dvdec: Don't set bogus timestamp/duration
	  This will happen if we have an incoming stream with a non-TIME segment
	  Could be improved later to figure out proper pts/duration.
	  CID #1199702
	  CID #1199703

2014-04-11 11:53:42 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/dv/gstdvdec.c:
	  dvdec: Properly refuse incoming stream without framerate
	  The return value wasn't properly propagated back if the caps
	  didn't contain a framerate

2014-04-10 16:35:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Also retry on unexpected network failures

2014-04-10 15:45:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: New property to specify the maximum number of retries before we give up

2014-03-13 10:56:11 +0100  Alexander Zallesov <zallesov@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Change default timeout to 15 seconds
	  If nothing happens after 15 seconds, chances are good that
	  our connection will never will work. Stop after 15 seconds
	  instead of waiting until the system's default timeout, which
	  can be > 1 minute.

2014-04-09 17:30:54 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: replace duplicated variable when parsing trex atom
	  https://bugzilla.gnome.org/show_bug.cgi?id=727878

2014-04-09 10:56:29 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Use GST_FLOW_FLUSHING when flushing, not GST_FLOW_EOS
	  ... and reset it properly after flushing is done. Fixes playback
	  in many cases when buffering is used.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727821

2014-04-09 08:58:04 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Properly return stream flags when parsing trex atom
	  https://bugzilla.gnome.org/show_bug.cgi?id=727867

2014-03-19 19:18:11 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: use the video frame API instead of the video meta API
	  https://bugzilla.gnome.org/show_bug.cgi?id=726738

2014-03-19 18:47:39 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: advertize video meta API support
	  https://bugzilla.gnome.org/show_bug.cgi?id=726737

2014-04-08 11:31:06 +0200  Edward Hervey <edward@collabora.com>

	* gst/interleave/interleave.c:
	  interleave: Add missing break in switch statement
	  The caps query is handled entirely already before.
	  CID #1139757

2014-04-06 18:03:11 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: use SoupKnownStatusCode if needed
	  From libsoup docs:
	  Prior to 2.44 SoupStatus was called SoupKnownStatusCode,
	  but the individual values have always had the names they
	  have now.
	  Fixes:
	  https://bugzilla.gnome.org/show_bug.cgi?id=727329

2014-04-07 12:58:23 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: use frames, not bytes, for position query in VBR streams
	  Coverity 1139648

2014-04-07 12:42:14 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/smpte/gstsmpte.c:
	  smpte: fix copy/paste error causing unmap on wrong buffer
	  Coverity 1139647

2014-04-07 12:16:17 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: guard against finding no suitable pattern
	  The code handles a -1 pattern index, and it seems plausible
	  that a pattern might be found later, so it seems best to not
	  send an element error here.
	  Coverity 1139766

2014-04-04 17:38:14 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: update for new MIKEY API

2014-04-03 17:40:01 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: send sender SSRC in the MIKEY message
	  Allocate a new SSRC for our RTCP messages back to the server and set
	  this in the MIKEY message.

2014-04-03 17:39:30 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: make random number for the CSB
	  As recommended in the RFC

2014-03-26 12:10:44 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't put spaces in keymgmt header

2014-03-25 17:47:49 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: create and send the RTCP encryption key
	  Create and make a key for encrypting the RTCP packets back to the server
	  and wrap this in a MIKEY message that we send as a header in the SETUP
	  request.

2014-04-03 12:18:39 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: free the srtpdec element

2014-04-03 12:16:25 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: cleanup stream_free function
	  There is no reason to NULL all fields, we will free the stream anyway.

2014-04-03 12:07:31 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: demote warning to debug
	  For TCP, it is normal that we don't have timestamps so don't WARN on
	  it.

2014-03-29 19:13:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Fix support for caps without width, height, framerate or format
	  For format like mpegts, width and height is rarely in the negotiated caps. This
	  patch fixes failure when setting format, and prevent introducing width, height,
	  framerate and format to the caps when fixating.
	  https://bugzilla.gnome.org/show_bug.cgi?id=725860

2014-03-31 18:34:13 +0200  Thibault Saunier <tsaunier@gnome.org>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Always set PTS=DTS on raw video streams

2014-03-31 18:31:22 +0200  Thibault Saunier <tsaunier@gnome.org>

	* gst/avi/gstavidemux.c:
	  avidemux: Always set pixel-aspect-ratio on raw video streams
	  That field is mandatory in caps and if it is not present in the
	  AVI container, it means square pixels thus 1/1.

2014-03-30 00:35:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: add mapping for Opus audio
	  Might want to consider adding channels/rate
	  requirement to template caps, but requires
	  fixing up of encoder and parser first.

2014-03-30 00:31:11 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroska-demux: add mapping for Opus audio codec
	  https://bugzilla.gnome.org/show_bug.cgi?id=727305

2014-03-29 17:21:17 -0400  William Manley <will@williammanley.net>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: Fix support for mpegts streams
	  It seems that GStreamer's mpegts elements (tsdemux, tsparse) require caps
	  `video/mpegts,systemstream=true`.  As far as I can see the significance
	  of systemstream is to indicate that this is a container format rather than
	  an elementary stream.  As this is the case (and I can't understand how it
	  could not be the case with mpegts) I add systemstream=true to v4l2src's
	  caps.
	  This allows v4l2src to be linked with tsdemux for playback from my
	  Hauppauge HD-PVR with the pipeline:
	  v4l2src ! queue ! tsdemux ! video/x-h264 ! decodebin ! xvimagesink
	  In combination with the next commit this fixes using Hauppauge HD-PVR with
	  GStreamer 1.0+.

2014-01-14 14:48:42 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: attempt to fix infinite (for small version of infinite) loop

2014-03-29 13:20:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpmanager: copy sticky events when exposing pads in more places
	  https://bugzilla.gnome.org/show_bug.cgi?id=724712

2014-03-28 20:11:36 +0100  Rico Tzschichholz <ricotz@ubuntu.com>

	* sys/v4l2/Makefile.am:
	  v4l2: fix distcheck
	  Make sure ext/*.h are dist'ed

2014-03-27 19:51:50 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: only extrapolate alpha mask for 32-bit depth
	  Instead of passing bogus alpha mask values when there's no alpha.
	  https://bugzilla.gnome.org/show_bug.cgi?id=726833

2014-03-21 13:03:17 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Add ARGB/BGRA support

2014-03-20 15:28:26 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtp/gstrtpjpegpay.c:
	  jpegpay: consider header len when calculating payload len
	  Fixed https://bugzilla.gnome.org/show_bug.cgi?id=726777

2014-03-26 08:03:22 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: All frames are sync points

2014-03-26 08:02:43 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/libpng/gstpngdec.c:
	  pngdec: All frames are sync points

2014-03-22 17:07:46 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: segment closing not needed in 1.x
	  ... as sender should keep track of segment base accumulation.
	  Rather, it may have some adverse effects as a spurious segment event,
	  e.g. in collectpads.

2014-03-22 17:05:17 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: early sending pending codec-data for all streams
	  ... at least before syncing across all streams might cause some gap
	  activity on any of those streams, notably sparse streams.
	  See also #712134

2014-03-22 17:01:27 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: handle both sticky and non-sticky custom event

2014-03-25 11:44:27 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: only expose streams on dataflow
	  Only probe on buffers, we don't want to expose the streams on events.

2014-03-25 11:36:40 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: copy sticky events to ghostpad
	  When we expose internal pads as ghostpads, first copy the sticky events
	  so that we have the caps and segment etc.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724712

2014-03-24 14:25:43 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: srtp handling

2014-03-25 10:23:00 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: set SSRC on caps if known

2014-03-24 16:58:25 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: put caps on udpsrc instead of using the signals
	  Try to avoid using the request-pt-map to get caps but set them directly
	  on the udpsrc element. That way, the caps get nicely transformed as they
	  pass through the different elements in the rtpbin, including the AUX and
	  decoder/encoder elements.

2014-03-24 15:35:09 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: use profile to set rtcp caps
	  Use the negotiated profile to set x-rtcp or x-srtcp caps

2014-03-24 15:34:26 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: set udpsrc to READY
	  READY is enough to allocate ports now

2014-03-24 14:25:28 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: improve caps handling
	  Protect caps with the lock.
	  Don't push the caps event from the set_property function but mark the
	  pad for reconfiguration so that it will renegotiate and push the new
	  caps event in the streaming thread.

2014-03-24 15:15:34 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: open/close socket in NULL<->READY state
	  We should open the socket when going to NULL<->READY and not in the
	  start/stop vemthod, which is called in READY<->PAUSED. This makes it
	  possible to allocate a socket without going to PAUSED (and starting the
	  negotiation).

2014-03-24 14:35:01 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: free caps in ptmap array
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726696

2014-03-20 11:12:51 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle NULL rtpmap and parse error better

2014-03-18 00:08:50 +0000  Руслан Ижбулатов <lrn1986@gmail.com>

	* tests/examples/gtk/gstgtk.c:
	  gl: fix the use of always-defined macros
	  After 2a0f0399ae226089c2ba07b1b904741b856f37af GST_GL_* macros are always
	  defined to 0 or 1. Don't use #ifdef ... or #if defined() on them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=726591

2014-03-16 23:46:22 -0400  Olivier Crête <tester@tester.ca>

	* configure.ac:
	  configure: Don't check for gudev if video4linux2 is not present

2014-03-16 23:19:55 -0400  Olivier Crête <tester@tester.ca>

	* configure.ac:
	  configure: Don't fail if gudev is not present
	  PKG_CHECK_MODULES has the bad habit of failing the build if it doesn't
	  get what it wants, prevent that.

2012-11-02 13:33:13 +0100  Olivier Crête <olivier.crete@collabora.com>

	* configure.ac:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2devicemonitor.c:
	* sys/v4l2/gstv4l2devicemonitor.h:
	  v4l2: Implement GstDeviceMonitor subclass
	  https://bugzilla.gnome.org/show_bug.cgi?id=678402

2013-08-12 11:49:21 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/Makefile.am:
	* ext/pulse/plugin.c:
	* ext/pulse/pulsedevicemonitor.c:
	* ext/pulse/pulsedevicemonitor.h:
	  pulse: Add device monitors
	  https://bugzilla.gnome.org/show_bug.cgi?id=678402

2014-03-16 19:24:26 -0400  Olivier Crête <tester@tester.ca>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Remove GstPropertyProbe leftovers

2014-02-19 03:04:03 +0100  Mathieu Duponchelle <mduponchelle1@gmail.com>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: Port to new collectpads API
	  See: https://bugzilla.gnome.org/show_bug.cgi?id=724705

2014-03-16 15:26:04 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/types-compat.h:
	* sys/v4l2/ext/videodev2.h:
	  v4l2: Add types compatiblity for other OS
	  Adds type compatiblity with other OS like BSD. This uses types mapping macro to
	  avoid conflict with existing defined types. We resuse glib types as these are
	  already available on supported platforms. This is GCC only because of the
	  le32 type that uses bitwise attribute.
	  https://bugzilla.gnome.org/show_bug.cgi?id=726453

2014-03-16 15:55:00 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/pulse/pulseutil.c:
	  pulse: fix format info to caps conversion for mulaw

2013-08-13 12:10:42 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulse: Make gst_pulse_format_info_to_caps() shared
	  https://bugzilla.gnome.org/show_bug.cgi?id=678402

2014-03-15 18:41:16 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/Makefile.am:
	  v4l2: Fix typo V4L_DIR intead of V4L2_DIR

2013-12-29 17:29:53 +1100  Matthew Waters <ystreet00@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	* tests/examples/gtk/gstgtk.c:
	  [864/906] examples: update to gtk3

2013-07-17 11:22:02 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* tests/examples/gtk/gstgtk.c:
	  [771/906] gl: Some less long/ulong/gulong usage

2013-07-16 18:27:07 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [769/906] tests/examples: fix and port some of the examples.
	  Realize widgets, remove glupload element.

2013-07-10 11:24:34 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	* tests/examples/gtk/gstgtk.c:
	  [729/906] gl: Include config.h everywhere

2013-06-28 11:00:46 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [720/906] examples: Stop using deprecated GLib thread API

2012-11-08 22:53:56 +1100  Matthew Waters <ystreet00@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	* tests/examples/gtk/gstgtk.c:
	  [603/906] update FSF address

2012-08-14 14:41:19 +1000  Matthew Waters <ystreet00@gmail.com>

	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [560/906] examples: update for bus api changes and glimagesink changes

2012-06-07 00:51:47 +1000  Matthew Waters <ystreet00@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	* tests/examples/gtk/gstgtk.c:
	  [511/906] tests: update for 1.0

2010-09-16 15:00:29 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/gtk/gstgtk.c:
	  [461/906] xoverlay: require base from git and update to new API

2010-07-12 18:38:59 +0200  Julien Isorce <julien.isorce@gmail.com>

	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [457/906] gtk examples: adapt code since the native-window changes from gtk
	  Fixes bug #599885

2010-01-12 18:32:39 +0300  Руслан Ижбулатов <lrn1986@gmail.com>

	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [413/906] Fix Windows compiler warning in test/examples/gtk/fxtest/pixbufdrop.c

2009-10-23 01:07:29 +0200  Julien Isorce <julien.isorce@gmail.com>

	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [386/906] pixbufdrop: fix example on win32

2009-07-14 20:36:13 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/gstgtk.c:
	  [361/906] gstgtk: add missing license and copyright information

2009-07-14 20:25:28 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [360/906] examples: add missing copyright/license to my examples

2009-04-12 20:03:30 -0700  David Schleef <ds@hutch-2.local>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	* tests/examples/gtk/gstgtk.c:
	  [328/906] Convert gtk examples to use helper library
	  Helper lib implements gst-gtk glue on all platforms

2009-02-10 22:39:14 -0800  David Schleef <ds@schleef.org>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [310/906] Global reindent
	  Indent parameters:
	  INDENT_PARAMETERS="--braces-on-if-line \
	  --case-brace-indentation0 \
	  --case-indentation2 \
	  --braces-after-struct-decl-line \
	  --line-length80 \
	  --no-tabs \
	  --cuddle-else \
	  --dont-line-up-parentheses \
	  --honour-newlines \
	  --continuation-indentation4 \
	  --tab-size8 \
	  --indent-level2"

2009-02-05 13:13:51 -0800  David Schleef <ds@schleef.org>

	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [308/906] Rename glpixbufoverlay to gloverlay

2009-01-23 02:04:23 +0100  Julien Isorce <julien.isorce@gmail.com>

	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [301/906] depends on libpng instead of gdk_pixbuf

2009-02-10 21:57:31 -0800  David Schleef <ds@schleef.org>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [298/906] Revert "Fix indention"
	  This reverts commit 96e4ab18c2cf9876f6c031b9aba6282d0bd45a93.
	  You should have asked first.  And you would have been told "no",
	  because it causes people on development branches to do a huge
	  amount of extra work.

2009-02-03 18:33:36 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [295/906] Fix indention

2008-10-15 16:18:22 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	  [247/906] Import xray effect
	  Add xray effect. Maps luma to a negative, slightly cyan tinted, curve,
	  applies some light gaussian blur and multiplies it with its sobel edges. Not
	  sure about the name, likely to change. Probably still needs some tuning.

2008-08-19 22:15:17 +0200  Julien Isorce <julien.isorce@gmail.com>

	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [199/906] add pixbufdrop vs8 project

2008-08-19 21:04:29 +0200  Julien Isorce <julien.isorce@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [198/906] add fxtest vs8 project

2008-08-19 08:50:14 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/pixbufdrop.c:
	  [195/906] fix gstgldifferencematte and add an example app to test it dragging an image over the video (works with pixbufoverlay too, see pixbufdrop --help)

2008-08-16 17:36:10 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	  [180/906] minor cleanup in fxtest

2008-08-16 10:15:31 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	  [178/906] improve fxtest command line option handling, default to videotestsrc if no source bin description is given

2008-08-16 09:13:39 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	  [175/906] add sin effect (desaturate everything but red shades). still needs some tuning.

2008-08-14 21:29:02 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	  [173/906] add lumaxpro (desaturate + cross process) effect. nothing too impressive but I like it.

2008-08-14 20:54:54 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	  [172/906] add support for command line parsing to fxtest (try fxtest videotestsrc ! desired caps ! identity). report a new issue on BUGS.

2008-08-14 20:02:04 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* tests/examples/gtk/fxtest/fxtest.c:
	  [171/906] import fxtest (little gtk app to easily test effects) from cvs branch, fixed rgbtocurve.

2014-03-15 18:05:32 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	  v4l2-build: Set HAVE_GST_V4L2 if headers are present
	  The name of HAVE_ need to match the USE_. Now set HAVE_GST_V4L2 if
	  videodev2.h is found.

2014-03-15 16:47:51 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	* sys/Makefile.am:
	  v4l2: Actually build the plugin
	  The checks were removed inadvertedly in previous patch and not replaced.
	  Re-introduce the configure checks and some of the checks in order to enable
	  this plugin again. We only check if videodev2.h exist on the platform to
	  avoid building on Windows or OSX, though we build against our own copy. This
	  was breaking the build on built-bot.

2014-03-15 13:47:42 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  translation: PO file changes caused by POTFILE.in update

2014-03-15 13:17:21 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	* po/POTFILES.in:
	* po/POTFILES.skip:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2videooverlay.c:
	* sys/v4l2/gstv4l2videooverlay.h:
	  v4l2: Remove XV support
	  XV support for v4l2 never became upstream and ended up being
	  commented out with an undef for a long time now.

2014-03-15 11:13:05 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	* gst-plugins-good.spec.in:
	* sys/Makefile.am:
	* sys/v4l2/ext/v4l2-common.h:
	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2vidorient.c:
	* sys/v4l2/v4l2_calls.c:
	* tests/icles/Makefile.am:
	  v4l2: Use a copy of videodev2.h header
	  With years the amount of ifdef have grown up and we are not even sure if the
	  old code path compiles. Each time we need to update the v4l2 framework to add
	  the new feature, we break compilation on older kernel. With exception of two
	  controls in the video orientation control, this patch get rid of all ifdef by
	  including the latest version of videodev2.h inside GStreamer.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723446

2014-03-12 15:32:55 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Add properties for selecting SSL/TLS certificate checking
	  And by default properly check certificates against the system's CA
	  certificates. Everything else is not a good default at all.

2014-03-11 14:56:30 +0100  Per x Johansson <perxjoh@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix assert on fps lower than 1
	  Fixes assert caused by gst_duration_to_fraction calling
	  gst_util_uint64_scale_int with a denominator of 0 when fps is less
	  than 1.
	  https://bugzilla.gnome.org/show_bug.cgi?id=726106

2014-03-11 00:46:06 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/videomixer/videomixer2.c:
	  videomixer2: store video info with buffers to keep it in sync
	  Instead the queued buffer might have an old caps while the pad
	  is already storing the information for a new caps. Mixing those
	  while handling buffers will often lead to issues
	  https://bugzilla.gnome.org/show_bug.cgi?id=725948

2014-03-08 19:29:58 -0500  William Manley <will@williammanley.net>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Fix typo contol -> control
	  https://bugzilla.gnome.org/show_bug.cgi?id=725632

2014-03-04 01:15:49 +0000  William Manley <will@williammanley.net>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Normalise control names in the same way as v4l2-ctl
	  V4L2 kernel drivers allow configuration of the hardware settings via a
	  mechanism called controls.  These can be referred to by name such as
	  "Brightness" and "White Balance Temperature".  The user-space command line
	  client for setting these controls (v4l2-ctl) normalises these names such
	  that they only contain lower case alphanumeric characters and the
	  underscore '_'.  e.g:
	  Kernel                     v4l2-ctl
	  ----------------------------------------------------
	  Brightness                 brightness
	  White Balance Temperature  white_balance_temperature
	  Focus (absolute)           focus_absolute
	  GStreamer seems to want to follow this pattern but failed for controls with
	  more than one consecutive non-alphanum character.  e.g. GStreamer would
	  produce "focus__absolute_" rather than "focus_absolute".
	  This commit fixes that issue.  Backwards compatibility is preserved by
	  normalising all control names before comparison.
	  https://bugzilla.gnome.org/show_bug.cgi?id=725632

2014-03-07 16:17:29 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Make sure to not return EOS immediately if we finished a range request
	  Only return EOS the next time create() is called, if at all. basesrc
	  should already take care of not calling it again.
	  Also always return immediately if the previous flow return was
	  not OK. This indicates an error somewhere.

2014-03-06 12:06:43 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	  rtp: Remove caps restrictions from RTP depayloader sink caps
	  Remove caps restrictions that correspond to the default and are not
	  required in SDP. With the new usage of having pads require a subset
	  of the caps, they will make the negotiation fail.

2014-03-06 11:02:09 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpspeexdepay.c:
	  rtpspeexdepay: Remove caps restrictions for depayloader
	  The "encoding-params" is optional in the SDP, because we now require
	  a subset of the caps, it would fail caps negotiatioin if it wasn't present.
	  So removed it from the template caps.

2014-03-06 13:38:09 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't forget to quit mainloop after we cancelled when we got data after the stop position

2014-03-06 13:35:47 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: If we had a stop position, allow for the server to finish our connection instead of just cancelling
	  Otherwise keep-alive does not make much sense and also the server will have
	  confusing things in the logs.

2014-03-06 12:24:01 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: skip streams with same control url
	  Keep track of what streams we did the SETUP for. We only need to
	  configure caps, wait for pads and push events on setup streams. We can
	  remove the disabled state of the stream and simplify some checks.
	  After we setup a stream, skip the other streams that have the same
	  control url. Use a skipped flag to mark streams that should be skipped.

2014-03-06 12:22:47 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove obsolete code

2014-03-05 16:19:19 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: just use the SDP index as the stream id
	  Use the index of the media stream in the SDP as the stream id instead of
	  keeping a separate counter.

2014-03-05 13:35:19 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideo: fix build on Mac OSX Mavericks and put new window in front
	  GetCurrentProcess/SetFrontProcess/TransformProcessType was deprecated
	  and now removed in Mac OSX 10.9. orderFrontRegardless is used to make
	  the video window the most front window.

2014-03-05 17:33:56 +0100  Christian Fredrik Kalager Schaller <uraeus@linuxrising.org>

	* gst-plugins-good.spec.in:
	  Add docs directory to spec file

2014-03-05 15:44:25 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle NULL control urls better

2014-03-05 14:28:26 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpsession.c:
	  session: small cleanups
	  It's nicer to explicitly check for NULL on pointer types to make it
	  clear that it's a pointer and not a boolean.

2014-03-05 14:26:02 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpsession.c:
	  session: handle unknown SSRC in FIR
	  https://bugzilla.gnome.org/show_bug.cgi?id=725712

2014-03-05 11:39:09 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix seeking
	  Call gst_rtspsrc_connection_flush (src, FALSE) to reset connections as
	  non-flushing before sending PAUSE and PLAY with the new npt range. Without this
	  patch, those commands would fail with EINTR as the connections were still
	  flushing.

2014-03-03 16:39:26 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: expose xsub as a subtitle instead of as a video
	  It is placed inside a 'vids' struct, so it was being exposed on
	  a pad named video_%d. XSUB are subtitles and this patch adds
	  an special case for it to be exposed in a subpicture_%d pad

2014-03-03 16:38:45 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: do not try to add a tag with tag_name set to NULL
	  This can happen if there are subtitles in the stream, leading to
	  an assertion

2014-03-04 16:40:34 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add support for multiple payload types
	  A media stream can have multiple payload types. Parse all the payload
	  types and collect the caps information. We then have to store the
	  pt<->caps mapping instead of 1 pt and 1 caps.
	  Parse the profile from the SDP and use that to negotiate the transport
	  instead of always using AVP.
	  Rework how we do some tweaks for ASF and Realmedia.

2014-03-04 11:34:39 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: refactor payload handling

2014-03-03 11:34:00 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: fix buffer level with invalid DTS
	  It is possible that the DTS is invalid (when we receive RTP packets from
	  TCP, for example). As a fallback, use the reconstructed PTS value to
	  calculate the buffer level.

2014-03-02 05:10:13 +0100  Sebastian Rasmussen <sebras@hotmail.com>

	* .gitignore:
	  .gitignore: Ignore gcov intermediate files
	  https://bugzilla.gnome.org/show_bug.cgi?id=725480

2014-02-28 09:34:46 +0100  Sebastian Dröge <sebastian@centricular.com>

	* common:
	  Automatic update of common submodule
	  From fe1672e to bcb1518

2014-02-27 23:15:04 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/audioparsers/gstaacparse.c:
	  Revert "aacparse: put codec data on caps for loas format"
	  This reverts commit e459cf3e01a08f1a3ef1fb954a41cfa36b3e510c.
	  This was pushed by accident, the bug should likely be fixed in
	  libav https://bugzilla.libav.org/show_bug.cgi?id=644

2014-02-27 18:55:04 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: mark all parsed frames as sync points
	  all jpeg frames are sync points, so mark them as such so
	  reverse playback can properly work with the video decoder
	  base class
	  https://bugzilla.gnome.org/show_bug.cgi?id=725104

2014-02-25 01:12:05 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: put codec data on caps for loas format
	  gst-libav audio decoder also needs codec data for LOAS format, otherwise
	  it will complain about not having a decoder config and skip all packets
	  https://bugzilla.gnome.org/show_bug.cgi?id=596772

2014-02-27 00:43:48 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: align raw audio memory to powers of two
	  https://bugzilla.gnome.org/show_bug.cgi?id=725008

2014-02-27 00:37:20 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: calculate alignment properly for audio depths not a multiple of 8

2014-02-23 19:09:24 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix crash with 24-bit raw audio
	  Do not try to align audio buffers to odd numbers,
	  which will get us a NULL buffer which we then
	  crash on.
	  https://bugzilla.gnome.org/show_bug.cgi?id=725008

2014-02-27 00:11:42 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/Makefile.am:
	  rtpmanager: re-enable -Werror

2014-02-27 00:11:11 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix compiler warning
	  gstrtpjitterbuffer.c: In function 'gst_rtp_jitter_buffer_loop':
	  gstrtpjitterbuffer.c:2978:3: error: 'result' may be used uninitialized in this function
	  while (result == GST_FLOW_OK);
	  ^

2014-02-26 22:11:41 +0100  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 1a07da9 to fe1672e

2014-02-26 21:11:23 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix uninitialized variable compiler warning

2014-02-26 07:32:32 -0500  Jake Foytik <jake.foytik@ipconfigure.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Remove raw comparisons of RTP sequence numbers
	  Several conditional statements perform comparison on RTP sequence
	  numbers without taking the sequence number rollover into account.
	  Instead, use the gst_rtp_buffer_compare_seqnum function to perform the
	  comparison.
	  https://bugzilla.gnome.org/show_bug.cgi?id=725159

2014-02-03 01:44:21 +0100  Sebastian Rasmussen <sebras@hotmail.com>

	* tests/check/Makefile.am:
	  tests: Don't build disabled plugins' check tests
	  https://bugzilla.gnome.org/show_bug.cgi?id=723502

2014-02-26 11:29:45 +0100  Stefan Sauer <ensonic@users.sf.net>

	* docs/Makefile.am:
	  docs: install prebuilt plugin docs if gtk-doc is disabled
	  Sync to the Makefile.am from gst-plugin-base where it is done right.
	  Fixes #725034

2014-02-25 16:10:54 -0500  Hugues Fruchet <hugues.fruchet@st.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: do not emit "parsed" caps for vp8
	  VP8 doesn't require parsing (vp8parse doesn't exist, so negotiation with demux fails
	  if "parsed" is set in caps).
	  https://bugzilla.gnome.org/show_bug.cgi?id=724636

2014-02-11 16:27:08 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Don't require parser for VP8
	  Until GStreamer has one (see bug722760), we should not require a parser for VP8.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722128

2014-02-10 17:08:25 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: CAPTURE_MPLANE is well tested now
	  https://bugzilla.gnome.org/show_bug.cgi?id=722128

2013-12-18 09:56:35 +0100  Benjamin Gaignard <benjamin.gaignard@linaro.org>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videodec.h:
	  v4l2videodec: Create one element per device
	  For each videoCdevice probe it input/output capabilities
	  if it match with video decoder requirement register a new element.
	  Signed-off-by: Benjamin Gaignard <benjamin.gaignard@linaro.org>
	  https://bugzilla.gnome.org/show_bug.cgi?id=722128

2013-12-19 15:26:52 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Calculate latency from device information
	  Decoders or other devices that expose a minimum buffers required produce
	  an first output. We use this information to calculate latency.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722128

2013-11-28 17:14:18 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videodec.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2videodec: Implement v4l2videodec
	  Implement an element that can driver V4L2 M2M decoder device.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722128

2014-02-11 12:41:29 +0100  Göran Jönsson <goranjn@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: only update last_spspps time if all sps/pps got sent successfully
	  This fixes an issue with gst-rtsp-server where no sps and pps are
	  sent for the first intra frame, because the payloader starts working
	  already when receiving DESCRIBE but there is no transports so it tries
	  to send sps and pps, but that fails with a FLUSHING flow. But the time
	  for last sent sps and pps would still be set, so when PLAY arrives and
	  the first intra frame is to be sent there is no sps and pps sent due to
	  that time since last sps pps is less than spspps_interval.
	  https://bugzilla.gnome.org/show_bug.cgi?id=724213

2014-02-25 09:00:45 +0100  Santiago Carot-Nemesio <sancane@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix deadlock when task creation is no successful
	  https://bugzilla.gnome.org/show_bug.cgi?id=725124

2014-02-22 20:19:49 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/autodetect/gstautodetect.c:
	  autodetect: demote candidate error to warning and plug fake{sink,src}
	  In the case where we have no suitable candidate we post a warning and plug a
	  fake-element. Do the same when non of the candidate work.
	  This is more consistent and plugin the fakesink as a fallback is probably
	  helpful for running unit tests without requiring hardware src/sink elements.
	  Fixes #722981

2014-02-23 12:34:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: make some more controls configurable
	  ... at least if one tries hard enough using extra-controls property.

2014-02-23 10:39:20 +0100  Dan Kegel <dank@kegel.com>

	* configure.ac:
	  v4l2: Require mplanar support for now in configure
	  The code fails to compile without currently, see
	  https://bugzilla.gnome.org/show_bug.cgi?id=723446
	  It's better to disable it instead of failing compilation
	  until this is fixed properly.

2014-02-23 00:14:04 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/jack/gstjackaudioclient.c:
	  jack: add some simple log handlers for jack
	  Add log handlers for jack that write to the gst debug log. This avoids spamming
	  the console when e.g. using autoaudiosink, having the jack elements installed,
	  but not running jack.

2014-02-22 21:31:21 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: handle old and odd driver behaviour when listing controls

2013-11-28 16:54:58 -0800  Darryl Gamroth <dgamroth@uvic.ca>

	* gst/audiofx/audiofxbaseiirfilter.c:
	  audiofxbaseiirfilter: check if coefficients are provided inside filter lock
	  https://bugzilla.gnome.org/show_bug.cgi?id=719524

2014-02-21 19:46:44 +0000  Tim-Philipp Müller <tim@centricular.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2src: also unset INTERLACED flag on buffers if frame is not interlaced
	  https://bugzilla.gnome.org/show_bug.cgi?id=724899

2014-02-21 14:31:59 +0000  Simon Farnsworth <simon.farnsworth@onelan.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2src: Flag interlaced buffers as interlaced.
	  We correctly indicate the field ordering on interlaced buffers, but fail to
	  flag them as containing interlaced video, which we need to do here because
	  we signal interlace-mode=mixed in our caps. This means that downstream
	  elements (like vaapipostproc from gstreamer-vaapi) don't recognise these
	  buffers as in need of deinterlacing.
	  Fix this by setting the interlaced flag on all interlaced buffers.
	  Signed-off-by: Simon Farnsworth <simon.farnsworth@onelan.co.uk>
	  https://bugzilla.gnome.org/show_bug.cgi?id=724899

2014-02-19 13:56:37 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: be more strict at ADTS header parsing
	  Adds two extra checks:
	  - Sampling frequency on header can't be 15.
	  - Frame size should be at least 9 or 7, depending
	  on whether CRC protection is present.
	  https://bugzilla.gnome.org/show_bug.cgi?id=724638

2014-02-19 13:35:59 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: make sure we have enough ADTS data
	  We need at least 6 bytes to pass over to _get_frame_len()
	  but we were just checking for a minimum of 2 bytes for the
	  syncword.
	  https://bugzilla.gnome.org/show_bug.cgi?id=724638

2014-02-20 22:52:57 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/autodetect/gstautodetect.c:
	* gst/autodetect/gstautodetect.h:
	  autodetect: check if the kid has a sync property
	  previously autovideosrc did not have a sync property and v4l2src has none either.

2014-02-19 21:55:52 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosink.h:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautoaudiosrc.h:
	* gst/autodetect/gstautodetect.c:
	* gst/autodetect/gstautodetect.h:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosink.h:
	* gst/autodetect/gstautovideosrc.c:
	* gst/autodetect/gstautovideosrc.h:
	  autodetect: use a common baseclass
	  This makes the actual elements super simple. We're using the ELEMENT_FLAG to
	  configure source/sink and a string for the Audio/Video type.

2014-02-14 17:14:42 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add tls-database property
	  Add support for a new property: tls-database. If the property is set,
	  the certificate database will be given to the rtsp connection if TLS
	  protocol is being used. If the server certificate can't be verified with
	  the default database, this additional database will be used.
	  https://bugzilla.gnome.org/show_bug.cgi?id=724396

2014-02-19 22:21:54 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudio: remove unused variables

2014-02-19 21:26:03 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautodetect.c:
	* gst/autodetect/gstautodetect.h:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	  autodetect: extract common helper code
	  The function to generate the pretty names is basically the same. Use one and add
	  a parameter.

2014-02-19 21:01:39 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/Makefile.am:
	* tests/check/elements/autodetect.c:
	  autodetect: improve the tests
	  Add fake audio/video sinks. Previously running the test might be flaky due to
	  the use of real elements (hardware in use), which we don't want to test here.
	  Add two more tests that check that the fakes are chosen.

2014-02-19 15:19:30 +0100  Branislav Katreniak <bkatreniak@nuvotechnologies.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: do not emit error when connection with unknown size ends
	  Commit 46fd12ae5ec53200b16dfd7f17048d6bc60fbfbc introduced connection
	  recovery. But when server does not specify content-size,
	  souphttpsrc tries to reconnect even after regular end of stream.
	  Http server replies  with SOUP_STATUS_REQUESTED_RANGE_NOT_SATISFIABLE
	  but souphttpsrc still emits error instead of EOS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=724717
	  Signed-off-by: Branislav Katreniak <bkatreniak@nuvotechnologies.com>

2014-02-19 11:26:22 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/autodetect.c:
	  autodetect: fix the disabled test
	  Use a shared helper for both tests. It turns out that the valgrind variant is
	  fine (maybe due to picking up pulsesink though).

2014-02-19 11:05:35 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/autodetect.c:
	  autodetect: remove cruft from the test
	  Remove the obsolete version check and use the ignore macro for the disabled test.

2014-02-18 22:54:45 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/level/gstlevel.c:
	* gst/spectrum/gstspectrum.c:
	  docs: use docbook markup for xi:include
	  It turns out that the change in gtk-doc-1.20 which wraps the |[]| content in
	  CDATA break xi:inlcude examples. As in a whole jhbuild checkout these where
	  the only 4, we're fixing them instead.

2014-02-18 22:35:45 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/isomp4/gstqtmux-doc.h:
	  isomp4mux: fix copy and paste
	  This fixes doc warnings.

2014-02-18 21:44:24 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/debugutils/gstcapssetter.c:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux.c:
	* gst/level/gstlevel.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrgvolume.c:
	  docs: use the gtk-doc syntax to link to properties
	  Don't use docbook unless needed. Also stip other docbook tags in the the files we fix.

2014-02-18 11:28:18 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix crash when getting the current-device in NULL->READY
	  The "goto unlock" is wrong as in this code path we haven't take the lock yet.
	  Fixes #724619

2014-02-14 22:50:49 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  soup: We need libsoup >= 2.40 for proper usage of the content decoder
	  Previous versions did not consider our chunk allocator and allocated
	  memory by themselves, which caused crashes and broken behaviour.

2014-02-14 15:27:20 -0500  William Jon McCann <william.jon.mccann@gmail.com>

	* gst/audiofx/audiocheblimit.c:
	* gst/udp/gstudpsrc.c:
	  docs: fix mismatched para tags
	  newer gtkdoc is more sensitive to mismatched docbook tags.
	  This fixes the build in master.

2014-02-14 15:59:46 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: add support for serialized queries
	  See https://bugzilla.gnome.org/show_bug.cgi?id=723850

2014-02-14 15:53:55 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: fix typecast to fix compilation

2014-02-14 12:01:00 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: proxy caps and allocation on RTP pads
	  recv_rtp_sink: allow proxying of the allocation query.
	  send_rtp_sink: allow proxying of caps and allocation. This allows us to
	  query caps downstream as well as get an allocator from downstream.
	  send_rtp_src: allow proxy of caps, this makes the caps query do
	  upstream.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=723850

2014-02-13 12:29:13 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: handle tags in mac encoding
	  Check the charset from (C)*** tags and set the charset
	  to convert from MAC encoding if suitable.
	  https://bugzilla.gnome.org/show_bug.cgi?id=723166

2014-02-13 12:09:13 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Use new automatic_eos API from basesrc
	  We want to notice ourselves that we're EOS. Otherwise we will
	  always cancel requests in the very end and confuse the server...
	  and also make it impossible to use persistent connections.

2014-02-13 11:11:13 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Consistently use have_size instead of content_size!=0

2014-02-13 10:30:09 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Free extra headers when finalizing the element
	  It's set as property by the application, we should not just reset
	  properties when going back to READY.

2014-02-13 10:28:13 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Properly close the session when going back to NULL
	  Don't wait for that until the element is disposed.

2013-02-28 12:20:52 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: add support for keep-alive sessions
	  https://bugzilla.gnome.org/show_bug.cgi?id=699926

2014-02-12 13:00:13 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Add "compress" property to enable/disable automatic gzip/deflate content encoding handling

2014-02-12 12:39:10 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Retry connection if we're finished before the content size only if we actually have a content size
	  https://bugzilla.gnome.org/show_bug.cgi?id=722185

2014-02-12 10:08:50 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouputils.c:
	  souputils: Fix compiler warning
	  gstsouputils.c:35:25: error: comparison of constant 9 with expression of type
	  'SoupLoggerLogLevel' is always false
	  [-Werror,-Wtautological-constant-out-of-range-compare]

2014-01-07 23:00:56 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* ext/soup/Makefile.am:
	* ext/soup/gstsoup.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpclientsink.h:
	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	* ext/soup/gstsouputils.c:
	* ext/soup/gstsouputils.h:
	  souphttp*: add ability to do HTTP session logging
	  This changeset adds the loggin infrastructure and
	  mods both souphttpsrc and souphttclientsink to use it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=721764

2014-02-07 14:00:15 +0100  divhaere <dirk.vanhaerenborgh@ugent.be>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: add support for GRAY8, BGR and RGB video colourspaces in V_UNCOMPRESSED codec
	  https://bugzilla.gnome.org/show_bug.cgi?id=723849

2014-02-11 13:25:46 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Add mapping for NOT_FOUND and NOT_AUTHORIZED errors

2014-02-11 13:25:22 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't duplicate status_code to GStreamer error mapping

2014-02-09 23:38:44 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/goom/filters.c:
	* gst/goom2k1/filters.c:
	  goom: Remove unused functions

2014-02-09 23:21:20 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: Comment out some unused functions used only from the commented out pull-mode code

2014-02-08 21:01:32 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: Fix another compiler warning

2014-02-08 17:43:32 +0100  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/souphttpsrc.c:
	  souphttpsrc: Fix implicit enum conversion compiler warning
	  error: implicit conversion from enumeration type
	  'SoupStatus' to different enumeration type 'SoupKnownStatusCode'

2014-02-08 17:41:21 +0100  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/interleave.c:
	  interleave: Fix unitialized variable compiler warning in test
	  error: variable 'mask' is used uninitialized
	  whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]

2014-02-08 17:27:51 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: Fix unitialized variable compiler warning
	  error: variable 'image_type' is used uninitialized
	  whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]

2014-02-08 17:25:27 +0100  Sebastian Dröge <sebastian@centricular.com>

	* sys/oss4/oss4-audio.h:
	  oss4: Fix typo in header include guard
	  error: 'GST_OSS4_AUDIO_H' is used as a header guard here,
	  followed by #define of a different macro [-Werror,-Wheader-guard]

2014-02-08 17:24:06 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: Fix unitialized variable compiler warning
	  variable 'rtx_ssrc' is used uninitialized whenever
	  'if' condition is false [-Werror,-Wsometimes-uninitialized]

2014-02-08 17:21:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpac3depay.c:
	  rtpac3depay: Remove unused variable

2014-02-08 17:19:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/flx/flx_fmt.h:
	  flx: Fix typo in header include guard
	  error: '__GST_FLX_FMT__H__' is used as a header guard here,
	  followed by #define of a different macro [-Werror,-Wheader-guard]

2014-02-07 10:07:41 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: remove have_dts flag from pads
	  It was used in the past in 0.10 when there was no explicit DTS
	  field in buffers, now we have it in 1.x series and we can
	  check it directly with GST_BUFFER_DTS_IS_VALID

2014-02-07 01:49:26 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: improve support for sparse streams
	  Do not try to use subsequent buffer timestamps to calculate
	  sparse streams durations because the stream is sparse and
	  the buffers might not be 'time adjacent'. So rely on the
	  duration and give the option to the pad to provide
	  custom 'empty' buffers to represent the gaps in the
	  stream, this can vary on how the data is represented.
	  Right now, the only sparse stream supported is tx3g subtitles.

2014-02-06 12:15:22 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: add support for text/x-raw subtitles
	  Adds it to mp4mux, qtmux and gppmux.
	  Buffers need to be prefixed with 2 bytes for the text length before
	  being muxed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=581295

2014-02-06 12:09:01 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	  qtmux: add support for the TX3G atoms
	  Adds functions for creating and setting values related to the
	  tx3g atom for raw text subtitle support.
	  QTFF spec has information on those atoms
	  https://bugzilla.gnome.org/show_bug.cgi?id=581295

2014-02-05 10:27:54 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/gstqtmuxmap.h:
	  qtmux: add subtitle support to qtmuxmap structures
	  adds basic stubs for subtitle support around the qtmux and
	  qtmuxmap structures. Still no real subtitle implemented, but
	  basic functions in place
	  https://bugzilla.gnome.org/show_bug.cgi?id=581295

2014-01-20 17:31:14 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: factor out read context init/reset
	  While at this, move _track_reset() to track-ids
	  so it can be called from the common read context
	  reset routine.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722705

2014-02-06 12:21:07 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/effectv/gstrev.c:
	  effectv: fix doc section of revtv element

2014-02-05 12:46:54 +0100  Edward Hervey <bilboed@bilboed.com>

	* sys/osxvideo/Makefile.am:
	  osxvideo: Fix libtool usage
	  --tag=CC is needed for static build

2014-01-16 11:26:41 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: do not try set deinterlace method if passthrough is enabled
	  Fixes an issue with progressive content and unsupported video formats
	  for the deinterlace method.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719636

2014-02-04 21:26:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/flac/gstflacenc.c:
	  flacenc: order format in template caps by preference
	  To minimise risk of bad fixation, though audioconvert
	  at least should be smart enough to avoid it.

2014-02-02 09:57:03 -0800  Dan Kegel <dank@kegel.com>

	* configure.ac:
	  v4l2: Remove obsolete definition GST_V4L2_MISSING_BUFDECL
	  The only use was removed by 9edc0c0365f79ab07ff2e65461c6696e3931a3f0
	  https://bugzilla.gnome.org/show_bug.cgi?id=723446

2014-02-04 13:43:56 +0100  Rafał Mużyło <galtgendo@o2.pl>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* gst/cutter/gstcutter.c:
	  gst: Don't use endianness-specific S8 audio format
	  It does not exist.
	  https://bugzilla.gnome.org/show_bug.cgi?id=723331

2014-01-31 14:17:54 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* ext/cairo/gstcairooverlay.c:
	  cairooverlay: add support for RGB16
	  https://bugzilla.gnome.org/show_bug.cgi?id=723289

2014-01-30 09:43:50 +0100  Per x Johansson <perxjoh@axis.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: Fix constantly growing used uid list
	  Moves the used uid list to the class to avoid having it grow forever.
	  https://bugzilla.gnome.org/show_bug.cgi?id=723269

2014-01-30 10:44:05 +0100  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From d48bed3 to 1a07da9

2014-01-24 01:52:08 +0000  Mike Sheldon <elleo@gnu.org>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Ignore Broadcast Wave Format (BWF) tags when searching for 'fmt' chunk
	  https://bugzilla.gnome.org/show_bug.cgi?id=723125

2014-01-29 10:37:53 +0100  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/rtpaux.c:
	  check: Use fakesink sync=True instead of an audio sink
	  Ensures the test can run on systems without alsa (or any audio output for
	  that matter), and will avoid people running build slaves wondering what
	  the hell was beeping during the night :)

2014-01-27 20:05:42 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: custom get_sink_caps handling for private stream caps
	  ... now that those are transformed rather than parsed, some transforming
	  of caps is required as well to make auto-plugging succeed.

2014-01-25 02:06:00 -0500  Ryan Lortie <desrt@desrt.ca>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: guard use of ENODATA with #ifdef
	  Not all systems with v4l have ENODATA defined, so check that we have it
	  before attempting to use it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722953

2014-01-24 12:37:39 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  Revert "rtspsrc: Proxy rtpjitterbuffer do-retransmission property"
	  This reverts commit 9f7b1128b1f00a2b87a232ff890867549ab95ba5.
	  This should be handled automatically be rtspsrc if the AVPF profile
	  is used, and manual enabling of it can be done with the new-manager
	  signal.

2014-01-24 10:21:11 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add signal to notify of new manager
	  So that you can configure and connect to signals on the rtpbin.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=722866

2014-01-23 15:17:58 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Proxy rtpjitterbuffer do-retransmission property
	  https://bugzilla.gnome.org/show_bug.cgi?id=722866

2014-01-21 17:52:44 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: handle expected packet being an RTX packet
	  If the expected packet (do_next_seqnum is TRUE) is the one we requested
	  for retranmission earlier, do the logic to update the retransmission
	  statistics as well before setting up the timers for the next expected
	  packet.
	  Also reset the retransmission counter if the timer is reused for another
	  seqnum.

2014-01-21 15:48:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: add a caps accumulator for the request-pt-map signal
	  Add an accumulator that stops the signal emission as soon as a caps has
	  been retrieved. Otherwise the default handler would continue emitting
	  the signal and possibly overwrite the result with NULL again.

2014-01-21 15:25:54 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtxreceive: copy flags and timestamps from original buffer

2014-01-21 15:24:52 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: ignore invalid timestamps in rtt calculation
	  When the input buffer does not have a valid timestamp, don't try to
	  calculate the round-trip-time.

2014-01-16 14:23:13 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroskaparse: better default caps when none set
	  Uses information gathered during EBML parsing to
	  forge a more suitable set of caps instead of blindly
	  assuming everything is video/x-matroska.
	  For consistency, stream type reset was added to
	  matroska-demux too.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722311

2014-01-15 17:29:35 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests: rtprtx::test_rtxreceive_data_reconstruction: remove useless code for triggering retransmission
	  There is no need anymore to push yet another buffer in rtxsend
	  in order to trigger the previously requested retransmissions
	  to actually happen.

2014-01-15 17:27:19 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests: rtprtx::test_rtxreceive_data_reconstruction: fix race condition
	  Now with rtprtxsend pushing rtx buffers from a different thread,
	  this is necessary to ensure that the result of the test is deterministic.
	  This code makes use of GstCheck's global GMutex and GCond that are
	  being used inside GstCheck's sink pad chain() function in order
	  to synchronize with it.

2014-01-15 17:17:57 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests: rtprtx::test_rtxsender_packet_retention: fix race condition
	  Now with rtprtxsend pushing rtx buffers from a different thread,
	  this is necessary to ensure that the result of the test is deterministic.
	  This code makes use of GstCheck's global GMutex and GCond that are
	  being used inside GstCheck's sink pad chain() function in order
	  to synchronize with it.

2014-01-15 11:26:33 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests: rtprtx::test_push_forward_seq: fix race condition
	  Now with rtprtxsend pushing rtx buffers from a different thread,
	  this is necessary to ensure that the result of the test is deterministic.
	  This code makes use of GstCheck's global GMutex and GCond that are
	  being used inside GstCheck's sink pad chain() function in order
	  to synchronize with it.

2014-01-15 09:47:03 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests: rtprtx::test_push_forward_seq: fix buffer refcounting

2014-01-21 13:42:38 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: ensure that no rtx buffers are sent after EOS
	  To do that, enqueue the EOS event to be sent from the srcpad task
	  thread and flush the queue right afterwards, so that no more rtx
	  buffers can be sent, even if there are more requests coming in.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722370

2014-01-15 09:46:14 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	  rtprtxsend: run a new GstTask on the src pad
	  The reason behind this is to minimize the retransmission delay.
	  Previously, when a NACK was received, rtprtxsend would put a
	  retransmission packet in a queue and it would send it from chain(),
	  i.e. only after a new buffer would arrive.
	  This unfortunately was causing big delays, in the order of 60-100 ms,
	  which can be critical for the receiver side.
	  By having a separate GstTask for pushing buffers out of rtxsend,
	  we can push buffers out right after receiving the event, without
	  waiting for chain() to get called.

2014-01-03 17:47:55 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2send: error out if no caps were received
	  Instead of assuming that input is ogg.

2014-01-03 17:30:12 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/shout2/gstshout2.c:
	  shout2send: accept audio/webm, audio/ogg and video/ogg as well
	  Those are advertised in the template caps, but the
	  setcaps handler didn't handle them. But then oggmux
	  and oggparse at least for now still always output
	  application/ogg anyway, so that wasn't a real problem.

2014-01-20 10:12:45 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: Don't leak input buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=722414

2014-01-19 17:40:56 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavimux.c:
	  avimux: reset some more audio pad data when needed

2014-01-19 17:38:59 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avimux: write correct blockalign for vbr audio
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720659

2014-01-16 17:36:12 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: do not drop serialized events when latency is set
	  Serialized events are now queued in the jitter buffer, so we don't
	  want to drop them even latency is set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=722372

2013-12-11 09:36:22 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/avi/gstavimux.c:
	  avimux: don't make the buffer writable unless absolutely necessary
	  https://bugzilla.gnome.org/show_bug.cgi?id=722396

2013-09-12 16:56:56 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: set GST_BUFFER_FLAG_DELTA_UNIT when appropriate
	  https://bugzilla.gnome.org/show_bug.cgi?id=722394

2014-01-17 07:46:09 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: don't ref the newly created allocator
	  Otherwise the allocator will never be deleted.
	  https://bugzilla.gnome.org/show_bug.cgi?id=712612

2014-01-15 22:47:12 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't skip all video frames until the first keyframe
	  Instead do it like all other demuxers and let parsers and decoders
	  handle that. The keyframe information inside the container might
	  be completely wrong like in the sample file of the bug report,
	  and if it is correct and we push no keyframes, then the parsers
	  and decoders will handle that properly anyway.
	  https://bugzilla.gnome.org/show_bug.cgi?id=682276

2014-01-13 10:08:09 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: remove elst_offset variables
	  They are not used anymore

2014-01-06 21:36:17 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: remember reverse playback when verifying the segment end
	  Check if the rate is positive or negative to correctly compare the current
	  position with the segment to make reverse playback work

2014-01-03 10:59:35 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: do not ignore empty segments
	  Make sure empty segments are used and pushed with a gap event
	  to represent its data (or lack of it)
	  Each QtSegment is mapped into a GstSegment with the corresponding
	  media range. For empty QtSegments a gap event is pushed instead
	  of GstBuffers and it advances to the next QtSegment.
	  To make this work with seeks, need to keep track of the starting
	  'base' to make sure it remains consistently increasing when
	  pushing new segment events.
	  For example: if a seek makes qtdemux start from 5s, the first
	  segment will have a base=0. When the next segment is activated,
	  its base time will be QtSegment.time - qtdemux.segment_base so
	  that it doesn't include the first 5s that weren't played and
	  shouldn't be accounted on the running time
	  This purposedly will remove the fix made for
	  https://bugzilla.gnome.org/show_bug.cgi?id=700264, at this
	  point it was decided to respect the gaps, even if they cause
	  a delay on playback, because that's the way the file was crafted.
	  https://bugzilla.gnome.org/show_bug.cgi?id=345830

2013-12-12 23:05:43 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests: Remove usage of the system clock from the rtprtx test

2013-12-12 23:22:41 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpcollision.c:
	  tests: Initial segment in rtpcollision test

2014-01-14 15:56:42 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/examples/rtp/client-rtpaux.c:
	* tests/examples/rtp/server-rtpaux.c:
	  examples/*-rtpaux: specify payload type association for the audio stream, so that rtx works also for audio

2014-01-14 13:08:18 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: remove wrong check for payload type not having been set
	  1) pt can be lower than 96
	  2) there is no point in checking that because rtprtxsend will not
	  even store buffers for payload types that it doesn't know about,
	  so this case will never be reached

2014-01-14 13:01:41 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: fix data locking when creating rtx packets
	  This patch moves the creation of rtx packets to be done early,
	  in the src_event() function, when they are requested. The purpose
	  is to run gst_rtp_rtx_buffer_new() with the object locked to
	  protect internal data, because if it is done at the pushing stage,
	  we would have to lock and unlock multiple times in a row while we
	  are pushing the rtx buffers.
	  Previously there was no locking at all, which was terribly wrong.

2014-01-14 12:50:23 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: lock access to internal data in sink_event() function

2014-01-14 12:44:06 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: remove unnecessary call to reset() from finalize()
	  ...and use _free_full() on the pending buffers queue now that
	  reset() is not being called

2014-01-14 12:38:51 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: remove unused parameter from the internal reset() method

2014-01-14 12:32:38 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: Use g_slice_* for allocating internal structures

2014-01-14 12:28:01 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtprtxreceive: remove stupid mutex unlock in the middle of chain()

2014-01-14 12:25:36 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtprtxreceive: use GST_DEBUG_OBJECT / GST_WARNING_OBJECT instead of GST_DEBUG / g_warning

2014-01-14 12:19:58 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtprtxreceive: fix integer format specifiers in GST_DEBUG
	  seqnum in this function is 32-bit, so G_GUINT16_FORMAT would
	  produce undefined output on big endian systems

2014-01-14 12:13:49 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	  rtprtxsend: change the rtx_pt_map directly in set_property() instead of delaying it for chain()
	  The same lock is held, so there is no point in complicating it...

2014-01-14 12:07:58 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxreceive.h:
	  rtprtxreceive: change the rtx_pt_map directly in set_property() instead of delaying it for chain()
	  The same lock is held, so there is no point in complicating it...

2014-01-14 11:55:00 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	  rtprtxreceive: simplify the code of finalize()

2014-01-14 11:52:21 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxreceive.h:
	  rtprtxreceive: use the GstObject lock instead of a new one

2014-01-14 11:45:52 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	  rtprtxsend: use the GstObject lock instead of a new one

2013-12-10 14:29:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add NV12_64Z32 support
	  https://bugzilla.gnome.org/show_bug.cgi?id=722127

2014-01-14 19:08:49 +0900  Justin Joy <justin.joy.9to5@gmail.com>

	* sys/oss/gstosshelper.c:
	  osshelper: Don't leak fd when getting card name
	  https://bugzilla.gnome.org/show_bug.cgi?id=722163

2014-01-14 09:43:33 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  Revert "aacparse: relax the detection of ADTS"
	  This was pushed by mistake along with the V4L2 fix.
	  This reverts commit 8eb4b032bef444397c4d211f2095c173ba114187.

2014-01-14 15:42:01 +0900  Justin Joy <justin.joy.9to5@gmail.com>

	* gst/rtp/gstrtpg726pay.c:
	  rtpg726pay: don't leak encoding_name string
	  https://bugzilla.gnome.org/show_bug.cgi?id=722159

2014-01-13 09:14:00 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: fix build break using V4L2_CAP_VIDEO_M2M_MPLANE
	  This may not be defined. Since the previous version used
	  only the other define (V4L2_CAP_VIDEO_OUTPUT_MPLANE), fall
	  back on this only when not available.

2013-02-27 01:45:52 +0900  Akihiro Tsukada <atsukada@users.sourceforge.net>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: relax the detection of ADTS
	  According to ISO/IEC 13818-7, "channel_config" field in ADTS header
	  may have value of 0, as in the case of frame with PCE.
	  gst_aac_parse_detect_streams() returned FALSE for those frames
	  and discarded them.

2014-01-07 11:58:23 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: check set_config return value in gst_v4l2_buffer_pool_new
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2014-01-10 12:40:31 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Add parsed=1 field for encoded output
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2014-01-10 12:39:16 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't leak empty caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2014-01-08 16:51:21 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: do not stop a stream not previously started
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-12 16:27:21 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't enforce dimension field on encoded formats
	  Don't enforce having width, height and framerate in template caps for encoded
	  formats. These don't always need to be exposed and may break negotiation for
	  decoder and decoding sink. If needed, these field will be automatically added
	  when probed caps are known.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-12 17:09:59 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: unref downstream pool
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-18 13:37:23 -0500  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: add gst_v4l2_buffer_pool_flush
	  STREAMOFF set all v4l2buffers to DEQUEUE state.
	  Then for CAPTURE we call QBUF on each buffer.
	  For OUTPUT the buffers are just push back in the GstBufferPool
	  base class 's queue.
	  But the loop actually looks like the same.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-16 17:29:30 -0500  Benjamin Gaignard <benjamin.gaignard@linaro.org>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Add vp8 support
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-12 16:46:09 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't force framerate field for OUTPUT
	  If there is nothing that seems to force a certain framerate on output device, it is
	  preferable to simply not set that feild. This allow negotiation with tsdemux in a
	  decoder for example.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-12 14:07:03 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: _v4l2fourcc_to_structure() can be static
	  This function is not used anymore outside v4l2object.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-12 14:22:26 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Add MPEG1/2 support
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-12 12:18:45 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Ask for a decent buffer size when dealing with encoded formats
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-07 14:03:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: On warn on size change if n_planes > 1
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-31 16:38:09 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: check if translated format is valid
	  Also add a FIXME in gst_v4l2_object_setup_format
	  to note that the whole function has to be improved
	  in order to support ENCODED formats.
	  It requires to have an encoder device which we do not
	  have right now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-07 10:31:15 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Validate returned dimensions
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-05 19:36:25 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Ensure max is not smaller then min in decide_allocation
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-05 19:36:06 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't keep the max paramter when using our own pool
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-05 19:34:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Respect the suggested min buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-05 18:48:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Allocate pool if needed in decide_allocation
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-05 18:49:19 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Add V4L2_CID_MIN_BUFFERS_FOR_CAPTURE support
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-05 18:48:15 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Move decide allocation into v4l2object
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-05 13:51:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Implement _setup_format()
	  This method allow setting up the object from the currently configured format on the
	  device. This is useful for M2M element where input data decides the format that will
	  be set on capture side.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-10 14:34:17 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Split out saving format from set_format()
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-31 15:37:26 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: set only one plane for encoded format
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-04 16:49:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Move code block where it belongs
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-04 16:26:12 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't check format specific information
	  The number of plane, and the stride does not represent a capability change. Same caps
	  can have different stride from the default GstVideoInfo and the number of planes will
	  never change for 1 format.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-04 16:23:18 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Move the extrapolation of stride at the right place
	  Now that we have a stride array, we should extrapolate only when
	  eeded (non multi-planar buffer).
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-04 15:09:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Move back assertions where they should be
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-04 15:09:10 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Move mplane logic into gst_v4l2_object_get_caps_info()
	  It makes the gst_v4l2_object_set_format() slightly simplier and will make that
	  logic reusable. Note that gst_v4l2_object_has_mplane() will always return the
	  same value for one device. There is no need to check against the caps as this
	  has already been done by _open.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-03 18:27:47 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Split _v4l2fourcc_to_video_format
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-02 18:05:11 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Request buffers only once
	  VIDIOC_REQBUFS allocates buffer, it has no place inside set_config. Also, some driver do
	  no allow multiple calls to this ioctl.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-02 15:26:50 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't validate dimension for encoded format
	  We set the dimensions just in case but don't validate them
	  afterwards. For some codecs the dimensions are *not* in the
	  bitstream, IIRC VC1 in ASF mode for example.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-11-28 17:10:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Quirks for dev without initial format
	  Most M2M have undefined behaviour initially when VIDIOC_G_FMT is called.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-11-28 17:09:26 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Add gst_v4l2_object_open_shared()
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-11-28 17:07:05 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	  v4l2object: Implement gst_v4l2_dup()
	  This will duplicated the FD from another object and copy over the probed result.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-11-28 16:59:59 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: make IO_MODE enum public
	  This is to allow adding a second io-mode property on M2M device like decoder so
	  input and output can be controlled separatly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-06-04 23:42:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: better handle quirks activation
	  This way we can activate deactivate those quirks all at once at one
	  place.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-06-04 23:34:04 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Fix h264 caps
	  V4L2_PIX_FMT_H264 is documentated as byte-stream (with start code). The ensure proper
	  negotiation with element like h264parse.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2013-12-06 14:44:51 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Split caps in different categories
	  This is need to correctly expose capabilities on specialized devices
	  like decoders and encoders.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720568

2014-01-10 14:16:00 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: don't leak TOC chapter list

2014-01-10 08:52:16 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: remove obsolete write-dummy-and-overwrite-on-eos code
	  The need for rewriting apparently is obsolete 0.10 leftover.
	  We now have caps for subtitles when we create the headers,
	  so we always write the correct data in the first place.

2014-01-09 23:55:16 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: remove duplicate assignment
	  Coverity CID 1151680

2014-01-09 18:25:04 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: write subtitle codec ID and data at start when known
	  This avoids issues with writing dummy data first, then having
	  to come back and write correct data later. Doing so prevents
	  the muxed stream from being actually streamable.
	  https://bugzilla.gnome.org/show_bug.cgi?id=712134

2014-01-09 17:32:15 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  configure: Include AvailabilityMacros.h for osxvideo check
	  Otherwise MAC_OS_X_VERSION_MIN_REQUIRED might not be defined

2014-01-09 11:56:31 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	  qtmux: respect the HDLR box string format for mov and isomedia
	  Mov spec says it uses a pascal style string, while isomedia uses
	  a null terminated one. Store the current atoms flavor into the HDLR
	  to be able to generate the correct output.
	  https://bugzilla.gnome.org/show_bug.cgi?id=705982

2014-01-08 11:28:04 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/matroska/matroska-mux.c:
	  Revert "matroskamux: Use the running time for container timestamps, not buffer timestamps"
	  This reverts commit b3aa8755fe07639f22e4104f4932d769d6c9075a.
	  We are already using the running-time because they were placed on the
	  buffers with gst_collect_pads_clip_running_time(). Arguably it would be
	  better to not modify the incomming buffers but collectpads seems to want
	  to use absolute timestamps from the buffers for finding the best buffer
	  (this can be changed with a custom compare function..).

2014-01-08 10:41:24 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  configure: Fix AC_COMPILE_IFELSE usage

2014-01-08 10:31:18 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  osxvideosink: Improve configure check for OSX >= 10.6
	  https://bugzilla.gnome.org/show_bug.cgi?id=721245

2014-01-07 12:13:51 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: remove unused list of decoders
	  remove list of decoders, which are already handled by the list of elements.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719938

2014-01-08 09:46:55 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Error out if ADPCM caps don't contain the layout field

2014-01-03 15:25:23 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Add support for g726 ADPCM
	  https://bugzilla.gnome.org/show_bug.cgi?id=720995

2014-01-07 15:04:02 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: use new method to get media-type
	  Use the new method to get the media type of a transport.

2014-01-06 21:12:17 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/wavparse.c:
	  wavparse: split the test
	  This way one failure won't shadow the other test and also if one fails we get
	  better disgnostics through the test-name.

2014-01-06 14:54:46 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Add HEVC / h265 support

2014-01-06 14:54:38 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Add HEVC / h265 support

2014-01-06 13:36:38 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  wavparse: remove ifdef'ed code
	  We do have adtl and cue parse as part of toc handling alreday. The fmt code is a left over from <0.10 times.

2014-01-06 13:32:58 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	* gst/wavparse/gstwavparse.c:
	  avidemux, waveparse: more logging for unhandled chunks
	  Always print a warning with the tag and if possible do a memdump.

2014-01-05 22:47:42 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avidemux: expose 'strn' - stream name - as title tag

2014-01-05 22:41:24 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avidemux: parse fuji strd
	  We can get maker, model and capture date from this chunk.
	  Fixes #636143

2014-01-05 21:46:33 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avidemux: ... and use the local api both times

2014-01-05 21:38:14 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avidemux: copy the riff api for ncdt into the element
	  This chunk is avi specific, no need to expose this as public api.

2014-01-05 10:28:21 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Add missing semicolon from last commit

2014-01-05 10:22:37 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Use the running time for container timestamps, not buffer timestamps
	  Buffer timestamps have no real meaning here, and for selecting the next
	  buffer we already use the running time anyway.

2014-01-04 21:34:38 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avi: use new riff api to extract nikon metadata
	  Fixes #636143

2013-11-01 16:41:43 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	  rtprtxsend/rtprtxreceive: generate gtk doc

2013-12-02 11:26:09 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  test/check: Verify rtprtxsend::ssrc-map property works as expected

2013-11-29 19:35:44 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxreceive.h:
	* tests/check/elements/rtpaux.c:
	* tests/check/elements/rtprtx.c:
	* tests/examples/rtp/client-rtpaux.c:
	  rtprtxreceive: modify to use a payload-type map like rtprtxsend

2013-11-29 19:58:26 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: do not keep history of packets with an unknown payload type
	  This allows to disable retransmission per payload type by not putting
	  a certain payload type in the map.

2014-01-02 15:18:52 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	* tests/check/elements/rtpaux.c:
	* tests/check/elements/rtpcollision.c:
	* tests/check/elements/rtprtx.c:
	* tests/examples/rtp/server-rtpaux.c:
	  rtprtxsend: Allow SSRC-multiplexing and multiple payload types in the original stream
	  Conflicts:
	  tests/examples/rtp/server-rtpaux.c

2013-11-25 15:00:45 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: Add an rtx-ssrc property to allow external control of the ssrc
	  This is useful when one needs to know the SSRC beforehands, so that it can
	  be used for SRTP for example.

2013-11-13 15:11:35 -0500  Torrie Fischer <torrie.fischer@collabora.co.uk>

	* tests/examples/rtp/.gitignore:
	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-rtpaux.c:
	* tests/examples/rtp/server-rtpaux.c:
	  examples: rtp: Add end-to-end rtpbin example with RTX elements
	  This example demonstrates how to use rtpbin with retransmission (rtx)
	  elements set in the place of rtpbin's "aux" elements in order to
	  enable RTP retransmission according to the rules of RFC4588.

2013-11-05 17:35:01 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* docs/design/Makefile.am:
	* docs/design/design-rtpauxiliary.txt:
	  doc: add design-rtpauxiliary.txt to describe how rtpbin deals with auxiliary elements

2014-01-02 14:48:49 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpsession.c:
	  session: also push EOS event to RTCP srcpad

2014-01-02 14:46:11 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: place SSRC in Retransmission event

2013-11-01 16:57:15 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/rtpaux.c:
	  tests/check: add rtpaux::test_simple_rtpbin_aux
	  It shows how to use "set-aux-receive" and "set-aux-send"
	  properties of rtpbin to set rtprtxsend and rtprtxreceive
	  Build 2 pipelines, one for rtpbin as a sender and one for
	  rtobin as a receive. Then transmit an audio stream.
	  It also drops some packets to activate restransmission and
	  check they are actually retransmited.

2013-11-01 17:09:42 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* tests/check/elements/rtpcollision.c:
	  tests/check: add rtpcollision::test_rtx_ssrc_collision unit test
	  check that rtxrtpsend changes its retransmission ssrc when
	  collision happens

2013-11-06 12:34:13 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests/check: add rtprtx::test_rtxreceive_data_reconstruction
	  This unit test verifies that retransmitted rtp packets coming out
	  of rtprtxreceive are the same as the original ones.

2013-11-05 09:33:51 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: use a realistic limit for the value of max-size-packets
	  G_MAXINT16 is chosen because if the queue contains more than
	  G_MAXINT16 packets, seqnum comparison will not work properly.

2013-11-04 20:05:03 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	  rtprtxsend: use a GSequence to implement the buffer queue
	  This has the advantage that searching the queue to find the
	  buffer with the requested seqnum is done with binary search.

2013-11-04 18:38:24 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	* tests/check/elements/rtprtx.c:
	  rtprtxsend: retransmit packets in the same order as the rtx requests

2013-11-02 19:56:44 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests/check: Add unit test for rtxsend's max_size_time property

2013-10-29 18:27:00 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	  rtprtxsend: Handle the max_size_time property
	  This property allows you to specify the amount of buffers
	  to keep in the retransmission queue expressed as time (ms)
	  instead of buffer count (which is the max_size_buffers property).

2013-11-02 15:21:08 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: keep important buffer information in a private structure
	  This is to avoid mapping a buffer every time we need to read a seqnum
	  or a timestamp.

2013-11-01 11:58:47 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests/check: Add rtprtx::test_rtxsender_packet_retention
	  This unit test verifies that the rtxsend element correctly maintains
	  a buffer of already transmitted rtp packets and that it can
	  re-transmit all of them correctly on demand. It also verifies
	  that the limit of this buffer (max-size-packets property) is respected.

2013-11-01 16:22:13 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* tests/check/elements/rtprtx.c:
	  tests/check: add rtprtx::test_drop_multiple_sender unit test
	  Several senders / one receiver
	  Similar than test_drop_one_sender but with multiple senders
	  mixed through the funnel element.
	  It drops some packets and checks that they are retransmited
	  correctly.

2013-11-01 16:21:00 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* tests/check/elements/rtprtx.c:
	  tests/check: add rtprtx::test_drop_one_sender unit test
	  Test for one sender / one receiver
	  Build the pipeline
	  videotestsrc ! rtpvrawpay ! rtprtxsend ! rtprtxreceive ! fakesink
	  and drop some buffers between rtprtxsend and rtprtxreceive
	  Then it checks that every dropped packet has been re-sent.
	  It also checks that not too much requests has been sent.

2013-11-01 16:17:51 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/rtprtx.c:
	  tests/check: add rtprtx::test_push_forward_seq
	  add simple unit test that manually push buffers
	  in rtprtxsend connected to rtprtxreceive.
	  Drops some buffers and make sure they are retransmisted.

2013-11-01 15:52:03 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/rtpmanager/gstrtprtxreceive.h:
	* gst/rtpmanager/gstrtprtxsend.c:
	* gst/rtpmanager/gstrtprtxsend.h:
	  rtpmanager: add new rtprtxsend / rtprtxreceive elements
	  The purpose of the sender RTX object is to keep a history
	  of RTP packets up to a configurable limit (in time). It will
	  listen for custom retransmission events from downstream. When
	  it receives a request for retransmission, it will look up the
	  requested seqnum in its list of stored packets. If the packet
	  is available, it will create a RTX packet according to RFC 4588
	  and send this as an auxiliary stream.
	  The receiver will listen to the custom retransmission events
	  from the downstream jitterbuffer and will remember the SSRC1
	  of the stream and seqnum that was requested. When it sees a
	  packet with one of the stored seqnum, it associates the SSRC2
	  of the stream with the SSRC1 of the master stream. From then
	  on it knows that SSRC2 is the retransmission stream of SSRC1.
	  This algorithm is stated in RFC 4588. For this algorithm to
	  work, RFC4588 also states that no two pending retransmission
	  requests can exist for the same seqnum and different SSRCs or
	  else it would be impossible to associate the retransmission with
	  the original requester SSRC.
	  When the RTX receiver has associated the retransmission packets,
	  it can depayload and forward them to the source pad of the element.
	  RTX is SSRC-multiplexed
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084

2013-11-05 16:36:46 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* docs/design/Makefile.am:
	* docs/design/design-rtpretransmission.txt:
	  doc: add design for rtp retransmission
	  Describe how rtprtxsend and rtprtxreceive generally work
	  but also how the association algorithm is implemented.

2014-01-02 20:23:05 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: use status code macro instead of 407
	  Rest of the code is using the _PROXY_AUTHENTICATION_REQUIRED
	  macro too. Easier to understand if you don't recall HTTP
	  error codes by heart.

2013-12-31 21:31:43 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2send: change audio_format field to format
	  This element and the underlying libshout2 library
	  can handle video media files too. The code already
	  handles video/webm so the name gets confusing. Also
	  add and use DEFAULT_FORMAT macro Instead of hardwiring
	  SHOUT_FORMAT_VORBIS at init
	  https://bugzilla.gnome.org/show_bug.cgi?id=721342

2013-12-31 20:09:29 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* ext/shout2/gstshout2.c:
	  shout2send: clarify meaning of the URL prop
	  https://bugzilla.gnome.org/show_bug.cgi?id=721342

2013-12-27 12:27:32 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* ext/shout2/gstshout2.c:
	  shout2send: docs, add a sample pipeline
	  And finish adding shout2send to the docs while at it
	  https://bugzilla.gnome.org/show_bug.cgi?id=721342

2013-12-31 15:00:22 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: remove spurious @see_also

2013-12-06 17:08:54 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: support any video formats and any caps features if deinterlace mode allows it
	  https://bugzilla.gnome.org/show_bug.cgi?id=719636

2013-12-31 13:31:52 +0100  Sebastian Rasmussen <sebras@hotmail.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Handle v4l2_ioctl() errors even in error handling
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721268

2014-01-01 12:11:43 -0800  Jeremy Huddleston Sequoia <jeremyhu@apple.com>

	* sys/osxvideo/Makefile.am:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideo: unifdef -DRUN_NS_APP_THREAD

2014-01-01 12:10:01 -0800  Jeremy Huddleston Sequoia <jeremyhu@apple.com>

	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	  osxvideo: Assume SDK and deployment target are at least Snow Leopard

2014-01-01 12:23:50 -0800  Jeremy Huddleston Sequoia <jeremyhu@apple.com>

	* configure.ac:
	  configure: Disable osxvideo on Leopard and earlier
	  This also moves the "other platforms" check in OS X video to before the
	  variable is read
	  https://bugzilla.gnome.org/show_bug.cgi?id=721245

2013-12-31 14:57:27 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/rtpbin.c:
	  tests: add AUX receiver unit test

2013-12-31 13:20:01 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/rtpbin.c:
	  tests: improve rtpbin test

2013-12-31 13:16:46 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: add some docs about AUX elements

2013-12-31 13:01:22 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/rtpbin.c:
	  tests: add AUX sender unit test

2013-12-31 12:31:25 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: add support for AUX sender and receiver
	  AUX elements are elements that can be inserted into the rtpbin
	  pipeline right before or after 1 or more session elements.
	  The AUX elements are essential for implementing functionality such
	  as error correction (FEC) and retransmission (RTX).
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087

2013-12-31 12:22:39 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/rtpbin.c:
	  tests: add decoder test

2013-12-30 17:36:42 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: make request_element method internally
	  We can use the same method to create encoder and decoder elements, they
	  are just internal elements that we create.

2013-12-31 10:25:28 +0100  Stéphane Cerveau <scerveau@gmail.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Skip id3 tag
	  Skip id3 tag during wav parse.
	  https://bugzilla.gnome.org/show_bug.cgi?id=721241

2013-12-31 10:10:05 +0100  Sebastian Dröge <sebastian@centricular.com>

	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	  osx: Make OSX version checks more consistent
	  And especially also consider update versions, e.g. 10.5 with updates
	  will be 1051 or similar and thus bigger than MAC_OS_X_VERSION_10_5 but
	  still won't have the API we want to use.

2013-12-31 10:07:22 +0100  Jeremy Huddleston <jeremyhu@freedesktop.org>

	* sys/osxvideo/osxvideosink.h:
	  osxvideosink: Fix build on updated OS X Leopard
	  https://bugzilla.gnome.org/show_bug.cgi?id=721245

2013-12-30 17:23:22 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavimux.c:
	  avimux: Add missing break
	  I guess no-one noticed we no longer could mux WMV3 ...
	  COVERITY CID 1139759

2013-12-30 17:20:37 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: Add missing break
	  COVERITY CID 1139762

2013-12-30 17:00:45 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: internal-ssrc is no longer deprecated

2013-12-30 16:59:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: add Since tags

2013-12-30 16:52:28 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: add signal for new jitterbuffer
	  Emit a signal when a new jitterbuffer is created so that the app can
	  have a chance to configure it.

2013-12-30 16:28:57 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	* tests/check/elements/rtpbin.c:
	  rtpbin: handle multiple encoder instances
	  Keep track of elements that are added to multiple sessions and make sure
	  we only add them to the rtpbin once and that we clean them when no
	  session refers to them anymore.

2013-12-30 15:16:09 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/rtpbin.c:
	  tests: add unit test for encoder element

2013-12-30 15:15:43 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix memory leaks

2013-12-30 15:03:34 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/rtpbin.c:
	  tests: fix leak

2013-12-30 15:00:50 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: expect the pads on the encoders
	  Don't use request pads for the encoder elements, the signal handler
	  should request the pads and make sure they are available with the right
	  name.

2013-12-30 14:56:07 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: request-rtp-encoder are no action signals
	  The request-rtp-encoder signals are not action signals so mark them
	  correctly and use an accumulator to collect the result value.

2013-12-30 14:36:45 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  wavparse: emit midi-base-note tag from data in 'smpl' chunk
	  Add parsing of the 'smpl' chunk. Right now we only grab the midi-base-note and
	  emit it as a tag.

2013-12-26 12:05:19 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision
	  When a collision is found on the internal ssrc, we have to change it.
	  Ideally, we want also the payloader upstream to follow this change and use
	  the new internal ssrc. Ideally we want this condition to be always met:
	  if there is one payloader sending on this session, its ssrc should match the
	  internal ssrc.

2013-12-26 11:04:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: allow setting internal-ssrc again

2013-12-30 13:31:45 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/y4m/gsty4mencode.c:
	  y4mencode: Remove dead code
	  set/get property isn't used

2013-12-30 13:30:24 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpqcelpdepay.c:
	  rtpqcelpdepay: Remove uneeded variable

2013-12-05 15:53:52 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>

	  rtpbin: allow dynamic RTP/RTCP encoders/decoders
	  * gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
	  added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
	  and request-rtcp-decoder). The user will be able to provide encoders
	  or decoders dynamically. The encoders must follow the srtpenc API and
	  the decoders the srtpdec API. Having separate signals for RTP and RTCP
	  allows the user to use different encoders/decoders or provide the same
	  one (e.g. that would be the case for srtpenc).
	  Also, rtpbin now allows application/x-srtp in its pads.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719938

2013-12-27 16:51:32 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: dynamically recalculate RTX parameters
	  Use the round-trip-time and average jitter to dynamically calculate the
	  retransmission interval and expected packet arrival time.
	  Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412

2013-12-27 16:50:52 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: calculate average jitter

2013-12-27 16:48:48 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: use RTT from the Retransmission event
	  Place the estimated RTT in the Retransmission event and let the session
	  manager use that instead of the hardcoded value.

2013-12-27 15:57:39 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: take more accurate running-time for NACK
	  Don't use the current time calculated from the tmieout loop for when we
	  last scheduled the NACK because it might be unscheduled because of a max
	  packet misorder and then we don't accurately calculate the current time.
	  Instead, take the current element running time using the clock.

2013-12-30 11:06:38 +0100  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/wavpackdec.c:
	  wavpackdec: Send a CAPS event in the unit test

2013-12-27 02:14:02 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: improve mss_mode/fragmented special handling
	  Make it clear what should be handled purely by mss mode:
	  1) Expose the streams on the first moof as there are no moov atoms
	  2) Properly cleanup streams on flushes
	  Add a note about the meaning of upstream_newsegment and mss_mode
	  for future reference.
	  Make all other special fragment handling shared for both dash
	  and mss streams.

2013-12-12 10:50:27 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: drain the adapter before pushing EOS
	  In a fragmented scenario, qtdemux is operating in push mode
	  and it gets a fragmented buffer. While processing its data
	  downstream gets unlinked (or a input-selector changes its
	  active pad and returns not-linked). Qtdemux stops processing
	  this fragment and returns not-linked upstream, leaving the
	  remaining data in its adapter.
	  When it gets an EOS it should make sure that all the data it
	  had received is pushed before pushing EOS.

2013-12-26 23:21:47 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* ext/shout2/gstshout2.c:
	  shout2send: drop IP only requirement for _set_host()
	  libshout2 (we require > 2.0 at config time) supports
	  both IP and hostname for _set_host(). Dropped an
	  outdated FIXME regarding this limitation, adjusted
	  some comments and changed the param blurb to reflect
	  this too.

2013-12-26 21:43:34 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>

	* ext/shout2/gstshout2.c:
	  shout2send: Retarget FIXME to 2.0

2013-12-26 11:21:36 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: use aggregate control for PLAY/PAUSE/TEARDOWN
	  Use the aggregate control instead of the original request url to perform
	  PAUSE/PLAY and TEARDOWN.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721003

2013-12-24 14:40:25 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: Proxy CAPS, ALLOCATION, SCHEDULING and srcpad events properly

2013-12-24 00:43:39 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: adpcm max block align is 8192

2013-12-23 12:23:27 -0600  Brendan Long <b.long@cablelabs.com>

	* configure.ac:
	  vp9dec: Require vpx >= 1.3.0 for building vp9dec and vp9enc
	  Previous versions did not have a stable bitstream for VP9.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720986

2013-12-23 15:46:48 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Use correct codec id for ADPCM/DVI

2013-12-23 15:44:30 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Check for the correct size of codec_data in the ACM case

2012-01-14 19:58:17 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: basic adpcm support
	  https://bugzilla.gnome.org/show_bug.cgi?id=664339

2013-12-20 11:45:38 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/descriptors.c:
	  qtdemux: Fix calcuation of descriptor length
	  https://bugzilla.gnome.org/show_bug.cgi?id=720813

2013-12-22 22:33:39 +0000  Tim-Philipp Müller <tim@centricular.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From dbedaa0 to d48bed3

2013-12-22 21:56:03 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/Makevars:
	  po: set gettext domain in Makevars so we don't have to patch the generated Makefile.in.in
	  https://bugzilla.gnome.org/show_bug.cgi?id=705455

2013-12-19 16:50:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: on receive error only unmap and unref buffer if one was alloced and mapped
	  coverity CID 1139866.

2013-12-19 12:47:22 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: fix misleading comment
	  Those are not allocated on the stack.

2013-12-17 18:28:25 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  vpx: Mark VP9 support as non-experimental
	  There was a libvpx release with VP9 support now and the bitstream
	  is frozen too.

2013-12-15 21:04:11 -0800  Todd Agulnick <todd@agulnick.com>

	* gst/deinterlace/gstdeinterlace.c:
	  Some compiler warning fixes to satisfy XCode compiler
	  https://bugzilla.gnome.org/show_bug.cgi?id=720513

2013-12-16 16:17:07 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: Set picture type in the APIC frames

2013-12-16 16:14:52 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: Set image-description from the info struct, not the caps

2013-12-16 10:02:37 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	* gst/audioparsers/gstwavpackparse.h:
	  wavpackparse: Post AUDIO_CODEC tag

2013-12-16 10:00:37 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstsbcparse.h:
	  sbcparse: Post AUDIO_CODEC tag

2013-12-16 09:58:31 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: Post AUDIO_CODEC tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=720512

2013-12-16 09:56:29 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstdcaparse.h:
	  dcaparse: Post AUDIO_CODEC tag

2013-12-16 09:54:38 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstamrparse.h:
	  amrparse: Post AUDIO_CODEC tag

2013-12-16 09:49:48 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	  ac3parse: Post AUDIO_CODEC tag

2013-12-16 09:46:16 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	  aacparse: Post AUDIO_CODEC tag

2013-12-16 09:41:14 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: Use pbutils functionality to create the AUDIO_CODEC tag

2013-12-13 17:36:36 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Add error message if the app tries to set the internal-ssrc

2013-12-13 16:08:35 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Only count nacks when a nack packet is received
	  Not when any RTCP feedback packet is.

2013-12-12 23:22:41 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpcollision.c:
	  tests: Initialize segment in rtpcollision test

2013-12-13 15:57:36 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Process PSFB FIR requests which lack the media ssrc
	  According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
	  have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
	  So in that case, we ignore the retained feedback and just let it through
	  to the rtp_session_process_fir() function which will check for the actual
	  SSRC inside the FCI.
	  Fixes a regression introduced by commit 57c27ec3

2013-11-14 16:19:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders
	  Previously, when the session had multiple internal sender SSRCs, it would
	  issue SR reports with RB blocks only on the first RTCP timeout and afterwards
	  SR reports would be sent empty. This was because the "generation" number
	  in RTPSource would increase more than once during the same cycle and afterwards
	  it would always be greater than the session's generation, which would cause
	  it to be skipped from being included in RBs.
	  This commit fixes this problem by:
	  1) Increasing the RTPSource generation only at the end of each cycle,
	  which essentially fixes the problem but only when the internal senders
	  are less than GST_RTCP_MAX_RB_COUNT.
	  2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
	  SR the given RTPSource has been reported in, which also fixes the problem
	  when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
	  necessary because of the fact that any RTPSource is marked as reported
	  in itself's SR and makes it impossible to know if it has been reported
	  in other SRs too or not, and which.

2013-11-14 16:23:35 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtpsession.c:
	  tests/check: add an rtpsession unit test to verify all RBs are included in all SRs, roundrobin
	  This test checks that when we have multiple internal sender sources
	  in rtpsession, SRs contain RBs for every other sender source, and that
	  they are included roundrobin when they exceed ST_RTCP_MAX_RB_COUNT,
	  which is the max number of RBs that can fit in a SR.

2013-12-12 16:01:10 +0100  Wim Taymans <wtaymans@redhat.com>

	* docs/design/design-rtpcollision.txt:
	  docs: improve docs

2013-11-05 18:03:48 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* docs/design/Makefile.am:
	* docs/design/design-rtpcollision.txt:
	  doc: add design-rtpcollision.txt that explains when GstRTPCollision is created
	  It also talks about "BYE only the corresponding source, not the whole session."

2013-11-05 12:31:54 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* tests/check/elements/rtpcollision.c:
	  tests/check: improve rtpcollision::test_master_ssrc_collision to ensure that a collision does not BYE the whole session
	  Conflicts:
	  tests/check/elements/rtpcollision.c

2013-11-01 17:07:57 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/rtpcollision.c:
	  tests/check: add rtpcollision::test_master_ssrc_collision unit test
	  It checks the payloader changes its ssrc when collision happens

2013-12-12 10:38:43 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: keep extra stats for scheduling BYE
	  Keep an extra stats structure for scheduling the BYE packets. When we
	  decide to schedule BYE, make a copy of the current stats into the
	  bye_stats. Then while we schedule the BYE, update and use only the
	  bye_stats. When we finished scheduling the BYE packet, we use the
	  regular stats again.

2013-12-12 10:34:38 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: when we schedule BYE, only deal with BYE sources
	  When we are doing the RTCP timeout to schedule BYE packets, don't
	  generate RTCP for all sources but only for the sources marked as BYE.

2013-12-12 10:32:48 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: reset state after scheduling BYE
	  After we do RTCP, we are not scheduling bye anymore.

2013-12-12 10:31:38 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: also count NACKS when no signal was pending

2013-12-12 10:09:25 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  session: ignore RTCP packets for the BYE sources
	  When we are scheduling BYE packets, ignore all RTCP for the sources that
	  are scheduling a BYE packet. Other sources that are not scheduling BYE
	  should continue receiving RTCP packets as usual.

2013-11-04 11:48:21 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: determine if the session is doing point-to-point
	  In this case T_dither_max is set to 0 according to RFC 4585

2013-12-10 11:57:37 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: serialize events in the buffer
	  Serialize events into the jitterbuffer by inserting them with a -1
	  seqnum.
	  Update unit test to expect events from the streaming thread.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986

2013-12-10 11:04:06 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: detect -1 seqnum
	  Keep the seqnum as a full guint so that we can check for -1 entries and
	  deal with them correctly.
	  Immediately try to push -1 seqnum.

2013-12-10 11:01:03 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: reorganize jitterbuffer items
	  Keep the oldest item at the head and the newest items on the tail. This
	  makes it easier to deal with -1 seqnums.

2013-12-09 23:34:10 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: correctly check for invalid values
	  Check for -1 on the guint from the buffer item instead of on the guint16
	  or guint32.
	  Also insert -1 seqnum at the head of the jitterbuffer.

2013-12-08 16:49:55 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix segfault when dealing with padded frames
	  Fixes crashes with vtdec ! osxvideosink where VideoToolbox outputs padded UYVY

2013-12-06 17:58:13 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/audiofx/gststereo.c:
	  stereo: Port to GStreamer 1.0 API

2013-12-05 12:15:29 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/law/mulaw-decode.c:
	  mulawdec: Require caps to be set before accepting any data

2013-12-05 12:15:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: Require caps to be set before accepting any data

2013-12-05 12:13:33 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/speex/gstspeexdec.c:
	  speexdec: Require caps to be set before accepting any data

2013-12-05 12:13:10 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Require caps to be set before accepting any data

2013-12-05 11:42:15 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  vpx: Use new gst_video_decoder_set_needs_format() API

2013-12-04 16:23:43 -0500  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Free device_info in accepts caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=719811

2013-12-04 21:57:48 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: Don't send headers twice if we got them from the caps already

2013-12-04 21:57:04 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: Don't leak config data when receiving a second CAPS event

2013-12-04 21:55:53 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: Don't send headers twice if we got them from the caps already

2013-12-04 21:54:16 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: Don't leak config data when receiving a second CAPS event

2013-12-04 21:17:03 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpstreamdepay.c:
	* gst/rtp/gstrtpstreamdepay.h:
	  rtpstreamdepay: Add RFC4571 RTP stream depayloading element
	  https://bugzilla.gnome.org/show_bug.cgi?id=719829

2013-12-04 10:12:46 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpstreampay.c:
	* gst/rtp/gstrtpstreampay.h:
	  rtpstreampay: Add RFC4571 RTP stream payloading element
	  https://bugzilla.gnome.org/show_bug.cgi?id=719829

2013-12-03 15:08:25 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: improve fragment-start tracking
	  Some buffers can have multiple moov atoms inside and the strategy
	  of using the gst_adapter_prev_pts timestamp to get the base timestamp
	  for the media of the fragment would fail as it would reuse the same
	  base timestamp for all moofs in the buffer instead of accumulating
	  the durations for all of them.
	  Heres a better explanation of the issue:
	  qtdemux receives a buffer where PTS(buf) = X
	  buf -> moofA | moofB | moofC
	  The problem was that PTS(buf) was used as the base timestamp for
	  all 3 moofs, causing all buffers to be X based. In this case we want
	  only moofA to be X based as it is what the PTS on buf means, and the
	  other moofB and moofC just use the accumulated timestamp from the
	  previous moofs durations.
	  To solve this, this patch uses gst_adapter_prev_pts distance
	  result, this allows qtdemux to calculate if it should use the
	  resulting pts or just accumulate the samples as it can identify
	  if the moofs belong to the same upstream buffer or not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719783

2013-11-21 12:29:28 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: add support for multi-planar V4l2 API in DMABUF mode
	  Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754

2013-11-19 17:16:27 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: refactor by emulating one v4l2_plane in non-MPLANE mode
	  so that the buffer informations can be retrieved the same way
	  in both MPLANE and non-MPLANE mode.
	  Here "emulating" means "manually fill in the plane".
	  Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754

2013-11-13 12:05:40 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: add support for multi-planar V4L2 API
	  This api is in linux kernel since version 2.6.39,
	  and present in all version 3.
	  The commit that adds the API in master branch of the
	  linux kernel source is:
	  https://github.com/torvalds/linux/commit/f8f3914cf922f5f9e1d60e9e10f6fb92742907ad
	  v4l2 doc: "Some devices require data for each input
	  or output video frame to be placed in discontiguous
	  memory buffers"
	  There are newer structures 'struct v4l2_pix_format_mplane'
	  and 'struct v4l2_plane'.
	  So the pixel format is not setup with the same API when using
	  multi-planar.
	  Also for gst-v4l2, one of the difference is that in GstV4l2Meta
	  there are now one mem pointer for each maped plane.
	  When not using multi-planar, this commit takes care of keeping
	  the same code path than previously. So that the 2 cases are
	  in two different blocks triggered from V4L2_TYPE_IS_MULTIPLANAR.
	  Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754

2013-12-04 09:12:07 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: don't leak template caps

2013-12-03 21:41:28 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	* tests/check/elements/aacparse.c:
	  audioparsers: use ACCEPT_INTERSECT flag
	  The parser can accept input that is not completely specified. Use the
	  ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
	  check for intersection only. This allows us to proxy downstream
	  constraints while still allowing non-subset caps as input.
	  We can then also remove the appended template caps workaround.
	  Make a unit-test to check the new feature.
	  This reverts commit 26040ee38cb9e7c42f3d9a0282b3e5cace7ca42d
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024

2013-12-03 21:36:54 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: remove fields from filter
	  We need to remove the fields from the filter when we can convert
	  between them.

2013-12-03 21:29:13 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: refactor code to remove caps fields

2013-12-02 00:10:43 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: microoptimisation: avoid some unnecessary GValue copies

2013-12-01 23:32:20 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix off-by-one crash when downstream caps contain a list of framerates
	  https://bugzilla.gnome.org/show_bug.cgi?id=719544

2013-11-29 11:26:05 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use the timestamp of the moof as the base fragment start
	  In SmoothStreaming fragmented scenario, the timestamps are calculated
	  starting from the fragment buffer timestamp. When there is a not-linked
	  return from downstream, qtdemux will return upstream and will keep the
	  non-pushed data into its adapter.
	  On a new fragment buffer pushed to qtdemux, the new buffer timestamp
	  would overwrite the previous one that should be used on the still
	  to be pushed buffers. Because of this, this patch will also
	  update the fragment_start timestamp from the adapter last pts
	  to make sure the moof and timestamps are in sync and will result
	  in correct timestamps for all fragments.

2013-11-15 08:54:07 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: avoid re-reading the same moov and entering into loop
	  In the scenario of "mdat | moov (with fragmented artifacts)" qtdemux
	  could read the moov again after the mdat because it was considering the
	  media as a fragmented one.
	  To avoid this loop this patch makes it store
	  the last processed moov_offset to avoid parsing it again.
	  And it also checks if there are any samples to play before
	  resturning to the mdat, so that it knows there is new data to be played.
	  https://bugzilla.gnome.org/show_bug.cgi?id=691570

2013-11-15 00:52:53 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: do not free streams if they were not created locally
	  When parsing a trak only free streams on failures if those streams
	  were created locally. They could have been created from a previous
	  fragment, in this case we they have valid info from the other fragment.
	  Including pads.
	  https://bugzilla.gnome.org/show_bug.cgi?id=691570

2013-11-29 19:57:46 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/blend.c:
	  videomixer: Simplify NV12/21 blending code macros

2013-11-29 19:50:24 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/videomixer/blend.c:
	  videomixer: Fix segfault when filling the background of a UYVY frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=712401

2013-11-29 09:21:52 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix compilation with gst debuging disabled
	  qtdemux.c:9452:1: error: label at end of compound statement

2013-11-27 17:02:00 +0100  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Map inbuffer once only
	  Do not call gst_buffer_extract() twice since each call will map and
	  unmap the biffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719434

2013-11-28 11:58:42 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/videofilter.c:
	  videoflip: Add unit test for the 'automatic' method
	  These new tests send a tag event before seding the buffer. Tested case are an
	  empty tag list, a tag list with orientation-180 set and an invalid orientation value.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719497

2013-11-28 16:09:04 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: don't crash on tag events without orientation tag
	  Would crash in g_free() trying to free an uninitialised pointer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=719497

2013-11-28 16:50:42 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: don't unref buffer twice
	  Cleaning the packet info will already unref the buffer.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078

2013-11-28 22:35:02 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add HydrogenAudio ReplayGain tags
	  Identical to the itunes (tm) version, but labelled with
	  org.hydrogenaudio.replaygain as the producer.

2013-11-27 16:15:12 +0100  Mathieu Duponchelle <mduponchelle1@gmail.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: explicitly fail when alpha information would have been lost.

2013-05-29 16:06:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* .gitignore:
	  gitignore: Updated to ignore *.swp and .dirstamp

2013-11-26 11:17:42 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Allow a bit more variation when detecting common framerates
	  Instead of +/- 1ns we allow 2ns now. Due to rounding errors there are
	  some Matroska files out there with 33.333331ms per frame for 30fps.

2013-11-26 10:20:31 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Use gst_util_double_to_fraction() instead of GValue magic

2013-11-25 14:03:21 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Set default method at contruction
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712333

2013-05-29 15:57:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Use space instead of tabs
	  https://bugzilla.gnome.org/show_bug.cgi?id=712754

2013-05-29 15:44:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Fix header indentation so it's readable again
	  It's unfortunate to have to do this, but with the mix of tabs and space, plus all the random
	  indentation this header has become very hard to read.
	  https://bugzilla.gnome.org/show_bug.cgi?id=712754

2013-11-25 17:38:06 +0100  Wim Taymans <wtaymans@redhat.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  check: fix jitterbuffer check
	  Don't advance the clock to 240ms too early.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710013

2013-11-25 11:45:33 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: deprecate max-errors
	  The property wasn't use internally, let the base class handle the
	  number of errors to tolerate.

2013-11-25 15:49:07 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: improve clear-pt-map handling
	  Don't reset the expected output seqnum when clearing the pt map because this
	  could stall the jitterbuffer forever.
	  Add a unit test for this.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800

2013-10-28 21:33:22 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: let the base class decide when to return an error
	  The base videodecoder class has an error counting feature to tolerate
	  a few errors before posting an error message. So don't force the
	  error and let the base class decide when it should happen
	  https://bugzilla.gnome.org/show_bug.cgi?id=710762

2013-10-28 21:28:33 -0300  Thiago Santos <ts.santos@sisa.samsung.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Add data skipping on input
	  Add missing bytes skipping when bad input is received.
	  https://bugzilla.gnome.org/show_bug.cgi?id=710762

2013-11-25 12:13:43 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Discard 2 byte subpicture packets
	  As for text subtitles and as suggested in #712643, throw
	  away the 2 byte terminator packets that some encoders insert.
	  This will make things better when remuxing and causes generation
	  of gap events.

2013-11-25 00:34:21 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix wake-up when new buffers come in after running empty
	  Spotted by 'gratias' on IRC. Probably introduced in recent refactoring.
	  https://bugzilla.gnome.org/show_bug.cgi?id=715039

2013-11-23 12:15:40 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: correctly handle negative relative timestamps
	  ... rather than scaling these as unsigned.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712744
	  Based on patch by Krzysztof Kotlenga <pocek@users.sf.net>

2013-09-14 03:27:09 +0200  MathieuDuponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer2: Merge tag events to send them in collected.
	  Otherwise there were race conditions where we would send tags
	  on a flushing srcpad.
	  We have a test for that in GES, but this should be tested
	  systematically with harness in the future as I believe it
	  is useful for exactly that kind of cases.
	  https://bugzilla.gnome.org/show_bug.cgi?id=708165

2013-11-14 17:29:50 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use GstVideoInfo helper to create caps for raw video
	  This way we do not miss mandatory fields in caps.
	  At the same time use the gst_pb_utils_get_codec_description
	  helper to get codec description.
	  https://bugzilla.gnome.org/show_bug.cgi?id=712335

2013-11-14 16:11:38 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/matroska/Makefile.am:
	* gst/matroska/matroska-demux.c:
	  matroskademux: Use GstVideoInfo helper to create caps for raw video
	  This way we do not miss mandatory fields in caps.
	  At the same time use the gst_pb_utils_get_codec_description helper to
	  get codec description.
	  https://bugzilla.gnome.org/show_bug.cgi?id=712328

2013-11-13 20:18:17 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstmultifilesrc.h:
	  multifilesrc: Implement seeking in case of multiple images
	  https://bugzilla.gnome.org/show_bug.cgi?id=712254

2013-11-22 12:26:21 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: pass downstream flowreturn to upstream
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712722

2013-11-18 14:27:48 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: clear cached caps on close
	  A different device with different caps may be used for the next open.
	  https://bugzilla.gnome.org/show_bug.cgi?id=712611

2013-11-21 15:30:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/wavpack/gstwavpackcommon.c:
	* ext/wavpack/gstwavpackstreamreader.c:
	* gst/apetag/gstapedemux.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/isomp4/atoms.c:
	* gst/matroska/matroska-demux.c:
	  g_memmove() is deprecated
	  Just use plain memmove(), g_memmove() is deprecated in
	  recent GLib versions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=712811

2013-11-21 11:32:15 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbisdepay: handle packets > 0xffff
	  Handle input packet sizes larger than 16 bits in the depayloader.
	  Remove size restrictions on the payloader.

2013-11-21 11:30:28 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	  rtptheoradepay: handle packets > 0xffff
	  Reorganize some things in the depayloader so that it can handle packets larger
	  than 16 bits.
	  Remove the size restriction on the payloader.

2013-11-21 02:28:27 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_types.c:
	  isomp4: Handle mp4s subpicture streams better.
	  Clean up the handling of mp4s streams. Use the generic esds
	  descriptor function to extract the palette, instead of hard coding
	  a wrong magic offset.
	  Add some more size safety checks when parsing ES descriptors, and
	  replace magic numbers with the descriptive constants that are already
	  defined.
	  Enhance dump output for stsd atoms.
	  Streams from both bug 712643 and historic bug 568278 now both work
	  correctly.
	  Fixes: #712643

2013-11-20 22:08:25 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/isomp4/fourcc.h:
	  qtdemux: Sort fourcc declarations and remove duplicates

2013-11-20 21:41:47 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/isomp4/Makefile.am:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/ftypcc.h:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Merge all the fourcc headers into one
	  Remove qtdemux_fourcc.h and ftypcc.h and put it all in fourcc.h

2013-11-19 10:10:51 +0100  Wim Taymans <wim.taymans@gmail.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: avoid mapping the buffer
	  Reuse the parsed structure to get the timestamps.

2013-11-18 17:13:49 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix 'make check'
	  Fix generic/states check. Also, g_return_if_fail() is
	  not for internal state checking.

2013-11-18 14:44:36 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiopanorama.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/flv/gstflvmux.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/videobox/gstvideobox.c:
	* gst/wavparse/gstwavparse.c:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/ximage/gstximagesrc.c:
	  docs: get rid of 'Since: 0.10.x' markers
	  And some gtk-doc markup fixes.

2013-11-16 12:15:14 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: fix Since markers
	  Should be next stable release series version

2013-11-15 13:48:07 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Fix stats property field names and documentation

2013-11-15 15:20:14 +0100  Torrie Fischer <torrie.fischer@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  gstrtpsession: Implement a number of feedback packet statistics
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693

2013-11-13 17:11:08 -0300  Thiago Santos <ts.santos@partner.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: remove math operation from loop
	  The elst_offset doesn't change inside the loop, so compute it
	  outside

2013-11-14 20:54:32 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/isomp4/qtdemux.c:
	  qtmux: fix playback regression
	  In ae1150e85cf99d7482933aa6f7e4f012fe45a3ec flipping a condition misaligned the
	  else branch, where for there condition that was change there is none.
	  Fixes #712303

2013-11-14 09:20:06 +0100  Wim Taymans <wim.taymans@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: rename property to 'stats'
	  This makes the unit test work.
	  We can later also add more stats, not specific to retransmission.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711411

2013-11-12 11:19:25 -0500  Torrie Fischer <torrie.fischer@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: implement rtx statistics

2013-11-13 10:42:21 +0000  Marc Leeman <marc.leeman@gmail.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: print FOURCC_FORMAT when enumerating
	  https://bugzilla.gnome.org/show_bug.cgi?id=712206

2013-11-06 12:40:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: advance expected seqnum after dropping
	  After dropping a buffer, move our expected seqnum
	  Conflicts:
	  gst/rtpmanager/gstrtpjitterbuffer.c

2013-11-04 15:46:22 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  gstpay: only send one caps
	  Only send one caps in a packet. Two caps can happen when setcaps is called and
	  the config-interval expires at the same time.

2013-11-13 10:23:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Use the synced buffer mode in auto mode if a clock provider is in the SDP

2013-11-08 11:09:21 +0000  Marc Leeman <marc.leeman@gmail.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: init v4l2_buffer to 0x0 before ioctl
	  https://bugzilla.gnome.org/show_bug.cgi?id=712137

2013-11-11 15:27:18 +0100  Wim Taymans <wim.taymans@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: remove collision reconfigure event
	  Remove bogus reconfigure event on collision, we don't want to send the event on
	  the receiving RTP pad and the collision event is now handling this
	  case.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=711560

2013-11-01 17:04:28 +0000  Julien Isorce <julien.isorce@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: send custom upstream event "GstRTPCollision" on send_rtp_sink pad
	  See https://bugzilla.gnome.org/show_bug.cgi?id=711560

2013-11-11 14:25:51 +0100  Wim Taymans <wim.taymans@gmail.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/rtpsession.c:
	  check: add rtpsession test
	  Add a basic rtpsession test to ensure that RR blocks are generated when
	  multiple SSRC senders are active.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711270

2013-11-11 13:17:25 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: correctly handle timestamps when parsing x-private1-ac3
	  ... the way it has always worked fine in a52dec.

2013-11-05 10:48:33 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix crash when do-retransmission=true and a lot of buffers are lost
	  The problem here was that the jitterbuffer lock was unlocked to push
	  the event, but that caused another thread to remove the timer currently
	  being processed, probably because the amount of rtx events
	  (and therefore timers) was getting too high. The solution is to
	  unlock and push the event only after timer processing has finished.
	  fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131

2013-10-24 13:16:42 +0200  Per x Johansson <perxjoh@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Avoid division by zero assert in gst_matroska_demux_search_pos
	  https://bugzilla.gnome.org/show_bug.cgi?id=711829

2013-11-08 17:59:24 +0100  Philippe Normand <philn@igalia.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: generate a non-empty data header
	  Restore the behavior of the element to the state before commit
	  db29522a430e44450415ca3676abd1b77ee923d9. A non-empty header is
	  generated and when the EOS event is received the header is generated
	  again, this time with the correct size.
	  https://bugzilla.gnome.org/show_bug.cgi?id=711699

2013-11-07 16:17:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpsource: update receiver stats for sender
	  An internal sender in a session is also a receiver of its own packets so update
	  the receiver stats. Other senders in the session will use this info to generate
	  correct RB blocks in their SR reports.

2013-11-07 16:13:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: refactor receiver stats update

2013-10-25 18:22:00 -0300  Thiago Santos <ts.santos@partner.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: handle fragmented files with mdat before moofs
	  Assume a file with atoms in the following order: moov, mdat, moof,
	  mdat, moof ...
	  The first moov usually doesn't contain any sample entries atoms (or
	  they are all set to 0 length), because the real samples are signaled
	  at the moofs. In push mode, qtdemux parses the moov and then finds the mdat,
	  but then it has 0 entries and assumes it is EOS.
	  This patch makes it continue parsing in case it is a fragmented file so that
	  it might find the moofs and play the media.
	  https://bugzilla.gnome.org/show_bug.cgi?id=710623

2013-10-25 11:42:37 -0300  Thiago Santos <ts.santos@partner.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: When using a buffered mdat, store all received data for later use
	  In push mode, when qtdemux can't use a seek to skip the mdat buffer it has
	  to buffer it for later use.
	  The issue is that after parsing the next moov/moof, there might be some
	  trailing bytes from the next atom in the file. This data was being discarded
	  along with the already parsed moov/moof and playback would fail to continue
	  after the contents of this moov/moof are played.
	  This is particularly bad on fragmented files that have the mdat before the
	  corresponding moof. So you'd get:
	  mdat|moof|mdat|moof ...
	  When a moof was received, it usually came with some extra bytes that would
	  belong to the next mdat (because upstream doesn't care about atoms alignment).
	  So those bytes were being discarded and playback would fail.
	  This patch makes qtdemux store those extra bytes to reuse them later after the
	  mdat is emptied.
	  https://bugzilla.gnome.org/show_bug.cgi?id=710623

2013-11-07 09:49:55 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Also use the bind-port property if no bind-address was given

2013-11-07 00:51:12 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudiosink: fix segfault when we can't get the channels layout

2013-11-05 17:26:49 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: Make Picture ID mode configurable and default to no picture ID
	  Some implementations (linphone) only support no picture at all in the
	  stream and will fail if one is provided.
	  https://bugzilla.gnome.org/show_bug.cgi?id=711497

2013-11-05 11:18:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* common:
	  Automatic update of common submodule
	  From 865aa20 to dbedaa0

2013-01-29 10:51:07 +0100  Paul HENRYS <visechelle@gmail.com>

	* gst/rtp/gstrtph264pay.c:
	  Add call to gst_rtp_h264_pay_clear_sps_pps() when receiving a STREAM_START event
	  https://bugzilla.gnome.org/show_bug.cgi?id=692787

2013-11-02 22:50:47 +0100  Rico Tzschichholz <ricotz@ubuntu.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtspsrc.h:
	  rtsp: Add missing gio-2.0 deps and includes

2013-11-01 18:31:36 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/audioiirfilter.c:
	  audioiirfilter: Fix initialization coefficient handling
	  Broke unit test.

2013-10-31 14:05:43 -0700  Aleix Conchillo Flaque <aleix@oblong.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: allow setting tls certificate validation flags
	  Added a new property "tls-validation-flags". If the url transport is
	  TLS, the validation flags will be set to the rtsp connection.
	  https://bugzilla.gnome.org/show_bug.cgi?id=711230

2013-10-31 22:43:49 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	  audioiirfilter: Don't crash if no filter coefficients are provided
	  ...and by default use a identity filter.
	  https://bugzilla.gnome.org/show_bug.cgi?id=710215

2013-10-31 19:15:12 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackenc.c:
	  wavpackenc: Fix writing of MD5 sums and other metadata blocks
	  These don't have the FINAL_BLOCK flag set.

2013-10-31 13:02:11 -0200  Djalma Lúcio Soares da Silva <dlucio@impa.br>

	* ext/raw1394/gsthdv1394src.c:
	  hdv1394src: Make it possible to select a camera by its GUID
	  The source hdv1394src has the guid property that permits select a camera
	  connected from its GUID number.
	  However when this property is setted the selected camera is not changed.
	  The source continues using the default camera.
	  This problem was solved using the function iec61883_cmp_connect.
	  The reference for the function could be found here:
	  http://www.dennedy.org/libiec61883/API-iec61883-cmp-connect.html
	  The solution came from dvgrab source code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=710415

2013-10-31 13:20:41 -0300  Thiago Santos <ts.santos@partner.samsung.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: add explicit cast to silence warning
	  Silencing this warning:
	  elements/souphttpsrc.c:533:14: error: comparison between ‘SoupKnownStatusCode’ and ‘enum <anonymous>’ [-Werror=enum-compare]
	  if (status != SOUP_STATUS_OK && !send_error_doc)
	  With gcc 4.8.2 (debian)

2013-10-31 10:38:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.h:
	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: proxy new buffer mode

2013-10-30 16:49:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: add new timestamp mode
	  Add a new timestamp mode that assumes the local and remote clock are
	  synchronized. It takes the first timestamp as a base time and then uses the RTP
	  timestamps for the output PTS.

2013-10-30 22:12:45 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Fix compiler warning
	  matroska-demux.c: In function 'gst_matroska_demux_add_stream':
	  matroska-demux.c:1379:7: error: format '%u' expects argument of type 'unsigned int', but argument 4 has type 'guint64' [-Werror=format=]
	  "%03u", context->uid);
	  ^

2013-10-28 13:21:15 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>

	* gst/videomixer/videoconvert.c:
	  videomixer: remove unneeded guint comparaison
	  https://bugzilla.gnome.org/show_bug.cgi?id=711010

2013-10-28 14:13:12 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	  osxvideosink: fix missing selector name warnings
	  The spaces matter in ObjC
	  https://bugzilla.gnome.org/show_bug.cgi?id=711013

2013-10-28 13:31:34 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>

	* gst/y4m/gsty4mencode.c:
	  y4menc: fix uninitialized variable warning
	  https://bugzilla.gnome.org/show_bug.cgi?id=711011

2013-10-25 11:30:36 -0300  Thiago Santos <ts.santos@partner.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check if the end_time is defined before using it
	  Avoids sending EOS too soon because of overflow. Can happen on
	  fragmented mp4 playback.

2013-10-23 13:38:20 -0300  Thiago Santos <ts.santos@partner.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: use correct unref function
	  Events aren't GstObjects, but GstMiniObjects

2013-10-15 08:16:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: rename chunks_are_chunks to chunks_are_samples and flip the logic
	  As the variable name suggests, sometimes chunks are chunks. Rename the variable
	  to tell what they are when they are not chunks.

2013-10-09 08:04:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix typos and add more logging for unhandled parts

2013-10-14 16:23:25 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Fix memory leak
	  Unmap all GstMemory of the current buffer when flushing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=710110

2013-10-12 20:44:31 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/flv/gstflvmux.c:
	  flvmux: fix broken sample pipeline
	  which was muxing raw audio and video into flvmux, which won't work,
	  even if there were converters.

2013-10-12 20:37:41 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/flv/gstflvmux.c:
	  flvmux: require stream-format=raw for mpeg-2 too, but don't require framed field
	  raw implies that it's framed already. Fixes .. ! faac ! flvmux

2013-10-07 14:27:21 -0300  Thiago Santos <ts.santos@partner.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: do not emit EOS when connection drops
	  If the pipeline is stalled for too long, souphttpsrc will block and
	  stop fetching data from the network. This can cause the connection to
	  drop and souphttpsrc would handle it as an EOS. This patch makes it
	  persist and try to fetch more data until the end of the content length
	  or until receiving an error that it is beyong limits in case the content
	  is unknown.
	  https://bugzilla.gnome.org/show_bug.cgi?id=683536

2013-10-10 13:52:35 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdec.h:
	  dvdec: Don't send segment event before caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=709728

2013-10-09 17:46:33 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Send stream-start, caps and segment events in the right order
	  https://bugzilla.gnome.org/show_bug.cgi?id=709728

2013-10-08 11:28:04 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c:
	  wavenc: A-Law and Mu-Law don't have width/depth/signed caps fields
	  https://bugzilla.gnome.org/show_bug.cgi?id=709614

2013-10-07 12:54:11 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: Fix handling of planar video formats in greedyh method
	  https://bugzilla.gnome.org/show_bug.cgi?id=709507

2013-10-06 10:01:26 -0700  Reynaldo H. Verdejo Pinochet <r.verdejo@partner.samsung.com>

	* gst/matroska/matroska-mux.c:
	  matroska: Trivial grammar fix on debug msg

2013-10-06 09:17:00 -0700  Reynaldo H. Verdejo Pinochet <r.verdejo@partner.samsung.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	* gst/matroska/webm-mux.c:
	  matroskamux: Add context flag for WebM
	  WebM has a couple of specific requirements we need to handle.
	  Idea is to set this flag once and just rely on mux->is_webm
	  at run time instead of repeatedly figuring this out from
	  GST_MATROSKA_DOCTYPE_WEBM (which requires a strcmp()).

2013-10-04 14:42:59 -0700  Reynaldo H. Verdejo Pinochet <r.verdejo@partner.samsung.com>

	* gst/matroska/matroska-mux.c:
	  matroska: Do not write SegmentUID for WebM mux
	  WebM spec states SegmentUID is Unsupported. Files produced
	  with gstreamer without this change will spit an error like
	  this when passed to mkvalidator:
	  ERR201: Invalid 'SegmentUID' for profile 'webm' in Info at 192

2013-10-05 00:00:03 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: make dvd palette change event sticky
	  So they don't get lost.
	  https://bugzilla.gnome.org/show_bug.cgi?id=709454

2013-10-03 16:39:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	  videoflip: Add automatic flip mode driven by image-orientation tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=709312

2013-10-04 13:34:09 +0200  Peter Korsgaard <peter@korsgaard.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: O_CLOEXEC needs _GNU_SOURCE
	  On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only
	  defined when _GNU_SOURCE is specified, so do so.
	  _GNU_SOURCE needs to be defined before any system headers are included,
	  so move the fcntl.h section up.
	  https://bugzilla.gnome.org/show_bug.cgi?id=709423

2013-10-04 12:11:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix race in flush-start/flush-stop
	  When flush-stop arrives before we process the result of the _push() in the
	  loop function, we might pause even though we are not flushing anymore. Fix this
	  race by waiting for the srcpad loop function to completely pause after doing the
	  flush-start.

2013-10-03 22:38:43 +0200  Mathieu Duponchelle <mduponchelle1@gmail.com>

	* gst/videomixer/videoconvert.c:
	  videomixer: Update videoconvert copy
	  https://bugzilla.gnome.org/show_bug.cgi?id=709390

2013-10-03 21:36:34 +0200  Mathieu Duponchelle <mduponchelle1@gmail.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Check if the pad needs reconfiguration in collected
	  https://bugzilla.gnome.org/show_bug.cgi?id=709384

2013-10-03 14:39:35 +0100  Matthieu Bouron <matthieu.bouron@collabora.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Relax sink caps
	  Since jpegdec already parse the jpeg stream, the sink caps could be
	  relaxed. This will allow jpegdec to be selected in more case and in
	  particular when the jpeg typefinder does not find the width and height.
	  https://bugzilla.gnome.org/show_bug.cgi?id=709352

2013-10-03 18:33:01 +0100  Tim-Philipp Müller <tim@centricular.net>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: print probed caps as caps again in debug log
	  This got lost during refactoring.

2013-10-03 11:59:25 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add support for the mp2v fourcc for MPEG-2 video
	  https://bugzilla.gnome.org/show_bug.cgi?id=709270

2013-10-02 15:56:53 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=709266

2013-09-30 12:31:42 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add HEVC support
	  https://bugzilla.gnome.org/show_bug.cgi?id=709093

2013-09-30 12:24:32 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Fix memory leak
	  We were leaking the GList nodes of the pending buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=709079

2013-09-30 12:31:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: fix race when updating the next_seqnum
	  If we were not waiting for the missing seqnum when we insert the lost packet
	  event in the jitterbuffer, we end up not updating the next_seqnum and wait
	  forever for the lost packets to arrive. Instead, keep track of the amount of
	  packets contained by the jitterbuffer item and update the next expected
	  seqnum only after pushing the buffer/event. This makes sure we correctly handle
	  GAPS in the sequence numbers.

2013-09-30 12:30:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: small debug improvement

2013-09-30 11:53:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: reset skew does not reset clock-rate
	  Don't reset the clock-rate when we reset the skew correction algorithm.
	  Reset the skew correction algorithm when we change the clock-rate.

2013-09-30 11:16:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: pause timer when PAUSED
	  Also pause the timer when we go to the PAUSED state. It is possible that we
	  don't have a clock or base-time in PAUSED to perform the timeouts.

2013-09-30 11:15:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: improve debug

2013-09-26 20:41:26 +0200  Hans Månsson <hansm@axis.com>

	* gst/isomp4/gstqtmuxmap.c:
	  mp4mux: Do not require framerate in peer video caps
	  Remove the framerate restriction on the caps.
	  Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708864

2013-09-27 15:05:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: also go into the loop function after connect
	  When we have opened the stream, go into the loop function so that we can
	  receive messages from the server.

2013-09-27 12:53:06 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8
	  https://bugzilla.gnome.org/show_bug.cgi?id=707933

2013-09-26 16:20:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: don't calculate skew without rtptime
	  Skip trying to calculate the skew when we don't have an rtptime.
	  It causes problems when lost packet events are placed in the jitterbuffer.

2013-09-25 23:46:14 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	  configure: get rid of AS_SCRUB_INCLUDE
	  Should not be needed any more.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707658

2013-09-25 17:42:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: disable checks when linking pads
	  We know the pad links will work (and we don't check the return value
	  anyway).

2013-09-25 17:36:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: avoid some pad link checks
	  Link pads without checks, we know it will work.

2013-09-25 12:55:21 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't error out if downstream is not seekable for non-fragmented variants
	  Doing so would be a regression over 1.0 and breaks the unit test.
	  However the result will be most likely unusable, so let's post
	  a warning message on the bus.

2013-09-24 04:02:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: calculate some stats

2013-09-23 17:05:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: move send_lost_event function
	  Move the send_lost_event function to the do_lost_event handling, there is no
	  need to have a separate function.

2013-09-16 11:20:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add code to parse creation time earlier than 1970
	  Use g_date_time seconds manipulation to allow to cover the quicktime
	  spec for creation_time. It uses seconds since 1904.
	  Both paths could be done using the generic approach of seconds since
	  1904 with GDateTime handling, but the first path using seconds from
	  1970 should be more commonly found and avoids a few objects creation and
	  ref/unref, so keep it there for performance.
	  Additionally, the code for handling seconds since 1970 changed from >
	  to >= because having 0 seconds since 1970 is also a valid case for that
	  path to handle.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707975

2013-09-21 00:55:26 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily
	  https://bugzilla.gnome.org/show_bug.cgi?id=708505

2013-09-24 18:30:04 +0100  Tim-Philipp Müller <tim@centricular.net>

	* README:
	* common:
	  Automatic update of common submodule
	  From 6b03ba7 to 865aa20

2013-09-24 15:05:24 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  configure: Actually use 1.3.0.1 as version to make configure happy

2013-09-24 15:00:24 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development

=== release 1.2.0 ===

2013-09-24 14:21:08 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.2.0

2013-09-24 14:20:51 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2013-09-20 19:43:21 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix segfault releasing the sink
	  show_frame is deferred to the main thread and can be called
	  when the sink has been released, so we need to keep an extra ref
	  on ObjectiveC object helper.
	  https://bugzilla.gnome.org/show_bug.cgi?id=708501

2013-09-19 17:11:34 -0400  Robert Krakora <rob.krakora@messagenetsystems.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Restore original GstMemory in buffer if it has been changed
	  https://bugzilla.gnome.org/show_bug.cgi?id=706083

2013-09-23 16:34:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpmanager: update docs

2013-09-23 15:36:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	  docs: update docs with 1.0 element names

2013-09-23 14:13:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: add test for retransmission because of reordering

2013-09-23 14:12:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: always store lost event in jitterbuffer
	  Always prepare a lost event in the jitterbuffer, it is to wake up and make the
	  pushing thread continue. We drop the event when we are not supposed to push lost
	  events downstream.

2013-09-23 11:18:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: schedule lost event differently
	  Schedule the lost event by placing it inside the jitterbuffer with the seqnum
	  that was lost so that the pushing thread can interleave and push it properly.

2013-09-23 11:17:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: remove timeouts from check
	  Timeouts make the test unreliable and are not needed.

2013-09-23 11:15:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: remove list debug

2013-09-23 11:14:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: add type to the item
	  So that the upper layer can know what data is contained in the item.

2013-09-23 09:58:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: fix flush
	  Pass function to flush to properly free the queue items.

2013-09-21 00:08:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: append seqnum -1 packets

2013-09-20 23:48:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpjitterbuffer: use structure to hold packet information
	  Make the jitterbuffer operate on a structure containing all the packet
	  information. This avoids mapping the buffer multiple times just to get the RTP
	  information. It will also make it possible to store other miniobjects such as
	  events later.

2013-09-20 17:48:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: update expected timer when possible
	  When we receive a packet and we have some missing packets, we can update their
	  estimated arrival times based on the timestamp difference.

2013-09-20 17:18:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix order of timeout events
	  Improve the order of the timeout events, if there are timers with the same
	  timeout, we want to trigger the lowest seqnum first. For this we need to loop
	  over the complete array of timers to find the best one before triggering the
	  timeout.

2013-09-20 16:58:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: send lost event before signaling next buffer
	  First send the lost event, then update the next_seqnum counter and then
	  send the signal to the pushing thread that it can retry to push a buffer. This
	  avoids pushing out buffers before the lost event is pushed.

2013-09-20 15:35:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: configure clock-rate on jitterbuffer
	  Add a get and setter to configure the clock-rate in the jitterbuffer instead of
	  passing it as an argument to the insert method.

2013-09-20 12:29:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: add test for packet delay and retransmission

2013-09-20 12:27:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: add option to reset retransmission timers

2013-09-20 12:25:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: stop the timer thread
	  The timeout code could release the lock so we need to check if we are allowed to
	  wait for the clock some more.

2013-09-20 12:25:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: unlock only once

2013-09-20 11:30:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: check both PTS and DTS

2013-09-20 10:55:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: add unit-test for multiple missing packets
	  Check if multiple missing packets generate retransmission events and that the
	  retranmission requests are canceled when the missing packet arrives.

2013-09-20 10:53:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: improve flush and shutdown
	  There is no need to unschedule the timer in flush-start, flush-stop will remove
	  the timers and unschedule.
	  Unschedule the current timer before attempting to join the timer thread.

2013-09-20 10:43:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: improve debug

2013-09-20 10:42:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: set correct expected time
	  When we already have a timer for a packet, skip it but don't forget to adjust
	  the dts to the expected dts of the next packet.

2013-09-20 10:41:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve debug

2013-09-19 16:55:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: use POFFSET instead of OFFSET
	  Use the more correct POFFSET macro to get the offset of a component in its
	  plane. The offset macro gives the offset of the component relative to the start
	  of the frame.

2013-09-21 18:46:29 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/goom/mmx.h:
	  goom: Fix MMX assembly compilation with clang
	  clang does not want or need a clobber list for emms:
	  error: clobbers must be last on the x87 stack
	  Patch taken from the FreeBSD ports, provided by
	  Dan McGregor <dan.mcgregor@usask.ca>

2013-09-20 16:16:57 +0200  Edward Hervey <edward@collabora.com>

	* common:
	  Automatic update of common submodule
	  From b613661 to 6b03ba7

2013-09-20 10:19:22 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Make sure that subtitle buffers are \0-terminated
	  https://bugzilla.gnome.org/show_bug.cgi?id=707933

2013-09-17 12:17:54 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: handle issues correctly when downstream is not seekable
	  The streamable property only make sense for fragmented formats.
	  For regular MP4, when downstream is not seekable we can't rewrite
	  the headers, so qtmux can only work with fast-start=TRUE, where
	  the headers are written finishing the file.
	  For fragmented MP4, when streamable is not seekable and the streamable
	  property is FALSE, we must enforce streamable=TRUE warning the user
	  about this change
	  https://bugzilla.gnome.org/show_bug.cgi?id=707242

2013-09-17 12:06:06 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: make "streamable" TRUE as default
	  The most common use case for fragmented MP4 (Dash and Smooth Streaming)
	  is producing streamable content (even for VOD). streamable=FALSE would only
	  be used to generate fragmented MP4 with and index of MOOF's that could
	  be reproduced without a playlist/manifest
	  https://bugzilla.gnome.org/show_bug.cgi?id=707242

2013-09-17 12:01:30 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: deprecate the streamable property for non-fragmented MP4
	  The streamable property only makes sense for fragmented MP4.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707242

2013-09-19 17:08:19 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: Remove commented out line

2013-09-19 18:43:08 +0100  Tim-Philipp Müller <tim@centricular.net>

	* common:
	  Automatic update of common submodule
	  From 74a6857 to b613661

2013-09-19 17:35:27 +0100  Tim-Philipp Müller <tim@centricular.net>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From 098c0d7 to 74a6857

2013-09-19 16:50:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: don't assume planar formats have just 1 block
	  Don't assume planar formats have just one memory block with the data but use the
	  macros to access the right memory block where a component can be found.

2013-09-19 14:14:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: add retransmission jitterbuffer test
	  Store both DTS and PTS on buffers.
	  Make a queue for srcpad events.
	  Activate pads after linking so that we don't get RECONFIGURE events.
	  Add test for retransmission.

2013-09-19 14:12:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: keep delay as a separate variable in timer
	  Keep a separate delay in the timer so that we still know the original timestamp
	  of the packet that this timer refers to. We can then place the correct
	  running-time in the Retransmission event.

2013-09-19 14:08:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix writability of properties

2013-09-19 11:34:57 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development

=== release 1.1.90 ===

2013-09-19 10:50:23 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.1.90

2013-09-19 10:21:42 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2013-09-19 09:45:18 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/cs.po:
	* po/nl.po:
	* po/pl.po:
	* po/uk.po:
	* po/vi.po:
	  po: Update translations

2013-09-11 14:27:02 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: dmabuf is not a singleton anymore
	  https://bugzilla.gnome.org/show_bug.cgi?id=707793

2013-09-16 13:53:45 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: do not do http requests in READY
	  HEAD requests to discover if the server is seekable shouldn't be done in
	  READY as it might lock the main thread that is doing the state change.
	  https://bugzilla.gnome.org/show_bug.cgi?id=705371

2013-09-18 16:32:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: reevaluate the current timer after timeout
	  When we trigger the timeout logic of a timer, reevaluate it because it is
	  possible that it still has the lowest timeout.

2013-09-18 16:31:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: don't update time when unscheduled
	  Don't try to estimate the current time when we got unscheduled.

2013-09-18 16:29:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: init packet spacing on first buffer
	  Already init the packet spacing variables on the first buffer so that we can
	  calculate the spacing on the second buffer already.

2013-09-18 15:08:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: fix comments

2013-09-18 14:57:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: push the lost event from the timer thread
	  Instead of pushing the lost event from the chain function, schedule a timeout
	  that will push the lost event from the timer thread. This avoid blocking the
	  upstream thread while we push and sync the event.

2013-09-18 14:23:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: add another test
	  The test is modified slightly because the late lost packets are only
	  generated now when a large gap is received.

2013-09-18 14:12:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: round gap duration to multiple of duration
	  Make sure the gap duration in the lost event is a multiple of the packet
	  duration.
	  Enable another test.

2013-09-18 12:29:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: keep track of duration
	  Keep track of the estimated duration of missing packets and use it in the lost
	  event.
	  Enable another unit test

2013-09-18 11:59:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: handle large gaps with one lost event
	  When we have a large number of missing packets, generate one lost event for all
	  the packets that have no chance of being pushed out in time.
	  Fix and activate unit test for large gaps.

2013-09-18 11:56:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refactor lost event sending
	  Also make sure we only increment the expected seqnum and last
	  output timestamp.

2013-09-17 23:21:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor timeout triggers

2013-09-17 23:03:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: simplify the timeout code
	  Keep track of the current time in the timeout loop.
	  Loop over all timers and trigger all the expired ones, we can do this in the
	  same loop that selects the new best timer.

2013-09-17 23:01:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rearrange timer update code
	  Also update the timers when retransmission is disabled. We need to
	  do this because when we added LOST timers when we detected missing packets and
	  we need to remove those timers when the packet finally arrives.

2013-09-17 22:02:04 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/videomixer/Makefile.am:
	  videomixer: link to libm for maths stuff
	  Fixes undefined references to rint and pow on ubuntu
	  build bot.

2013-09-17 15:19:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: release lock on shutdown

2013-09-17 15:11:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/Makefile.am:
	  check: change for videomixer renamed orc file

2013-09-14 16:03:20 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: remove MAX_TOLERATED_LATENESS
	  https://bugzilla.gnome.org/show_bug.cgi?id=707411

2013-09-16 15:54:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/client-H264-rtx.sh:
	  examples: we don't need the queue anymore

2013-09-16 15:53:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: use separate thread for timeouts
	  Use a separate thread for scheduling the timeouts instead of using the
	  downstream streaming thread that might block at any time.

2013-09-14 15:56:04 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: set first_ts to DTS for streams that have DTS
	  https://bugzilla.gnome.org/show_bug.cgi?id=707340

2013-09-14 15:55:22 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: make sure duration is a valid number for last buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=707340

2013-09-14 15:54:29 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: use segment.start or last buffer end time in case of missing DTS
	  https://bugzilla.gnome.org/show_bug.cgi?id=707340

2013-09-03 18:14:04 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  Revert qtmux: Use buffer PTS if DTS is not set"
	  This reverts commit f72c3cf71fde622067f41f31a53978ba4c94469d.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707340

2013-09-16 11:03:06 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixerorc-dist.c:
	* gst/videomixer/videomixerorc-dist.h:
	  videomixer: Update orc generated files
	  https://bugzilla.gnome.org/show_bug.cgi?id=708131

2013-09-13 16:25:49 +0200  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Demux RTCP buffers from the RTP stream
	  If there are RTCP buffers in the RTP stream, process them as
	  RTCP. This way, we want receive streams following RFC 5761
	  https://bugzilla.gnome.org/show_bug.cgi?id=687657

2013-09-13 23:26:21 +1000  Jan Schmidt <thaytan@noraisin.net>

	* gst/rtp/gstrtpL24depay.c:
	  rtp: Remove bogus extra caps from L24 template.
	  The extra caps entry in the template was making it sometimes
	  get plugged for any dynamically allocated payload type.

2013-09-13 12:40:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpbin: use PacketInfo for the sender
	  Avoid mapping the packet multiple times when sending RTP.

2013-09-13 12:22:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpbin: store more in the PacketInfo
	  Store all info in the PacketInfo so that we can avoid mapping the packet
	  multiple times.

2013-09-13 11:32:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.h:
	  session: store more in the PacketInfo structure

2013-09-13 11:08:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpbin: RTPArrivalStats -> RTPPacketInfo
	  Rename a structure because we are also going to use this for the sender
	  bits.

2013-09-13 10:55:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  source: small cleanups

2013-09-12 13:31:01 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only update stop position if seek requests it
	  Check for GST_SEEK_TYPE_NONE for stop poistion and only update
	  the stop time if it is requested. Otherwise just maintain whatever
	  was stored at the segment
	  https://bugzilla.gnome.org/show_bug.cgi?id=707530

2013-09-13 08:53:25 +0200  Rico Tzschichholz <ricotz@ubuntu.com>

	* gst/rtp/Makefile.am:
	  rtp: Add missing headers tp fix make dist
	  In addition to a956a6ceb2deb87cc1361aee1d6626449f46dab2

2013-09-12 15:07:48 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Make sure we have enough data to read image tags
	  Thanks to iputinei for reporting this on IRC.

2013-09-12 15:01:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: handle segments with non-0 start
	  We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to
	  transform it back to a buffer timestamp before pushing out the buffer.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931

2013-09-11 13:11:58 -0600  Seán de Búrca <leftmostcat@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix off-by-one in validation of UTF-8
	  https://bugzilla.gnome.org/show_bug.cgi?id=707933

2013-09-11 14:32:17 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Do not check if caps are empty when they are NULL
	  In the case the caps are actually NULL, we should just concider it the
	  same way as empty caps in that case.

2013-09-10 16:44:53 -0600  Seán de Búrca <leftmostcat@gmail.com>

	* gst/videomixer/videomixerorc-dist.c:
	* gst/videomixer/videomixerorc-dist.h:
	  videomixer: fix build if orc is not installed
	  https://bugzilla.gnome.org/show_bug.cgi?id=707886

2013-09-10 17:57:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Preserve seqnum when pushing seek upstream
	  After converting a seek from time to bytes, use the same seqnum
	  on the event that goes upstream

2013-09-05 00:17:16 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: track streams that are EOS on push mode to finish earlier
	  When the segment has a defined stop position, qtdemux should check
	  when streams reach this position and mark those as EOS. When all
	  streams are EOS it will return GST_FLOW_EOS to upstream to allow
	  the pipeline to finish instead of continuously consume buffers
	  from upstream that are not useful for the segment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707530

2013-09-04 15:34:35 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: preserve stop of segment when doing seeks in push mode
	  When handling seeks in push mode, qtdemux converts the seek to bytes
	  and pushes upstream. It needs to keep track of the seek and the
	  subsequent segment to be able to map them back to the requested
	  seek time and properly preserve the segment stop of the seek.
	  This is done by using the start offset in bytes of the seek,
	  that should be the same of the segment from upstream. And this
	  is also backwards compatible with what qtdemux already was using.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707530

2013-07-26 19:40:53 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2pad.h:
	  videomixer: Add colorspace conversion
	  https://bugzilla.gnome.org/show_bug.cgi?id=704950

2013-08-06 15:38:39 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't send reconfigure event when formats or PAR are different
	  It is racy with multiple pads.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704950

2013-07-25 13:49:57 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend.c:
	* gst/videomixer/blendorc.orc:
	* gst/videomixer/gstcms.c:
	* gst/videomixer/gstcms.h:
	* gst/videomixer/videoconvert.c:
	* gst/videomixer/videoconvert.h:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixerorc.orc:
	  videomixer: Bundle private copies of videoconvert code
	  Ideally, this would be part of libgstvideo.
	  Prefixes videoconvert symbols with videomixer_.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704950

2013-08-22 00:03:48 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Use newly #defined metadata names.

2013-09-09 15:11:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: only wait if we flushed
	  Only wait for the STREAM_LOCK when we flushed something when sending
	  a command for PAUSED or PLAYING.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611

2013-09-09 15:09:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: return when a flush was issued
	  Make gst_rtspsrc_loop_send_cmd() return TRUE when the current
	  action has been flushed

2013-09-09 11:16:40 +0200  David Holroyd <dave@badgers-in-foil.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpL24depay.h:
	* gst/rtp/gstrtpL24pay.c:
	* gst/rtp/gstrtpL24pay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtp: add L24 pay and depayloader
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734

2013-09-09 14:46:42 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Fix missing condition in previous commit

2013-09-09 14:44:58 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Also fix strides for other semi-planar video formats

2013-09-09 14:41:42 +0200  Andreea Fulger <andreea.fulger@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Fix stride for NV12/NV21
	  https://bugzilla.gnome.org/show_bug.cgi?id=707758

2013-09-07 16:37:03 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: fix leaking buffer and caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=707688

2013-09-05 19:46:37 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix build on win32
	  gstudpsrc.c:855:15: error: #if with no expression

2013-09-04 15:50:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: handle unseekable streams
	  Handle streams that we can't seek in and ignore them in the
	  seek logic.

2013-09-04 15:25:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: only check video compression for video streams
	  Or else we might deref a stream with a NULL strf.vids and segfault

2013-06-18 13:27:20 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/atoms.c:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/ftypcc.h:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add support for the avc3 sample entry format of the AVC file format
	  Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
	  structure for fragmented MP4 called "avc3". The principal difference
	  between AVC1 and AVC3 is the location of the codec initialisation
	  data (e.g. SPS, PPS). In AVC1 this data is placed in the initial
	  MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data
	  goes in the first sample of every fragment (i.e. the first sample in
	  each mdat box).  The principal reason for avc3 is to make it easier
	  for client implementations, because it removes the requirement to
	  insert the SPS+PPS in to the decoder pipeline every time there is a
	  representation change.
	  This commit adds support for the "avc3" atom, which is almost identical
	  to the "avc1" atom, except it does not contain any SPS or PPS data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702004

2013-09-04 00:27:50 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't set EOS to FALSE when the collectpad *is* EOS
	  https://bugzilla.gnome.org/show_bug.cgi?id=707238

2013-09-03 17:32:41 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: cleanup on error after state change
	  https://bugzilla.gnome.org/show_bug.cgi?id=707229

2013-09-03 11:23:24 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: Bind to multicast addresses on non-Windows systems
	  On Windows it's not possible to bind to a multicast address
	  but the OS will make sure to filter out all packets that
	  arrive not for the multicast address the socket joined.
	  On Linux and others it is necessary to bind to a multicast
	  address to let the OS filter out all packets that are received
	  on the same port but for different addresses than the multicast
	  address
	  And deprecate the multicast-group property and replace it with the
	  address property.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707042

2013-09-03 10:10:01 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Free GstBaseParseFrame if pushing a header failed

2013-09-02 16:02:37 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Refactor address resolval into its own function

2013-09-02 23:00:29 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/replaygain/gstrganalysis.c:
	  replaygain: fix taglist leak in rganalysis
	  And add some FIXMEs.

2013-09-02 22:50:58 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/rganalysis.c:
	  tests: rganalysis: rename function for clarity

2013-03-18 14:32:07 +0100  Christoph Reiter <reiter.christoph@gmail.com>

	* tests/check/elements/rganalysis.c:
	  tests: fix skipped rganalysis tests
	  In 0.10 elements would post tag messages on the bus
	  directly, and rganalysis would only post a tag message
	  when it changed tags. In 1.0, only sinks post tag
	  messages when they receive the serialised tag event.
	  This means that we get an additional tag message on
	  the bus now where we didn't expect one before.
	  https://bugzilla.gnome.org/show_bug.cgi?id=695090

2013-09-02 11:46:52 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Properly propagate downstream flow returns upstream
	  https://bugzilla.gnome.org/show_bug.cgi?id=707229

2013-09-01 21:18:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/shout2/gstshout2.c:
	* gst/avi/gstavi.c:
	* gst/isomp4/isomp4-plugin.c:
	* gst/rtsp/gstrtsp.c:
	* sys/sunaudio/gstsunaudio.c:
	* sys/v4l2/gstv4l2.c:
	  Don't use setlocale in plugins()
	  Only apps should call setlocale(), not libraries.

2013-08-29 13:15:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmpvpay.c:
	  rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay
	  RTP buffer allocation should not be done with padding for the specific MPEG2
	  header as the padding is done at the end of the buffer and the last byte is
	  the size of the padding.
	  https://bugzilla.gnome.org/show_bug.cgi?id=706970

2013-08-28 10:51:32 +0200  Bernhard Miller <bernhard.miller@streamunlimited.com>

	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosink.h:
	  autovideosink: add sync property
	  https://bugzilla.gnome.org/show_bug.cgi?id=706955

2013-08-28 07:15:00 +0200  Bernhard Miller <bernhard.miller@streamunlimited.com>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosink.h:
	  autoaudiosink: introduce sync property
	  https://bugzilla.gnome.org/show_bug.cgi?id=706955

2013-08-27 17:33:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: push buffers after segment stop until reaching a keyframe
	  This should make decoders able to precisely push buffers until the stop
	  time in case they need the next keyframe to do it.
	  Also, according to gst_segment_clip, it should only push a buffer that
	  the starting ts is strictly smaller than the segment stop, so we change
	  the min < comparison for <=

2013-08-28 13:26:47 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development

=== release 1.1.4 ===

2013-08-28 12:52:25 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* gst/audiofx/audiopanoramaorc-dist.c:
	* win32/common/config.h:
	  Release 1.1.4

2013-08-28 12:52:16 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2013-08-28 12:32:10 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/pt_BR.po:
	  po: update translations

2013-08-27 15:25:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: remove framerate restriction
	  Remove the framerate restriction on the caps.

2013-08-27 09:38:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: only update next check time when reconsidering
	  Don't update the next RTCP check time in all cases but only when we
	  reconsidered. This avoids delaying sending a full RTCP packet when we
	  are doing early feedback.

2013-08-27 09:37:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add more debug

2013-08-27 09:34:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	  jitterbuffer: fix types of the retransmission event

2013-08-27 09:33:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: only timeout EXPECTED timers on gap
	  Only timeout the EXPECTED timers when we detect a large seqnum gap.

2013-08-26 13:47:53 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  configure.ac: Don't set BZ2_LIBS if bz2 is not found

2013-08-26 11:50:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtsession: fix locking
	  We need to take the session lock when getting and manipulating the
	  source.

2013-08-26 11:50:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: add some more debug

2013-08-20 22:12:03 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: don't send flush_stop twice.
	  If we get flush start and a seek we need to only send flush_stop once.
	  More info at #706441

2013-08-23 15:56:43 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: propagate discont

2013-08-23 15:49:47 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: remove dynamic sourcpads when going from PAUSED to READY

2013-08-23 15:29:28 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last
	  https://bugzilla.gnome.org/show_bug.cgi?id=637754

2013-08-23 15:47:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtxqueue: add property to configure queue size

2013-08-23 12:07:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/client-H264-rtx.sh:
	* tests/examples/rtp/server-VTS-H264-rtx.sh:
	  tests: add retransmission example

2013-08-23 11:55:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: proxy jitterbuffer do-retransmission property

2013-08-23 11:17:45 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/avi/gstavimux.c:
	  avimux: unmap the correct buffer
	  The audio buffer was mapped so unmap it and not the video buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=706642

2013-08-18 23:32:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Add property to find out the device currently in use
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:31:15 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: De-duplicate code to get the current sink input info
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 22:27:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Implement changing the device while playing
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:32:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: Add property to find out the device currently in use
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:31:15 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: De-duplicate code to get the current source output info
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 22:27:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Implement changing the device while playing
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-22 14:55:14 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  configure: Fix bz2 configure check for Windows
	  Due to function decorations on Windows AC_CHECK_LIB can't be used to check for bz2.
	  https://bugzilla.gnome.org/show_bug.cgi?id=465924

2013-02-22 20:57:00 +0900  Akihiro Tsukada <atsukada@users.sourceforge.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulsesink: Add support for AAC pass-through
	  https://bugzilla.gnome.org/show_bug.cgi?id=694445

2013-06-24 17:29:37 +0200  Kishore Arepalli <kishore.arepalli@gmail.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: crashes if any property changes during playback when location property is not set
	  https://bugzilla.gnome.org/show_bug.cgi?id=702988

2013-08-21 14:54:26 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.h:
	  pulse: Share static caps definition between src and sink
	  The src was also missing 24-bit sample formats

2013-08-21 16:53:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtx: various improvements
	  Use locking
	  Don't push from the event handler, collected packets in a queue and push from
	  the chain function.
	  Clear queues on shutdown.

2013-08-21 16:50:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  session: generate events correctly
	  Do correct shifting of the bitmask for lost packets.

2013-08-21 16:47:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpmanager.c:
	  rtp: register rtx element better

2013-08-21 16:32:50 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: WAVEFORMATEX is unsigned for 8 bit integers, and signed for others
	  Probably fixes
	  https://bugzilla.gnome.org/show_bug.cgi?id=705477

2013-08-21 13:03:34 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: don't ignore return value from _finish_frame()
	  gst_video_encoder_finish_frame() will return FLOW_OK here if
	  there's no output buffer.

2013-08-21 12:56:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	  jpegdepay: add some more debug

2013-08-21 12:10:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstdepay.h:
	  rtpgstdepay: only push events when they changed
	  Keep track of the STREAM_START and TAG events and only push them
	  when they changed.

2013-08-21 10:52:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: taglists should not be merged in 1.0

2013-08-21 10:28:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: flush on FLUSH_STOP event

2013-08-21 10:03:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: reset on state change
	  Do full reset on state change to READY

2013-08-21 09:55:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: reset on FLUSH_STOP
	  Clear the adapter and pending buffer list on FLUSH_STOP.

2013-08-21 09:39:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: don't use clock for config interval
	  We can't use the clock to time our config-interval because we are not
	  live (or there might not be a clock or the clock might not be running).
	  Instead just simply take the timestamp diff.

2013-08-21 09:33:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.h:
	  rtpgstay: don't use // comments

2013-08-08 11:55:22 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix response argument in handle-request signal

2013-08-08 11:54:41 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add sdes property and proxy it to rtpbin

2013-08-07 09:47:35 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  Send a stream-start whenever we send tags This is to make sure tags are cleared on the client if the stream-start was previously lost, otherwise, the client may end up with a merged taglist of multiple songs

2013-07-25 21:12:05 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval This is useful in case the packet containing the inlined caps was lost or if new client joins an already running RTP stream and they missed the previous tag events. This also makes the payloader keep a list of merged tags so the retransmitted tag event contains all previously received. A STREAM_START event will flush the list of tags.

2013-07-25 21:10:10 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time

2013-07-25 21:03:34 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps

2013-07-25 20:54:50 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList This is necessary to fix event/caps sending. If we send a STREAM_START packet, it will cause an error because the stream didn't receive its caps and new-segment events, so we must wait for the first buffer before sending the stream-start event buffer. However, the caps will be sent at the same time and so the 'inline caps' will be set for the event. We need to be able to payload individual packets (data, caps or events) and only send them when we call flush.

2013-07-25 17:56:38 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START

2013-07-25 17:52:16 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3

2013-08-20 14:36:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: handle EOS
	  When the queue is empty, and we received EOS, pause and push an EOS
	  event downstream.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387

2013-08-20 10:26:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update docs

2013-08-20 10:25:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update all timers
	  Keep looping over all registered timers so that we can mark them lost instead of
	  stopping as soon as we find the timer for the current seqnum.

2013-08-20 08:55:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: remove unused variables

2013-08-19 21:10:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reorganize timer handling
	  Restructure handling of incomming packet and the gap with the expected seqnum
	  and register all timers from the _chain function.
	  Convert a timer to a LOST packet timer when the max amount of retransmission
	  requests has been reached.

2013-08-19 21:37:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor packet spacing calculation

2013-08-19 21:34:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: keep track of last seqnum and dts

2013-08-19 21:29:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanups

2013-08-19 21:21:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reset retransmission timers in add/reschedule
	  Reset the retransmission timers when adding and rescheduling a timer.

2013-08-19 21:12:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rename variables for packet spacing

2013-08-19 14:58:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: remove lost timer when we get the packet
	  When we receive a packet, also remove the LOST timer for it.

2013-08-19 14:56:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: expected seqnum must increase
	  Only update the expected seqnum when it is bigger than the previous expected
	  seqnum.

2013-08-19 14:55:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add more debug

2013-08-12 16:15:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtxqueue: add retransmission queue element

2013-08-12 14:53:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add some docs

2013-08-06 16:29:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: handle NACK feedback and generate events
	  Handle and parse the feedback NACK packets and generate a Retransmission
	  event for each NACKed packet

2013-08-19 13:19:42 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add forward declaration for gst_v4l2_object_get_format_list

2012-10-22 17:58:07 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2: De-duplicate caps probing between src and sink

2013-08-13 17:32:17 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/Makefile.am:
	* ext/pulse/pulseprobe.c:
	* ext/pulse/pulseprobe.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulse: Remove unused GstPulseProbe

2013-08-19 12:46:45 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tunerchannel.c:
	* sys/v4l2/tunernorm.c:
	  v4l2: Use G_DEFINE_ macros for added thread safety

2013-08-17 11:28:13 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: Do not send flush_stop ourself after a flush_start
	  When we receive a flush_start, we should wait for the next flush_stop
	  and foward it, not create a flush_stop ourself.

2013-08-16 17:10:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: init debug category early
	  Init the debug variable when we register the element because it is also used by
	  the payloader element when it calls the add_sps_pps method.

2013-08-16 13:26:28 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/flac/gstflacenc.c:
	  flacenc: Properly set headers via the base class instead of just pushing them downstream
	  Prevents buffers from being send before the caps and segment events.

2013-08-15 10:59:10 +0100  Chris Bass <floobleflam@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check denominator isn't zero before scaling duration.
	  When gst_qtdemux_configure_stream sets fps_d, check that n_samples is
	  non-zero before using it as a denominator to scale the stream duration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=706076

2013-08-15 15:08:05 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  ext: Use new flush vfunc of video codec base classes and remove reset implementations

2013-08-14 16:19:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: forward flush before stopping dataflow
	  First forward the flush event and then stop our loop function.

2013-08-14 13:10:32 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	  configure: require libsoup >= 2.38
	  Bump libsoup requirement for newer API used, like headers_get_one().
	  2.38 is from early 2012 and is in linen with our GLib requirement.

2013-08-14 11:54:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/soup/gstsouphttpsrc.c:
	  soup: don't use deprecated soup_message_headers_get() API

2013-08-13 17:44:50 +0200  Edward Hervey <edward@collabora.com>

	* .gitignore:
	  .gitignore: Ignore files from automake test-driver

2013-08-12 15:28:34 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Use the SPS/PPS handling function from the depayloader
	  Remove duplicated copies
	  https://bugzilla.gnome.org/show_bug.cgi?id=705553

2013-08-12 15:26:08 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: Make the SPS/PPS deduplication function generic
	  Make it not touch any internals of the depayloader
	  https://bugzilla.gnome.org/show_bug.cgi?id=705553

2013-08-13 14:09:20 +0100  Chris Bass <floobleflam@gmail.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: allow conversion from raw AAC to ADTS
	  This patch will prepend ADTS headers to raw AAC audio frames, allowing
	  upstream elements to link to decoders that only support AAC in ADTS format.
	  Note that no error correction bits are added to ADTS frames in this code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=615740

2013-08-13 12:44:11 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Only free GCheckSum after its last usage
	  https://bugzilla.gnome.org/show_bug.cgi?id=705760

2013-08-13 12:02:29 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix critical setting a NULL uri redirection

2013-07-13 01:50:56 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: add redirection to the URI query

2013-07-31 10:42:07 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: elst should offset samples instead of buffers
	  The current approach where buffers are offset is not ideal, as during seek
	  and loop current time is compared to sample times.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700264

2013-08-07 19:32:07 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	* tests/check/elements/videomixer.c:
	  videomixer: Send EOS if buf_end >= segment.stop
	  That means the whole segment is already played, and we are sure we
	  are EOS at that point.
	  Also handle segment seeks, and do not send EOS in that case.

2013-08-04 14:40:38 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: send proper stream_start event
	  https://bugzilla.gnome.org//show_bug.cgi?id=705449

2013-08-08 11:51:17 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/ebml-read.c:
	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't print warnings during flushing and stop as soon as possible
	  https://bugzilla.gnome.org//show_bug.cgi?id=705442

2013-08-07 11:14:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpvp8depay: mark key frames and delta frames properly
	  https://bugzilla.gnome.org/show_bug.cgi?id=705550

2013-08-05 23:23:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add NACK feedback in RTCP

2013-08-05 23:22:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  source: add methods to register NACK
	  Add a method to register a missing packet for an ssrc along with
	  methods to get the missing packets and clear them.

2013-08-04 23:05:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: handle Retransmission event and schedule NACK
	  Handle the retransmission event from downstream and use it to schedule a NACK
	  request.

2013-08-05 23:20:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: pass data to remove func
	  Pass the data to the remove function because we are going to deref it when there
	  is pli or fir.

2013-08-06 15:28:50 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix compilation

2013-08-06 15:17:44 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE

2013-08-06 11:58:38 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Make sure to send EOS if the buffer end time equals the segment end time
	  Otherwize EOS never gets sent in that particular case.

2013-08-05 08:49:50 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: Ensure src caps are writable
	  In some cases the src caps determined by goom weren't writable, causing
	  a bunch of assertion failures and failed caps. Fixed by always
	  explicitely making the caps writable
	  https://bugzilla.gnome.org/show_bug.cgi?id=705475

2013-08-04 23:18:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: use common send_rtcp method
	  Reuse the send_rtcp method that already asks for the current time when
	  requesting a keyframe.

2013-08-04 23:12:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: Don't use ClockTimeDiff for unsigned delays

2013-08-04 16:52:15 +0200  Edward Hervey <edward@collabora.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use buffer PTS if DTS is not set
	  Avoids ending up with completely bogus scaled duration/pts when new
	  buffers have invalid DTS.

2013-08-04 14:32:47 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/souphttpsrc.c:
	  tests: skip https test if there's no TLS support in soup/glib

2013-08-04 11:20:41 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtsp/gstrtpdec.c:
	  rtpdec: use generic marshaller

2013-08-04 10:52:33 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* sys/v4l2/.gitignore:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2-marshal.list:
	* sys/v4l2/tuner-marshal.list:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tuner.h:
	* win32/MANIFEST:
	* win32/common/tuner-enumtypes.c:
	* win32/common/tuner-enumtypes.h:
	* win32/common/tuner-marshal.c:
	* win32/common/tuner-marshal.h:
	  v4l2: remove unused enumtypes and use generic marshaller

2013-08-04 10:47:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* gst/udp/.gitignore:
	* win32/common/gstudp-enumtypes.c:
	* win32/common/gstudp-enumtypes.h:
	* win32/common/gstudp-marshal.c:
	* win32/common/gstudp-marshal.h:
	  udp: remove unused marshal and enumtypes files

2013-08-04 09:38:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* gst/rtpmanager/.gitignore:
	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpsession.c:
	* win32/MANIFEST:
	* win32/common/gstrtpbin-marshal.c:
	* win32/common/gstrtpbin-marshal.h:
	  rtpmanager: use generic marshaller

2013-08-04 00:13:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: send event in right direction

2013-08-02 17:38:34 -0700  David Schleef <ds@schleef.org>

	* configure.ac:
	* tests/check/Makefile.am:
	  tests: create/remove orc directory at proper time
	  Before automake creates .deps directories, and during distclean.

2013-08-03 00:25:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add FIR and PLI like other RTCP packets
	  Add the FIR and PLI packets like the other RTCP packet instead of from the
	  on-sending-rtcp default signal handler.

2013-08-02 17:22:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix property ranges

2013-08-02 16:42:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: push retransmission events

2013-08-02 14:12:16 +0200  Lubosz Sarnecki <lubosz@gmail.com>

	* configure.ac:
	  build: add subdir-objects to AM_INIT_AUTOMAKE
	  Fixes warnings with automake 1.14
	  https://bugzilla.gnome.org/show_bug.cgi?id=705350

2013-08-02 14:54:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add support for retransmission retry
	  When we didn't receive a packet after requesting retransmission, retry
	  asking for retransmission for a certain period.

2013-08-02 14:19:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add properties
	  Add properties to control retransmission parameters

2013-08-02 12:44:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: use corrected timeout when rescheduling
	  When we recalculate the timeout, use the corrected timeout value depending on
	  the timer type.

2013-08-02 12:43:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update timers after queueing
	  Else we might update the timer needlessly for duplicates.

2013-08-02 12:42:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: move method up

2013-08-02 06:28:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanup

2013-08-01 23:26:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: unschedule old expected packets
	  When we receive a new packet, unschedule old outstanding packets when their
	  seqnum is too far away.

2013-08-01 23:29:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor timer update

2013-08-01 23:24:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update timers when removing
	  Update the timers when we remove a timer.
	  Handle canceled timers, make them unschedule the current timer and
	  trigger the timeout code.

2013-08-01 23:22:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix typo

2013-08-01 15:40:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve timeout management
	  If we change the seqnum of an existing timer and we were waiting for
	  that timer, unschedule it. If we change the timeout of an existing timer and we
	  were waiting on it, only unschedule when the new time is smaller.

2013-08-01 15:05:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: install timer for expected arrival
	  Install a timer that is triggered when the expected arrival time of a packet
	  expired.

2013-08-01 14:56:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve unschedule of timers
	  Conflicts:
	  gst/rtpmanager/gstrtpjitterbuffer.c

2013-08-01 12:21:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: move code around

2013-08-01 12:07:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: estimate inter packet spacing
	  When we see two packets with consecutive seqnums and a different RTP time, use
	  the DTS difference as the inter packet spacing estimate.

2013-08-01 12:01:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: keep track of current timeout

2013-08-01 11:49:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: cleanup timer handling

2013-08-01 11:40:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reset is only possible with a GAP

2013-08-01 11:29:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: operate on DTS
	  Make the jitterbuffer schedule the timeouts based on the DTS instead
	  of the PTS. This makes it all smoother with reordered frames and gives
	  the decoder time to reorder the frames in time.

2013-08-01 11:14:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rename timout variable

2013-07-31 17:08:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanup

2013-07-31 16:59:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: block output in paused or buffering

2013-07-31 16:59:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: store pts in timer
	  Only store the pts in the timer so that we can both do timeouts with timings on
	  the input and output of the jitterbuffer.

2013-07-30 23:14:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refactor jitterbuffer
	  Refactor the jitterbuffer code. Make separate function for peeking a buffer,
	  pushing the next buffer, waiting for timeouts and handling the timeouts.
	  The main loop now tries to push as many buffers as it can until it runs out of
	  buffers or when it detects a seqnum discont. Then it will wait for some event to
	  happen before attempting to push more buffers.
	  Make methods to register timeouts in an array. These timeouts are registered
	  when we detect a missing packet, sync for the first packet or when we find an
	  estimation for the end-of-stream.
	  This greatly simplifies and clarifies the code and also makes it possible to
	  register more complicated timeout schemes later.

2013-07-30 18:52:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: use NULL to ignore percent
	  If we pass NULL to pop and push we ignore the percent result.

2013-07-30 07:00:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor
	  Move eos estimation into separate function

2013-07-30 14:28:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: don't leak stream_id string
	  https://bugzilla.gnome.org/show_bug.cgi?id=705142

2013-07-29 19:53:52 +0100  Tim-Philipp Müller <tim@centricular.net>

	* po/LINGUAS:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/ja.po:
	* po/nb.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/sl.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: update translations

2013-07-29 19:48:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/.gitignore:
	  tests: ignore new test binaries

2013-07-29 14:47:49 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development

=== release 1.1.3 ===

2013-07-29 13:42:18 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.1.3

2013-07-29 13:42:05 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2013-07-29 12:12:41 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	  gst: Don't swap start/stop for negative rates in the SEGMENT query

2013-07-29 11:18:40 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check for data size when parsing h264 codec data from strf atom

2013-07-29 10:53:54 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Implement SEGMENT query

2013-07-29 10:53:47 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Implement SEGMENT query

2013-07-29 10:50:59 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	  avidemux: Implement SEGMENT query

2013-07-27 18:10:22 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: Support H264 fourcc
	  https://bugzilla.gnome.org/show_bug.cgi?id=704996

2013-07-28 18:09:33 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/flac/gstflacenc.c:
	  flacenc: Fix handling of image tags
	  The caps should be used to get the mimetype and there is
	  only an info structure for the GstSample if the image-type
	  is not NONE.

2013-07-28 18:04:32 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/flac/gstflacenc.c:
	  flacenc: Don't crash if there is no image tag information
	  https://bugzilla.gnome.org/show_bug.cgi?id=705018

2013-07-28 17:38:56 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix duration reporting in push mode
	  https://bugzilla.gnome.org/show_bug.cgi?id=700933

2013-07-28 17:32:27 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	  avidemux: Don't forget unmapping and unreffing buffer

2013-07-26 21:06:17 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: unmap buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=704951

2013-07-26 22:31:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: don't make buffer writable prematurely
	  There is no reason to make the SR buffer writable at this point. This is better
	  delayed until needed.

2013-07-26 22:25:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: ignore RTCP for inactive sources

2013-07-26 22:25:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: small cleanup

2013-07-26 17:17:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.h:
	  session: handle partial RTCP report blocks
	  When we have more SSRCs to report than what fit in an RTCP packet, use a
	  generation counter to make sure all of them end up in a packet eventually.

2013-07-26 17:23:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: create SSRC before doing session cleanup
	  Make the internal source before we do session cleanup

2013-07-26 17:21:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: reorganize the report block code

2013-07-26 16:02:01 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix memory leak in check_subtitle_buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=704921

2013-07-26 14:21:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: refactor active and sender checks

2013-07-26 12:06:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: remove internal sources on timeout
	  When an internal source times out and becomes a receiver, remove it.

2013-07-26 11:47:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: create an internal source for RTCP
	  When we need to do RTCP and we don't have an internal source yet,
	  make one.

2013-07-26 10:47:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	  session: remove old code to change SSRC
	  Remove code used to change the SSRC after a collision. We now send
	  a RECONFIGURE event upstream to make the upstream element change the SSRC.

2013-07-26 10:42:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  source: don't update packet SSRC
	  Remove the code to update the SSRC in packets, it can never be called now that
	  we always use a source with matching packet SSRC.

2013-07-26 10:24:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: delay allocation of internal source
	  Allocate the internal source when we receive a caps with the SSRC or when we see
	  a buffer with the SSRC.

2013-07-26 10:00:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  session: generate reconfigure on collision
	  When we detect a collision, change the SSRC that we suggest upstream
	  and trigger RECONFIGURE. This should make upstream select a new SSRC.

2013-07-26 09:37:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: produce RTCP for all internal sources
	  Loop over all the internal sources and produce RTCP. We also need
	  to queue the RTCP packets and send them when we are finished.

2013-07-26 01:40:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: deprecate internal source and ssrc properties
	  Deprecate the internal source and internal ssrc properties. There might
	  be more than one internal source.

2013-07-26 01:29:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: internal sources don't use probation

2013-07-26 01:24:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  session: give caps to session
	  Let the session parse the caps and update its SSRC when needed.

2013-07-26 01:14:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: make method to suggest available SSRC
	  Make a method to suggest the best available SSRC. This is the SSRC of the last
	  created internal source and is used to instruct upstream to produce this
	  SSRC.

2013-07-26 01:01:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: keep SDES and set on new internal sources
	  Keep track of the SDES ourselves and set it on all newly created
	  internal sources.

2013-07-26 00:48:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: make method to make internal sources
	  Add a method to obtain an internal source and use it to create
	  our internal source

2013-07-26 00:29:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.h:
	  session: count internal sources and how many are senders

2013-07-26 00:14:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: separate BYE marking and scheduling
	  First mark sources with BYE and then schedule the BYE RTCP message.

2013-07-25 23:56:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: get SSRC from RTCP packet itself
	  Get the SSRC from the RTCP packet instead.

2013-07-25 23:51:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: fix bandwidth calculation
	  We iterate over all sources and the internal one is also in the
	  hashtable so avoid adding it twice.

2013-07-25 23:38:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add some docs

2013-07-25 23:11:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: Rearrange RTCP reporting a little
	  Make a function to generate an RTCP packet for a source, pass the source as a
	  parameter.
	  Move timeout of collisions to session cleanup phase.

2013-07-25 22:39:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: move check for is_early around
	  Move the check for the early RTCP to where it is needed and used.

2013-07-25 17:35:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: parse packet outside of the session lock

2013-07-25 17:34:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: do nicer checks for internal sources

2013-07-25 17:15:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  session: let source keep track if it sent BYE

2013-07-25 17:06:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  source: reset more

2013-07-25 16:49:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  source: also use the source for bye_reason
	  Store the BYE reason in our internal source object. Rename the methods on the
	  source object a little because now the BYE can be received in RTCP or
	  set when the session wants to send BYE.

2013-07-25 16:24:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  session: configure sdes with structure only
	  Remove code to configure the SDES with methods and types, only
	  allow configuration with GstStructure

2013-07-25 15:56:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: refactor add and find source
	  Make functions to find and add a source to the hashtable.

2013-07-25 15:43:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: remove source from sync_rtcp
	  We don't need to know the sender source of the session in the
	  callback, the SR packet is for all participants in the session.

2013-07-24 14:18:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add some more debug

2013-07-15 17:11:45 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	  aacparse: allow conversion from ADTS to raw AAC
	  Some muxers (eg, qtmux) only support raw AAC, so this allows linking
	  an encoder that outputs ADTS only to those muxers.
	  The conversion is simple (omit the first 7 or 9 bytes of the frame),
	  but has to be done in pre_push instead of handle_frame as 1.0 does
	  not seem to allow skipping bytes there as 0.10 used to.
	  Other conversions are not supported (yet).

2013-07-15 17:15:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix object_type parsing off-by-one in ADTS frame
	  According to http://wiki.multimedia.cx/index.php?title=ADTS,
	  the value stored in ADTS headers is one less than the object
	  type of the AAC stream.
	  A look at ffmpeg shows it also adds 1 to the value read off
	  the ADTS header.
	  Note that this might break other things that happen to have
	  an inverse off by one to match the existing code.

2013-07-25 11:13:01 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix seqnum handling for seeks
	  Use the same seqnum as the seek for flushes/segments that are
	  caused by the seek. Also do the same for segment events
	  Fixes #676242

2013-07-25 01:39:58 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: fix seqnum handling for seeks
	  Use the same seqnum as the seek for flushes/segments that are
	  caused by the seek. Also do the same for segment events
	  Fixes #676242

2013-07-25 01:11:31 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: correctly handle seqnum for seeks and segments
	  Use the same seqnum on messages and events for derived events.
	  Fixed for flushes / stream-start / segment after a seek, and segment
	  after a segment.
	  Fixes #676242

2013-07-12 20:01:42 +0200  Arnaud Vrac <avrac@freebox.fr>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: always ignore HEAD errors
	  https://bugzilla.gnome.org/show_bug.cgi?id=704241

2013-07-25 14:26:07 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Clean up reset/start/stop handling

2013-07-25 14:13:10 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: Use base class error handling function instead of replicating it here

2013-07-25 14:12:56 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Clean up handling of reset/start/stop

2013-07-25 10:41:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/files/id3-407349-1.tag:
	* tests/files/id3-407349-2.tag:
	* tests/files/id3-447000-wcop.tag:
	  tests: fix test ID3 tags up not to rely on dodgy typefinding code
	  Change 0xff 0xfb 'mp3' marker to 'fLaC' marker, so we can fix
	  the typefinder.
	  https://bugzilla.gnome.org/show_bug.cgi?id=681368

2013-07-25 08:22:45 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: intersect the probed caps with the filter passed to get_caps()

2013-07-24 14:17:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  bin: fix compilation

2013-07-24 12:42:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  vrawdepay: fix UYVP format

2013-07-24 12:41:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  vrawpay: fix UYVP format

2013-07-24 12:41:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  vrawpay: fix caps

2013-07-24 10:49:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix locking
	  Take the lock earlier so that we do things that follow with the right
	  locking.

2013-07-23 17:40:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: don't use invalid times in RTCP timeouts
	  An invalid timeout can be calculated when we disabled RTCP by setting the
	  bandwidth to 0. Make sure all code can handle this case.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626

2013-07-23 17:38:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: lock session when changing bandwidth
	  Take the session lock when changing the bandwidth properties so that we don't
	  end up with inconsistent behaviour.

2013-07-23 17:37:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: reset some RTCP variables
	  The early_send time was set to 0 and always triggering an early RTCP packet.

2013-07-23 15:03:31 +0200  Edward Hervey <edward@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add all the mpeg XDCAM variants
	  This should cover all known XDCAM variants (which are all mpeg2 video)
	  Fixes #672227

2013-07-03 18:41:42 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: added custom downstream sync event
	  rtpbin can now send a custom in-band downstream event which informs
	  downstream that the bin has received an RTCP SR packet. This is useful
	  for applications which want to drop the initial unsynchronized received
	  RTP packets.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560
	  Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>

2013-07-22 18:00:16 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix on-the-fly changing of "mode" and "fields" properties
	  We call setcaps() to reconfigure ourselves, but we need to pass
	  the current *sink* caps, not the source caps then. Also fix a
	  caps leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=641599

2013-07-22 15:23:39 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Add support for group-id in the stream-start event

2013-07-22 15:23:20 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Add support for group-id in the stream-start event

2013-07-22 15:23:11 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Add support for group-id in the stream-start event

2013-07-22 15:22:55 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: Add support for group-id in the stream-start event

2013-07-22 15:22:47 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Add support for group-id in the stream-start event

2013-07-22 15:22:36 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Add support for group-id in the stream-start event

2013-07-22 15:22:16 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Add support for group-id in the stream-start event

2013-07-22 15:21:49 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: Add support for group-id in the stream-start event

2013-07-19 22:59:15 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: use gst_util_uint64_scale*_round.
	  There could be a case where:
	  1) you do a new set_caps after buffers have been processed.
	  2) ts_offset gets set to a different value, eg 0.033333333
	  3) your pads get EOS, but the check dor that doesn't work
	  because you use ts_offset + a truncated value < segment.stop
	  4) so in the next collected, you end up comparing for example:
	  0.9999999999 > 1., which is false and means you don't send EOS.
	  Also adds scale_round in two other places where it potentially could
	  have caused problems.

2013-07-15 17:55:19 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: Add WRLE support

2013-07-19 19:35:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: make files from Vivotek camera play
	  Skip tracks of 'vivo' subtype with empty stsd instead of
	  erroring out saying that the file is broken.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699791

2013-07-19 17:14:06 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: when streaming don't try to seek when stopping
	  It might cause errors in sinks that are not seekable and
	  have reported this (like e.g. fdsink)
	  https://bugzilla.gnome.org/show_bug.cgi?id=696228

2013-07-19 17:26:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: simplify some helpers
	  Some helper functions are not needed anymore or can be simplified.

2013-07-19 17:12:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: for non-raw video, move palette in caps
	  We only need to append the palette to raw video buffers, non-raw video has the
	  palette in the caps still.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-19 01:49:20 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: nitpicking in esds parsing

2013-07-19 01:49:07 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: set proper caps for mpeg-1 audio
	  Remove AAC specific fields from mpeg-1 audio caps, remove assumption
	  that the mpeg1 audio layer is 3, and set `parsed' field.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704548

2013-06-17 21:27:37 +0200  Arnaud Vrac <avrac@freebox.fr>

	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp8enc.h:
	* ext/vpx/gstvp9dec.h:
	* ext/vpx/gstvp9enc.h:
	  vpx: fix compilation when encoder or decoder headers are not installed
	  https://bugzilla.gnome.org/show_bug.cgi?id=704547

2013-07-16 20:41:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/videocrop.c:
	  videocrop: Fix unit for GRAY16 formats

2013-07-16 22:17:17 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: remove chapter stream
	  Remove all streams that are actually table of contents, since we will
	  never need the data after parsing them.

2013-07-16 21:59:37 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: send gap event for sparse streams in push mode
	  This allows to pre-roll at least if the next subtitle buffer
	  is far away.

2013-07-16 21:56:07 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: do not use indexes from sparse stream when seeking in push mode
	  This makes seeking more accurate in push mode, since the previous
	  keyframe on a sparse stream might be far away.

2013-07-16 21:04:07 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: advertise subtitle streams as sparse

2013-07-17 17:11:44 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/matroska/matroska-demux.c:
	  mastrokademux: do not push discont buffers if they aren't discont
	  Unset the discont flag instead of posssibly pushing a buffer with
	  a flag that's still set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=682110

2013-07-17 15:10:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: extract the palette from stsd
	  Sometimes a palette is inside the stsd, extract it instead of always using
	  the default one

2013-07-17 14:30:16 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/goom2k1/gstgoom.c:
	  goom2k1: Fix event handling and negotiate as soon as possible

2013-07-17 14:27:57 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/goom/gstgoom.c:
	  goom: Fix event handling and negotiate as soon as possible

2013-07-11 19:45:17 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: warn about the future deprecation of the "embed" property

2013-07-17 09:56:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for WRAW
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-17 09:54:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: palette is appended to buffers, not in caps
	  Fix the palette handling, in 1.0 we append the palette to the buffer instead of
	  placing it on the caps.
	  See also https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-16 15:37:49 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvpay.c:
	  rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders

2013-07-15 16:24:07 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: reset segment on flush stop
	  cca2f555d14 introduces a regression, where the demux segment is not
	  reset on flush stop, so the next upstream segment event will calculate
	  an invalid base time on the new segment to be sent downstream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704255

2013-07-06 17:20:49 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: offset samples according to edit list
	  https://bugzilla.gnome.org/show_bug.cgi?id=700264

2013-07-14 12:50:13 +1200  Douglas Bagnall <douglas@halo.gen.nz>

	* tests/examples/spectrum/spectrum-example.c:
	  level: Fix the spectrum example for 1.0
	  The "message" property has been replaced by "post-messages".
	  Pre-patch output:
	  (test_spectrum:23101): GLib-GObject-WARNING **: g_object_set_valist:
	  object class `GstSpectrum' has no property named `message'
	  New spectrum message, endtime 0:00:00.100000000
	  (test_spectrum:23101): GStreamer-CRITICAL **:
	  gst_value_list_get_value: assertion `GST_VALUE_HOLDS_LIST (value)' failed
	  [...]
	  Post-patch:
	  New spectrum message, endtime 0:00:00.100000000
	  band 0 (freq 400): magnitude -65.988777 dB phase 1.533397
	  band 1 (freq 1200): magnitude -65.545563 dB phase -0.780900
	  band 2 (freq 2000): magnitude -64.791946 dB phase -0.799611
	  band 3 (freq 2800): magnitude -64.556175 dB phase -0.063615
	  [...]
	  https://bugzilla.gnome.org/show_bug.cgi?id=704179

2013-07-13 20:56:26 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: be less verbose when parsing LOAS streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=704162

2013-07-12 12:31:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.h:
	  sink: alaw/mulaw caps don't have a layout property

2013-07-12 12:27:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseutil.c:
	  pulse: relax mulaw and alaw format checks
	  The audio library considers them as encoded formats and does not fill in the
	  sample width. The audio ringbuffers identifies the format as alaw/mulaw and that
	  is always 8 bits.

2013-07-11 16:13:05 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: unselect instead of ignoring disabled track, detect chapter track
	  https://bugzilla.gnome.org/show_bug.cgi?id=704007

2013-07-11 20:41:23 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: ignore errors from HEAD request
	  HEAD requests are used to check the server headers to see if it
	  seekable. Ignore errors from those requests as they shouldn't be
	  critical.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704053

2013-07-12 03:24:08 +0800  Kyosuke Nekomura <supercatexpert@gmail.com>

	* gst/audiofx/audioecho.c:
	  audioecho: Fix handling of delay property in PLAYING/PAUSED state
	  https://bugzilla.gnome.org/show_bug.cgi?id=703901

2013-07-09 17:56:57 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Enable proxy caps on the src pads

2013-07-11 16:57:15 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development

=== release 1.1.2 ===

2013-07-11 15:58:51 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.1.2

2013-07-11 15:58:29 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2013-07-09 15:34:04 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: defer the window handle setup to the main thread

2013-07-09 15:33:18 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: default to the main in case we are not setup yet

2013-07-07 22:16:05 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: close the internal window correctly

2013-07-07 21:14:22 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: only create the NS app thread for Cocoa once
	  The helper thread for Cocoa, in case no NS run loop is running,
	  should be started only once and shared across all the instances
	  running

2013-07-09 19:10:17 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: correct argument order in gst_util_uint64_scale_int_round
	  https://bugzilla.gnome.org/show_bug.cgi?id=703350

2013-07-09 17:42:59 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Keep caps order from the peer or the filter

2013-07-09 12:42:17 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixer2.c:
	  videomixer: Fix handling of buffers without a duration
	  We'll have to pop buffer from collectpads and store it
	  internally only to get the timestamp of the next buffer.
	  If we continue to keep it in collectpads, no new buffer
	  to calculate the end time will ever arrive.
	  https://bugzilla.gnome.org/show_bug.cgi?id=703743

2013-07-09 11:53:07 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixer2.c:
	  videomixer: Fix negotiation with 0/1 framerates
	  https://bugzilla.gnome.org/show_bug.cgi?id=703743

2013-07-09 11:17:59 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Unlock stream lock after use
	  Stream lock of sink pad was not unlocked after non-updating seek.

2013-06-27 13:26:31 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/multipart/multipartmux.c:
	  multipartmux: Re-set need_segment flag after FLUSH_STOP
	  https://bugzilla.gnome.org/show_bug.cgi?id=703182

2013-07-05 11:51:04 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: bufferpool: don't forget to release buffer on error
	  If the pool is stopped while gst_v4l2_buffer_pool_dqbuf() waits for a
	  buffer then the return value is GST_FLOW_FLUSHING. In this case the buffer
	  to queue must also be released. Otherwise is will never be deleted or
	  returned to its pool.
	  https://bugzilla.gnome.org/show_bug.cgi?id=703764

2013-07-08 14:15:10 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* tests/check/elements/rtp-payloading.c:
	  rtp: Fail payloading unit test if an error message is received

2013-07-08 14:09:37 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Don't pass upstream filter caps to downstream
	  Downstream usually can't accept video/x-h263 but only application/x-rtp,
	  so we would always get an empty intersection here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702632

2013-07-05 22:00:37 +0200  Piotr Drąg <piotrdrag@gmail.com>

	* po/POTFILES.in:
	  po: update POTFILES.in
	  https://bugzilla.gnome.org/show_bug.cgi?id=703685

2013-07-02 11:13:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid some strdup

2013-07-02 10:37:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add select-stream signal
	  Add a signal to let the app select what streams will be selected.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=634419

2013-07-02 10:37:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid strdup

2013-07-02 10:12:17 +0200  J. Rick Ramstetter <rick.ramstetter@gmail.com>

	* gst/rtp/README:
	* gst/rtpmanager/gstrtpbin.c:
	  rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin
	  https://bugzilla.gnome.org/show_bug.cgi?id=703426

2013-07-01 16:55:01 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: don't extract data from caps twice
	  gst_video_info_from_caps() always extract width, height, interlace mode and
	  framerate now. It is no longer necessary to do it again for encoded
	  formats.
	  https://bugzilla.gnome.org/show_bug.cgi?id=703399

2013-06-20 09:41:48 -0300  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: also consider stop positions in seeks
	  Use seek stop position as range end for requests
	  https://bugzilla.gnome.org/show_bug.cgi?id=702206

2013-06-19 14:06:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: allow seeks in ready
	  On is_seekable, check if the server's headers have already been
	  received. If not, do a HEAD request to get them before responding
	  to basesrc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702206

2013-07-01 17:28:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add signal to notify of the SDP
	  This way, the app can look and modify the SDP.

2013-06-21 18:10:28 +0200  Kishore Arepalli <kishore.arepalli@gmail.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: Allow negative offsets to specify offset from bottom/right
	  https://bugzilla.gnome.org/show_bug.cgi?id=702826

2013-06-30 21:01:20 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/Makefile.am:
	* gst/isomp4/qtdemux.c:
	  qtdemux: compute framerate from average sample duration
	  https://bugzilla.gnome.org/show_bug.cgi?id=703350

2013-06-25 21:16:38 +0200  Alban Browaeys <prahal@yahoo.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Add flvversion 1 to the flash-video caps
	  This allows using avdec_flv which requires this field to be
	  present in the caps. FLV only supports flash-video version 1
	  right now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=703076

2013-07-01 11:37:00 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/interleave/deinterleave.c:
	  deinterleave: Don't hold object lock while sending events downstream
	  Based on a patch by Kishore Arepalli <kishore.arepalli@gmail.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=703114

2013-07-01 10:59:07 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Add MPEG4 video profile/level to the caps

2013-07-01 10:56:28 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Add AAC profile/level to the caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=703312

2013-06-28 15:21:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvorbispay.h:
	  vorbispay: add support for config-interval
	  Align code with the theora payloader and add support for the config-interval to
	  periodically send out the config headers.

2013-06-28 15:21:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  theorapay: small cleanups

2013-06-28 12:08:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  theorapay: handle streamheaders as well

2013-06-28 12:06:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbispay.c:
	  vorbispay: always collect headers on data
	  When we see a data packet, always check if we need to collect any previous
	  headers.

2013-06-28 11:43:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbispay.c:
	  vorbispay: handle streamheader as well
	  Take config strings from the streamheader when we can
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=664312

2013-06-27 07:40:29 +0200  David Svensson Fors <davidsf@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: avoid double buffer unmap on error
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703171

2013-06-27 17:02:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: reset-sync before play
	  Call reset-sync on the rtpbin before we go to playing. This makes us require SR
	  packets for all streams again before we attempt to sync them. If we don't reset,
	  it might be that we combine SR packets from before and after the PAUSE/PLAYING
	  state change and end up with huge bogus offsets.

2013-06-27 16:23:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve sync on first packets
	  Don't throw away the first RTCP packet if it arrives before the first
	  RTP packet but remember and use it to signal sync once we get the
	  RTP packet.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=691400

2013-06-27 16:15:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: only signal loop when active
	  Only signal the loop function when it is active.

2013-06-27 16:13:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: signal timestamp discont
	  We can now use the RESYNC buffer flag to mark a timestamp discont when we update
	  the ts-offset property.

2013-06-26 20:49:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  jpegpay: turn some errors into warnings
	  Turn some errors into warnings, we can continue processing so this should
	  not be fatal.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=657079

2013-06-26 14:58:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: avoid some flushes

2013-06-26 14:41:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle data message when waiting for reply
	  When we are waiting for a server reply, handle data messages instead of
	  ignoring them.

2013-06-26 14:27:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle data messages in separate method
	  Refactor and make a method to handle a data message.

2013-06-25 20:36:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add some more docs to handle-request signal
	  See https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-10 17:20:30 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  Send a clock_provide message on the bus when we get a netclock

2013-06-10 17:20:14 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Expose use-pipeline-clock property

2013-06-24 17:11:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udpsink: bind to the given interface
	  Actually call BINDTODEVICE to bind to the interface as given by the
	  property.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702819

2013-06-22 10:59:17 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Error out gracefully if we get an unsupported color format
	  In theory we can only get I420 though, just to be on the safe side.

2013-06-22 10:57:41 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	  vp9: Add support for YV12, Y42B and Y444 color formats
	  The encoder does not work with Y42B and Y444 yet it seems.

2013-06-22 10:26:18 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/vpx/gstvp9dec.c:
	  vp9dec: Update default postproc settings from vp9_dx_iface.c

2013-06-21 13:11:32 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/webm-mux.c:
	  matroska: Add initial VP9 support

2013-06-21 13:07:30 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	* ext/vpx/Makefile.am:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9dec.h:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	* ext/vpx/plugin.c:
	  vpx: Add initial, experimental VP9 support

2013-06-21 10:32:30 +0200  Youness Alaoui <youness.alaoui at collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: go back into the loop after doing pause
	  After we do a pause request, go back to loop mode so that we can listen
	  for server messages again.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-20 23:16:17 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: Wait after the caps to forward the other events
	  First forward the stream-start, then the caps, then the rest

2013-06-21 00:42:02 +0100  Tim-Philipp Müller <tim@centricular.net>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: clear dts on buffer acquired from pool
	  When setting timestamps on outgoing buffers, clear the
	  dts explicitly, otherwise it may end up being set to a
	  bogus value from last time it was used. Avoids every
	  second or so buffer's dts being set to 0. Not that it
	  should matter for raw video.

2013-06-20 15:35:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2.c:
	  v4l2: don't redefine the PERFORMANCE debug variable
	  It is already defined in core.
	  fixes https://bugzilla.gnome.org/show_bug.cgi?id=702732

2013-06-20 14:43:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix race in state change to paused
	  When we go to paused, we first flush the connection and then send the pause
	  command. As a result of the flushing, the scheduled paused command can get
	  lost. Wait until the connection is completely flushed and the rtsp task is
	  waiting before issuing the paused or playing request.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-20 11:31:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: handle SEGMENT query

2013-06-19 12:37:31 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: Optimize negotiation by removing the query filter
	  As cameras tend to have a quite specific set of capabilities (specific
	  framerates for each resolution), getting the peer caps filtered by our
	  probed caps can cause a big increase in the caps size which slows down
	  things quire a bit.
	  As for negotiation v4l2 iterates through the caps of the peer to find the
	  first intersection with the probed caps, getting the fully expanded
	  intersection of capabilities is not useful.
	  Using the same testcase as for bug #702632, adding this patch on top of
	  the patches suggested there speeds up getting the inital frame from
	  around ~14-15 seconds to around ~3-4 seconds.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702638

2013-06-19 10:30:56 +0200  Kishore Arepalli <kishore.arepalli@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: duration query returns zero for DV video in avi
	  https://bugzilla.gnome.org/show_bug.cgi?id=702625

2013-06-19 11:06:37 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Disable usage of allocation queries
	  This can only reliably work if demuxers have a
	  separate streaming thread per srcpad. This should be
	  done in a demuxer base class, which integrates parts
	  of multiqueue
	  https://bugzilla.gnome.org/show_bug.cgi?id=701856

2013-06-11 15:02:21 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/qtdemux.c:
	  Avoid skipping moov atoms for fragmented MP4 files.
	  bug #700505
	  Following a representation change that causes a resolution change,
	  the video decoder fails to decode correctly. Dashdemux detects the
	  representation change and pushes a new caps event and an
	  initialization segment (a new moov atom) to the downstream qtdemux,
	  but it doesn't handle this new moov yet, it will only parse the
	  first one it receives.
	  This commit changes qtdemux to accept a new moov in a dash bitstream
	  switching scenario.

2013-06-19 00:42:54 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: send stream-start only once for each stream
	  Do not send stream start again when reconfiguring a pad for new caps.
	  That is common for adaptive streams

2013-06-05 17:02:49 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix support in VM's without hardware acceleration

2013-06-15 12:29:31 +0200  Jens Georg <mail@jensge.org>

	* gst/rtp/gstrtpmp2tdepay.c:
	  rtpmp2tdepay: accept mislabelled streams from GStreamer 0.10 as well
	  The mp2t payloader in 0.10 mislabelled the streams as MP2T-ES
	  instead of MP2T, so accept that as well for compatibility reasons.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702457

2013-06-16 05:40:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: manage element state ourselves
	  Lock the state of the all our elements and manage their states
	  outselves. Because we are working async, we can't rely on the state
	  change function to set the state at the right time or to return the
	  right return value from the state change function.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702046

2013-06-14 14:09:50 +0200  Bruno Gonzalez <stenyak@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't unlock stream lock without locking it first
	  https://bugzilla.gnome.org/show_bug.cgi?id=702167

2013-06-13 16:00:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Use the right hashtable to calculate bandwidth
	  Don't use an unused hashtable to iterate source to calculate bandwidth.
	  Remove unused code.

2013-06-12 16:27:24 -0600  Brendan Long <b.long@cablelabs.com>

	* configure.ac:
	  pulsesink: Require PulseAudio >= 2.0
	  This is needed for pa_format_info_get_prop_* functions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-06-13 14:23:08 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	  Revert "pulsesink: Make 2.0 dependency optional"
	  This reverts commit 01457027e0d384aca3e551ae684e0aa074ee5498.
	  We'll just depend on PulseAudio 2.0 or above instead of having the bug
	  partially fixed based on the installed libpulse version.

2013-06-13 12:40:15 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	  pulsesink: Make 2.0 dependency optional
	  The getcaps function we added uses some pa_format_info_get_prop...
	  accessor functions that were only added in 2.0, so we only have our
	  getcaps implementation exist if we're compiling against libpulse 2.0 or
	  above.
	  Eventually, we could bump the minimum requirement to 2.0 or above.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-06-12 18:23:46 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixer2.c:
	  Revert "videomixer: When all sinkpads are eos, update output segment stop and forward it"
	  This reverts commit 2d3910fc7901b5f29e16c0fdd4e9067a6d7f66fe.
	  It's not solving any problem and instead causes code to fall apart.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701519

2013-01-09 09:39:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: mark subtitle streams as sparse in stream-start event
	  And also mark the streams that should be selected by default if
	  marked so in the headers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=600648

2013-06-11 22:12:58 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanoramaorc-dist.c:
	* gst/audiofx/audiopanoramaorc-dist.h:
	  audiopanorama: add prebuilt files

2013-06-11 20:27:51 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c:
	  audiopanorama: cleanup and expand the tests
	  Split out two more tests. Extract more common code into helpers. Add coverage for float.

2013-06-10 21:15:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: cleanup of transform()
	  Only map input if we are reading it. Cleanup the logging and the comments a bit.

2013-06-09 20:35:18 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiopanorama.h:
	* gst/audiofx/audiopanoramaorc.orc:
	  audiopanorama: use orc to speedup processing
	  Use special variants for the case when we don't change the panorama (pan=0.0).
	  Simplify the processing functions by passing the panorama value directy instead
	  of the instance. Use orc for clearing buffers too.

2013-06-11 19:24:49 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: check last end_time after conversion to running segment
	  The last end_time was saved after conversion, so the comparison
	  had to be made after conversion for it to make sense.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701385

2013-06-11 19:22:20 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: add mix->segment.start to output_end_time
	  When the segment start is not 0, this created a situation where
	  the output_end_time is inferior to output_start_time, and the duration
	  of the next buffer ended up underflowing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701385

2013-06-11 13:54:53 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Send stream headers after the segment event
	  https://bugzilla.gnome.org/show_bug.cgi?id=700799

2013-06-11 12:26:24 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Do allocation query after exposing all pads and no-more-pads
	  Also configure video streams as early as possible.
	  Related https://bugzilla.gnome.org/show_bug.cgi?id=701856
	  but not fixing that.

2013-06-11 12:25:46 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't forward CAPS events from upstream
	  Just use the default pad event handler.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701976

2013-05-26 08:18:04 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Cache the getcaps/acceptcaps probe stream
	  getcaps is called frequently during stream setup, and creating a new
	  stream each time is very inefficient. There's some more room for
	  optimisation by caching the queried sink formats as well, but this needs
	  some more changes to listen for format changes on the sink (for when
	  supported formats change between probe stream creation and sink
	  querying).
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-05-23 21:39:08 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulsesink: Add a getcaps function
	  This allows us to have more fine-tuned caps in READY or above. However,
	  this is _really_ inefficient since we create a new stream and query sink
	  for every getcaps in READY, which on a simple gst-launch line happens
	  about 35 times. The next step is to cache getcaps results.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-05-10 11:32:44 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Take a lock on the ringbuffer in acceptcaps
	  This is needed as a concurrent state change could pull the context or
	  stream out from under our feet.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-06-09 20:29:09 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiopanorama.h:
	  audiopanorama: move the enum to the header and use instead of gint
	  Move the enum for the processing method to the header so that we can use the
	  type for the instance struct.

2013-06-09 20:32:22 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/level.c:
	  level: rework the tests to cover other formats too

2013-06-05 16:32:30 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: make sure the element is not deleted before the pool
	  The pool accesses data from the v4l2object so it must exist at least
	  as long as the pool. Refcount the element which controls the object
	  live-time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701650

2013-06-07 15:38:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/libpng/Makefile.am:
	  png: Link with libgstbase for GstByteReader and GstAdapter

2013-06-07 15:15:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavenc/Makefile.am:
	  wavenc: Link with libgstbase for GstByteWriter

2013-06-07 13:26:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Push stream-start event in pull mode before anything else

2013-05-10 12:09:19 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Get rid of acceptcaps side-effects
	  The sink info callback should not have side-effects on the GstPulseSink
	  object since we are sometimes using with a dummy stream in acceptcaps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-06-05 18:36:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  Back to development

=== release 1.1.1 ===

2013-06-05 17:58:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* common:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	* win32/common/config.h:
	  Release 1.1.1

2013-06-05 16:35:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2013-06-05 15:50:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Fix taglist ref handling that made the unit test fail

2013-06-05 15:14:54 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* common:
	  Automatic update of common submodule
	  From 098c0d7 to 01a7a46

2013-06-03 09:17:43 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: iterate controls with V4L2_CTRL_FLAG_NEXT_CTRL if possible
	  In v2.6.18 control classes where added to the v4l2 API.
	  Iterating over CIDs starting with V4L2_CID_BASE will only find controls for
	  the first control class.
	  By iterating with V4L2_CTRL_FLAG_NEXT_CTRL all controls are found.
	  This is necessary to make controls from other control classes available in
	  the extra-controls property.
	  If V4L2_CTRL_FLAG_NEXT_CTRL is not defined at compile time or not supported
	  at runtime then the old mechanism for iterating is used.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701540

2013-06-05 12:12:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsink.c:
	  udpsink: avoid leaking the host
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701586

2013-06-04 08:26:33 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: improve pixel aspect ratio handling
	  Instead of just assuming a aspect ratio of 1/1 use VIDIOC_CROPCAP to ask
	  the device.
	  This also add a pixel-aspect-ratio property to overwrite the value from the
	  driver and a force-aspect-ratio property to ignore it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700285

2013-06-04 17:04:11 +0200  Stirling Westrup <swestrup@gmail.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Fix compilation with older kernels
	  https://bugzilla.gnome.org/show_bug.cgi?id=701595

2013-06-03 17:07:10 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: call VIDIOC_REQBUFS with count = 0 in pool_finalize
	  Without this the following sequence fails:
	  - set_caps()
	  - object_stop() (does nothing)
	  - set_format() -> VIDIOC_S_FMT
	  - set_config() -> VIDIOC_REQBUFS with count = N
	  - set_caps()
	  - object_stop()
	  - pool_finalize()
	  - set_format() -> VIDIOC_S_FMT => EBUSY
	  Usually the pool is started after set_config(), in which case object_stop()
	  will result in a pool_stop and therefore VIDIOC_REQBUFS with count = 0 but
	  that is not guaranteed.
	  Also calling VIDIOC_REQBUFS with count = 0 in pool_finalize() if necessary
	  fixes this problem.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701543

2013-05-28 19:14:15 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: rework sink buffer refcounting
	  This is a followup patch for #700781, which is not quite correct.
	  The buffer handling is quite complicated here.
	  The original code intended to the the following:
	  - gst_v4l2_buffer_pool_process() calls QBUF and adds the buffer to the
	  local list.
	  - The sink calls gst_buffer_unref() which returns the buffer to the pool
	  but not the 'free list'.
	  - Some time later DQBUF returns the buffer and
	  gst_v4l2_buffer_pool_release_buffer() puts in on the 'free list'.
	  If the buffer must be copied then (parent_class)->acquire_buffer() is
	  called directly to keep the buffer in the pool.
	  This has two problems:
	  1. If gst_v4l2_buffer_pool_release_buffer() is called before the buffer is
	  returned to the pool, then the buffer is put on the 'free list' twice.
	  This can happen if a reference to the buffer is kept outside the sink,
	  of if DQBUF returns the buffer, that was just queued with QBUF.
	  2. If buffers are copied, then all buffers are in the pool at all times. As
	  a result gst_v4l2_buffer_pool_stop() and gst_v4l2_buffer_pool_dqbuf()
	  can access pool->buffers at the same time, which can lead to memory
	  corruption.
	  The patch for #700781 fixes those problems, but with the side effect that
	  there are always buffers outside the pool (because they are queued) and
	  the pool is never stopped.
	  This patch fixes this by releasing the reference to the buffer after
	  handling it (to avoid problem 2.) so it can be returned to the pool.
	  gst_v4l2_buffer_pool_release_buffer() is only called if the buffer is
	  already in the pool (to avoid problem 1.).
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701375

2013-06-02 15:24:38 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: make sure taglist is writable before adding tags
	  Avoids assertions

2013-05-30 19:24:13 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: effectively skip tracks that weren't listed on the 1st moov
	  Without this, stream is NULL and the code will try to access it, leading
	  to segfaults.

2013-05-30 19:23:50 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: skip redundant check
	  !got_moov is already checked the line above

2013-06-02 13:03:40 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/level.c:
	  tests: cleanup level tests
	  Split out a few more tests to avoid checking the same stuff over and over again.

2013-06-01 21:33:46 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.h:
	  level: remove unused variables in instance struct

2013-05-31 18:13:02 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/level.c:
	  level: add a test for continous timestamps
	  A test that checks that msg[n].ts + msg[n].dur == msg[n+1].ts.

2013-04-12 16:02:44 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	  wavenc: add tags & toc support
	  Write tags as LIST INFO chunk. Format the toc as cue + LIST adtl chunk. Remove
	  old #ifdef'ed code.

2013-05-31 15:12:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  Revert "rtph264pay: Restructuring to allow for adding optional caps"
	  This reverts commit 61666898cfe89a1b21d3e6850ab44f5b1633ed79.
	  This commit changes what the set_sps_pps() function does, not it doesn't
	  set caps anymore (and should have been renamed). The main problem is that
	  not all call sites are updated and thus leak the string.

2013-05-31 15:11:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.c:
	* tests/check/elements/rtp-payloading.c:
	  Revert "rtph264pay/depay: Add frame dimensions a payloaded caps"
	  This reverts commit 3dca756a5dba55266256f239e3e12a3d058e185a.
	  The H264 RTP spec has no attributes for width and height.

2013-05-31 15:09:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.c:
	  Revert "rtph264pay/depay: Add optional framerate caps for use in SDP"
	  This reverts commit d8825e2a5c0bfb883ff88e2c9da499c800ebca0a.
	  There is no framerate attribute in the h264 RTP spec.

2013-05-31 15:08:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	  Revert "rtpjpegpay/depay: Replace framesize caps with width/height"
	  This reverts commit 0075d111b475ca27895ee9476154260b6902940b.
	  Extra application/x-rtp are SDP fields, which are strings.

2013-05-31 15:05:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* tests/check/elements/rtp-payloading.c:
	  Revert "rtpjpegpay/depay: Replace framerate caps field with fraction"
	  This reverts commit 9fd25a810b859e0ec205176578735100d83de4af.
	  We deal with sdp attributes in application/sdp, which are always strings.

2013-05-31 12:33:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add extra TLS url protocols
	  We also support TLS protocols now.

2013-05-30 14:48:42 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixer2.c:
	  videomixer: Add FIXME comment about the DURATION query from adder
	  Currently the code just takes with maximum upstream duration, which
	  is wrong. It should be the maximum upstream duration in running time.

2013-05-30 21:20:59 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: Set a reference to mix->current_caps as the QUERY_CAPS result.

2013-05-30 17:37:13 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: misc cleanups
	  Fix some oudated comments. Sort out some confusion of interval_frames and num_frames.

2013-05-29 20:35:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Only conditionally use V4L2_CTRL_TYPE_INTEGER_MENU, it's not available in older versions

2013-05-20 16:45:37 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	  v4l2: add a property for arbitrary v4l2 controls
	  This makes it possible to set any controls that can be set with
	  VIDIOC_S_CTRL.
	  The controls are set when the property is set (if the device is open)
	  and when the device is opened.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698837

2013-05-28 18:31:07 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: fix discontinuities in timestamps

2013-05-28 15:46:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkanimation.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.h:
	  gdkpixbufdec: Keep serialized events in order, and don't send SEGMENT before CAPS

2013-05-28 15:45:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: create and push stream-start in TCP mode

2013-05-28 15:10:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove some obsolete code
	  It is not needed to do a state change from the _play() function on
	  ourselves. The state change function already did that and we don't want to
	  interfere with that (or use hacks to avoid interference).

2013-05-28 12:24:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: set RTCP caps on the RTCP pads

2013-05-28 12:23:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: send stream-start and segment events
	  Also send stream-start and segment event on the RTCP pad.
	  We don't need to send anything on the sync_src pad because we
	  already forwarded all incomming events.

2013-04-25 15:25:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add signal to handle server requests
	  Add a signal to be notified of a server request. The signal handler can then
	  construct the response message for the server.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=632207

2013-05-27 22:43:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: Maintain z-order when new pad are added
	  https://bugzilla.gnome.org/show_bug.cgi?id=701109

2013-03-06 13:17:54 +0000  Tom Greenwood <tcdgreenwood@hotmail.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	  vp8enc: Add property to manually specify the timebase of the encoder
	  https://bugzilla.gnome.org/show_bug.cgi?id=695709

2013-05-25 12:17:40 -0400  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Always handle flush_stop_pending atomically
	  It is not protected with the COLLECT_PADS_STREAM_LOCK anymore

2013-05-23 18:14:17 -0400  Thibault Saunier <thibault.saunier@collabora.com>

	* tests/check/Makefile.am:
	* tests/check/elements/videomixer.c:
	  tests: videomixer: Add a testsuite for videomixer
	  This is mostly copy pasted from -base/tests/check/elements/adder.c

2013-05-25 10:57:02 -0400  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Do not take COLLECT_PADS_STREAM_LOCK when unnecessary
	  Collectpad takes the lock itself when receiving serialized events
	  and we should not take it for not serialized ones

2013-05-24 19:34:05 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/flx/gstflxdec.c:
	  flxdec: Properly skip non-frame chunks

2013-05-24 19:31:14 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/flx/gstflxdec.c:
	  flxdec: Flush data from adapter after reading it
	  Otherwise we're going in an infinite loop, reading the same data
	  over and over again.

2013-04-24 15:39:54 +0000  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/goom2k1/Makefile.am:
	  goom2k1: fix more duplicated symbols

2013-05-22 02:40:52 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtpjpegpay/depay: Replace framerate caps field with fraction
	  The previous implementation had the formatting of SDP attributes happen
	  in each RTP payloader, now instead the constituent values are propagated
	  as caps fields. This allows for applications to do SDP offer/answer
	  based on caps negotiation.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748

2013-05-22 01:58:57 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay/depay: Replace framesize caps with width/height
	  The previous implementation had the formatting of SDP attributes happen
	  in each RTP payloader, now instead the constituent values are propagated
	  as caps fields. This allows for applications to do SDP offer/answer
	  based on caps negotiation.
	  Keep parsing a-framerate, x-framerate and x-dimensions in rtpjpegdepay
	  to be backwards compatible with previous payloaders.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748

2013-05-22 03:18:07 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.c:
	  rtph264pay/depay: Add optional framerate caps for use in SDP
	  This allows for applications to format SDP attributes and still do SDP
	  offer/answer based on caps negotiation.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749

2013-05-22 03:09:44 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtph264pay/depay: Add frame dimensions a payloaded caps
	  This allows for applications to format SDP attributes and still do SDP
	  offer/answer based on caps negotiation.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749

2013-05-20 22:14:44 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Restructuring to allow for adding optional caps
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749

2013-05-23 18:42:09 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  (dyn|multi)udpsink: Add properties to specify the bind address and port
	  By default we use the any addresses and a random port for binding the socket.

2013-05-23 18:05:07 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	  (dyn|multi)udpsink: Bind socket before using it
	  https://bugzilla.gnome.org/show_bug.cgi?id=700878

2013-05-23 17:25:29 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/udp/gstmultiudpsink.c:
	  (multi)udpsink: Add missing getters for socket-v6 and used-socket-v6 properties

2013-05-22 21:01:48 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't hold stream-lock while pushing non-serialized events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700868

2013-05-22 21:00:45 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't hold object lock while sending events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700868

2013-05-22 17:32:33 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: The return value of gst_pad_set_caps() is not relevant anymore
	  Caps can fail to be set because the pad is not linked yet for example.

2013-05-15 16:39:36 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add error if file has playready drm

2013-05-18 15:06:49 -0400  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Send a reconfigure event upstream if sinkpad caps are not usable
	  https://bugzilla.gnome.org/show_bug.cgi?id=684237

2013-05-21 12:02:51 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: keep a reference to all queued buffers
	  Without this, a queued buffer may be required, filled and queued before it
	  is dequeued.
	  Calling gst_buffer_pool_acquire_buffer() ensures that the buffer is set up
	  correctly and gst_buffer_unref() calls buffer_release().
	  https://bugzilla.gnome.org/show_bug.cgi?id=700781

2013-05-21 13:33:59 +0200  Alexander Schrab <alexas@axis.com>

	* gst/law/mulaw-decode.c:
	  mulawdec: Handle NULL buffers in handle_frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=698894

2013-05-20 21:44:13 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay/depay: Add framesize caps for use in SDP
	  The format of the value adheres to RFC6064 and it is meant to be parsed
	  and included in the SDP sent by gst-rtsp-server to its clients.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748

2013-05-20 21:34:13 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Add optional framerate caps for use in SDP
	  The format of the value adheres to RFC4566 and it is meant to be parsed
	  and included in the SDP sent by gst-rtsp-server to its clients.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748

2013-05-20 19:59:13 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: When all sinkpads are eos, update output segment stop and forward it
	  https://bugzilla.gnome.org/show_bug.cgi?id=699793

2013-05-20 19:51:07 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't reset the output segment on flush stop
	  Only init it when getting from READY to PAUSED, and change it on seek events.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699793

2013-05-17 10:16:48 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Don't stop streaming when set_caps is called with unchanged caps
	  This can happen if other parts of the pipeline are reconfigured.
	  Stop streaming even for a short amount of time can be quite visible, so it
	  should be avoided if possible.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700503

2013-05-18 15:39:36 -0400  Thibault Saunier <thibault.saunier@collabora.com>

	* tests/check/pipelines/simple-launch-lines.c:
	  tests: Re-enable videomixer test
	  https://bugzilla.gnome.org/show_bug.cgi?id=684237

2013-05-18 14:36:39 -0400  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: Send caps event from the streaming thread
	  This way we avoid races in caps negotiation and we make sure
	  that the caps are sent after stream-start.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684237

2013-05-05 20:25:20 +0100  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Do not send flush_stop when receiving a seek
	  There is no reason to send a flush-stop when receiving a seek event.
	  In the case of a flushing seek, we could eventually want to, but in
	  the code path were we check if the seek is "flushing", we have the
	  following comment that makes sense:
	  "we can't send FLUSH_STOP here since upstream could start pushing data
	  after we unlock mix->collect.
	  We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
	  forwarding the seek upstream or from gst_videomixer_collected,
	  whichever happens first."
	  https://bugzilla.gnome.org/show_bug.cgi?id=684237

2013-05-05 20:24:49 +0100  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Protect flush_stop_pending with the collectpad stream lock
	  And make sure to expect a flush-stop after a flush-start
	  https://bugzilla.gnome.org/show_bug.cgi?id=684237

2013-05-17 12:37:59 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/rtp/gstrtpmp4apay.c:
	  rtpmp4apay: clear config buffer before using it
	  This is necessary because parts of the memory are only modified with "|="
	  https://bugzilla.gnome.org/show_bug.cgi?id=700514

2013-05-14 17:30:07 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Do not expect EOS after a segment event if upstream is mss
	  In case qtdemux is handling a mss stream, do not mark the stream to wait
	  for EOS after a segment. Even if it seems to be the last one according to
	  the current streams information.
	  MSS handling is different here because there is another demuxer driving
	  the pipeline

2013-05-14 16:32:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only set channels and rate if qtdemux knows it
	  Setting both of those to 0 is pointless and means that qtdemux
	  doesn't know the real value. Avoid setting it in this case.

2013-05-14 15:23:08 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: set alac caps using info from codec buffer
	  The samplerate field in the STSD atom is not right for some ALAC files
	  (usually when audio is 96kHz/24bits), so the audio caps must be
	  extracted from the codec data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700382

2013-05-15 11:13:12 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/avi/gstavidemux.c:
	  avidemux: do not push discont buffers if they aren't discont
	  https://bugzilla.gnome.org/show_bug.cgi?id=682110

2013-05-15 10:51:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 5edcd85 to 098c0d7

2013-05-14 10:28:10 -0400  Joshua M. Doe <oss@nvl.army.mil>

	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	  videocrop: Add support for GRAY16_LE/GRAY16_BE
	  https://bugzilla.gnome.org/show_bug.cgi?id=700331

2013-05-14 17:29:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/replaygain/gstrgvolume.c:
	  rgvolume: Send all events through the proxypads instead of just sending to the target
	  Otherwise the sticky events are missing on the proxypads.

2013-05-14 17:29:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/rgvolume.c:
	  rgvolume: Fix event handling in the unit test

2013-05-14 16:34:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/rglimiter.c:
	  rglimiter: Fix event handling in unit tests

2013-05-14 16:31:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/rganalysis.c:
	  rganalysis: Fix event handling in unit test

2013-05-14 16:08:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  qtmux: Fix event handling in unit test

2013-05-14 16:00:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/multifile.c:
	  multifile: Fix event handling in unit test

2013-05-14 13:58:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/mulawdec.c:
	* tests/check/elements/mulawenc.c:
	  mulaw: Fix event handling in unit test

2013-05-14 13:52:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: Make sure to send a segment event before dataflow

2013-05-14 10:52:19 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: only add interlace-mode to the caps for raw formats
	  https://bugzilla.gnome.org/show_bug.cgi?id=700280

2013-05-14 12:03:03 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: copy and set the actual size of the content
	  https://bugzilla.gnome.org/show_bug.cgi?id=700282

2013-05-14 10:25:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/interleave.c:
	  interleave: Fix event handling in unit test

2013-05-14 09:45:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Improve handling of min/max buffer numbers of the buffer pool

2013-05-14 03:42:59 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: set caps for buffer pool config

2013-05-13 13:30:38 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: Let the base class do get_times
	  This will make sync=TRUE work, the default is still sync=FALSE

2013-05-11 23:08:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/interleave/interleave.c:
	  interleave: Send stream-start before caps event

2013-05-11 23:24:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* tests/check/elements/rtpmux.c:
	  rtpmux: Send stream-start before caps

2013-05-11 23:28:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer-test: Send stream-start before caps followed by segment

2013-05-11 23:34:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  rtpbin-test: Send missing stream-start and segment events

2013-05-13 15:36:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/level.c:
	* tests/check/elements/matroskamux.c:
	  tests: Fix some more event handling in tests

2013-05-13 15:19:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/icydemux.c:
	  icydemux: Fix event handling in unit test

2013-05-13 15:19:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/icydemux/gsticydemux.c:
	  icydemux: Fix sticky event handling

2013-05-13 15:06:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Push sticky events in the right order

2013-05-13 14:55:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/deinterleave.c:
	  deinterleave: Fix event handling in test

2013-05-13 14:07:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/interleave/deinterleave.c:
	  deinterleave: Fix sticky event handling

2013-05-13 13:55:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/interleave/deinterleave.c:
	  deinterleave: Code style fixes

2013-05-13 10:43:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: First let baseclass handle events, then put them into the stream
	  Fixes handling of sticky events.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700213

2013-05-09 22:05:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* tests/check/elements/shapewipe.c:
	  shapewipe-test: Send inital events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-09 18:32:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/spectrum.c:
	  spectrum-test: Send inital events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-09 18:25:17 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/videofilter.c:
	  videofilter-test: Send inital events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-09 18:23:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/wavpackparse.c:
	  wavpackparse-test: Send inital events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-09 18:21:54 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/y4menc.c:
	  y4menc-test: Send inital events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-10 14:00:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: fix example pipeline
	  Need jpegparse.

2013-05-10 13:34:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/alphacolor.c:
	* tests/check/elements/aspectratiocrop.c:
	* tests/check/elements/audioamplify.c:
	* tests/check/elements/audiochebband.c:
	* tests/check/elements/audiocheblimit.c:
	* tests/check/elements/audiodynamic.c:
	* tests/check/elements/audioecho.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/audiopanorama.c:
	* tests/check/elements/audiowsincband.c:
	* tests/check/elements/audiowsinclimit.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/avisubtitle.c:
	* tests/check/elements/capssetter.c:
	* tests/check/elements/deinterlace.c:
	* tests/check/elements/dtmf.c:
	* tests/check/elements/equalizer.c:
	  tests: Fix some more unit tests

2013-05-10 13:10:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/parser.c:
	  tests: Fix parser tests

2013-05-09 22:20:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: Can't map twice the same buffer for writing
	  I took the opportunity to simplify that code a bit. We now use
	  gst_buffer_make_writable() to make the buffer writable and map twice the
	  same buffer, with first map being read/write, and second read only. This
	  get rid of the critical:
	  GStreamer-CRITICAL **: gst_structure_set_name: assertion `IS_MUTABLE
	  https://bugzilla.gnome.org/show_bug.cgi?id=700044

2013-05-09 22:15:54 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: Ensure caps are writable
	  The exist one case where that we endup with original caps in ret, in which
	  case we are not guaratied to have writable caps. Simply ensure this is the
	  caps are writable before entering the loop.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700044

2013-05-09 22:13:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: Fix sample pipeline in documentation
	  https://bugzilla.gnome.org/show_bug.cgi?id=700044

2013-05-09 18:05:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/jpegenc.c:
	  jpegenc-test: Send inital events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-09 17:49:03 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/vp8enc.c:
	  vp8enc-test: Send inital events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-09 17:20:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/vp8dec.c:
	  vp8dec-test: Send inital events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-09 17:19:53 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/wavpackdec.c:
	  wavpackdec-test: Send initial events
	  https://bugzilla.gnome.org/show_bug.cgi?id=700033

2013-05-09 19:40:49 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: Tell GstAudioEncoder about the number of incoming samples
	  lame does internal resampling, but the base class only cares about
	  the number of raw samples, so tell finish frames about that, not
	  the number of samples in the outgoing frame.:

2013-05-09 16:26:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  Revert "videomixer2: Take into account new segments"
	  This reverts commit 84ae670ab40b258a10e1e21471e6dc9d786bf086.
	  Actually this is not how it is supposed to work. videomixer
	  creates a [0,-1] segment and then puts frames of the different
	  streams there based on their running times in their own segments.

2013-05-06 23:43:03 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Take into account new segments
	  Also forward the event downstream on the next opportunity.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699793

2013-05-09 09:07:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtsp/gstrtspsrc.c:
	  Revert "gstrtspsrc: set buffer-size for multicast buffers"
	  This reverts commit 2481e95d038b42297a016f1d2dc1af26d2175b42.
	  This is already done five lines above, it was added a year
	  ago in commit 561b131e.

2013-05-08 19:54:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* tests/check/elements/videofilter.c:
	  videofilter: Unit test send SEGMENT before CAPS
	  https://bugzilla.gnome.org/show_bug.cgi?id=699966

2013-05-08 19:22:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* tests/check/elements/avimux.c:
	  avimux: Unit test sends SEGMENT before caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=699966

2013-05-08 19:08:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* tests/check/elements/audiowsincband.c:
	  audiowsincband: Test should send segment after CAPS
	  This makes the unit test pass again.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699966

2013-05-08 19:00:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* tests/check/elements/audiowsinclimit.c:
	  audiowsinclimit: Test should send segment after CAPS
	  This makes the unit test pass again.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699966

2013-05-08 18:44:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/audiofx/audiowsinclimit.c:
	  audiowsinclimit: Frequence property renamed cutoff
	  Updating the documentation to reflect this change.
	  See: https://bugzilla.gnome.org/show_bug.cgi?id=699964

2013-05-08 15:25:58 -0300  Aha Unsworth <aha.unsworth@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  gstrtspsrc: set buffer-size for multicast buffers
	  For receiving video data via RTSP when the video is sent via
	  multicast there is no way to specify the udpsrc buffer-size.
	  On windows the native network buffer is not large and with video
	  i-frames being huge the buffer is to small and you get i-frame corruption,
	  it looks terrible, and there is no (easy) way to set the udpsrc buffer-size.
	  https://bugs.freedesktop.org/show_bug.cgi?id=52264

2013-05-08 16:02:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Send stream-start before caps event
	  https://bugzilla.gnome.org/show_bug.cgi?id=699895

2013-05-07 19:15:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix compiler warning on type check

2013-04-18 07:49:54 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: push new caps events when caps change
	  Whenever the demuxer has a new caps on a stream, it should set the
	  new_caps variable to true and a new caps event will be pushed before
	  the next buffer

2013-04-17 16:54:22 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: do not push discont buffers if they aren't discont
	  qtdemux takes its buffers from a GstAdapter. Those buffers are created
	  from the larger buffer that it obtained from upstream and they carry
	  the same flags, including DISCONT if it is set. In these cases, all
	  buffers that qtdemux is going to push would be marked as DISCONT.
	  This scenario can make parsers/decoders flush on every buffer leading
	  to no decoding at all hapenning. This patch prevents this by unsetting
	  the flag if it shouldn't be set.

2013-04-12 09:08:16 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: some code cleanup for mss handling code
	  * Explicitly init variables for fragmented formats at init
	  * Do not use GstClockTime type if the variable isn't a timestamp
	  * Fix a style/readability issue at an if block
	  * Group 2 mss mode conditional blocks together to improve readability
	  Conflicts:
	  gst/isomp4/qtdemux.c

2013-04-12 10:21:11 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid storing non-time newsegments to push later
	  This can confuse downstream when they get a byte segment after receiving
	  the natural time segment from qtdemux that it sends when starting to
	  push buffers. This is specially the case with parsers that try to
	  convert the position from byte to time format and might miss the
	  correct position for playback to start.

2013-04-10 18:02:28 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid setting fields to non-writable caps

2013-03-10 04:15:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't send so many segment events
	  Only send one segment event in the beginning of the stream, not
	  after each moov and moof atom.
	  Conflicts:
	  gst/isomp4/qtdemux.c

2013-03-08 16:02:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: place incomming timestamps on output
	  Place the incomming timestamp (if any) directly onto the outgoing buffers
	  and interpollate other timestamps.
	  Conflicts:
	  gst/isomp4/qtdemux.c

2013-05-07 10:16:18 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: improve reset of internal status
	  Reset different variables on state changes to ready and when
	  handling a flush-stop. For handling flush stops we should check
	  if there is an upstream adaptive demuxer driving the pipeline as this
	  means that qtdemux will get a new moov atom. For 'standard' isomedia
	  streams this isn't true and qtdemux should keep the previous moov
	  information around.
	  Conflicts:
	  gst/isomp4/qtdemux.c

2013-02-08 00:29:20 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: prepare qtdemux to accept multiple dash moovs in a row
	  Whenever dashdemux switches bitrates it sends a new moov with the
	  new stream configuration. qtdemux should now handle this by splitting
	  the exposing and configuration of streams into separate functions. When
	  the stream is new it is configured and exposed, when it is a new bitrate
	  of an existing stream it is only reconfigured.
	  Conflicts:
	  gst/isomp4/qtdemux.c

2013-02-07 14:12:53 -0200  Andre Moreira Magalhaes (andrunko) <andre.magalhaes@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Move FLUSH_STOP/PAUSED_TO_READY handling to a reset method.
	  Conflicts:
	  gst/isomp4/qtdemux.c

2013-01-23 10:55:33 -0500  Louis-Francis Ratté-Boulianne <louis-francis.ratte-boulianne@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Remove old pads when exposing streams and other general fixes.
	  Conflicts:
	  gst/isomp4/qtdemux.c

2013-04-16 10:41:43 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: handle mss streams
	  smoothstreaming streams should be handled as a special kind of
	  fragmented isomedia. In MSS the fragments will not contain a
	  'moov' atom with the media descriptions, this has to be extracted
	  from the caps.
	  Additionally, there should be another demuxer upstream that is likely
	  going to be the one to answer/act on queries and events, so qtdemux has
	  to forward those upstream.

2013-05-06 16:54:02 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: request 0 buffers when stopping
	  Without this stopping the pool in *_set_caps() is useless.
	  S_FMT will still fail with EBUSY.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699835

2013-05-07 16:32:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: By default assume that we're working on non-packetized input
	  Only detecting this in set_format() does not work because we might
	  not get any caps at all, e.g. from filesrc.

2013-05-07 16:30:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	  pngdec: Implement parsing functionality
	  This allows to plug pngdec directly without a parser if that
	  is desired.
	  Parsing code is based on pngparse.

2013-05-07 15:54:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/libcaca/gstcacasink.c:
	  cacasink: Fix support for RGB formats and add support for more of them

2013-05-04 13:19:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't consider the content size from the HTTP headers as absolutely correct
	  The HTTP server could give wrong information, e.g. if the HTTP stream is
	  chunk-encoded or compressed, or if the server does not know the complete size
	  at the time when the file is requested by the client.
	  Also see
	  https://bugs.webkit.org/show_bug.cgi?id=115354

2012-08-20 09:52:32 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: fill out v4l2_buffer.bytesused field for v4l2sink
	  When queuing a buffer for a sink, bytesused must contain the actual
	  amount of data.
	  For a source, the driver must overwrite this, so it doesn't matter
	  what is set here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699598

2013-05-03 23:43:26 +0200  Sebastian Rasmussen <sebras@gmail.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: fix invalid memory access in event handler
	  First process event in payloader, then hand it to the
	  base class which takes ownership of the event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699637

2013-05-04 09:48:02 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstdcaparse.c:
	  ac3parse, dcaparse: check buffer size before trimming
	  and unref old buffer as soon as possible.

2013-05-02 15:00:22 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstdcaparse.h:
	  dcaparse: add support for "audio/x-private1-dts"

2013-05-02 14:56:02 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	  ac3parse: add support for "audio/x-private1-ac3"

2013-05-03 12:46:37 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: always generate video info from caps
	  In the past gst_video_info_from_caps() only video/x-raw. Now it also
	  supports other video/* and image/* formats.
	  With this patch the format won't be GST_VIDEO_FORMAT_UNKOWN and
	  gst_v4l2_buffer_pool_set_config() handles strides correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699570

2013-05-02 09:41:01 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: try to allocate new buffers with VIDIOC_CREATE_BUFS if needed
	  If max_buffers is 0 then an arbitrary number of buffers (currently 4) is
	  allocated. If this is not enough v4l2src starts copying buffers.
	  With this patch VIDIOC_CREATE_BUFS is used to allocate a new buffer. If
	  this fails v4l2src falls back to copying buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699447

2013-04-15 17:37:01 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix setting window handle after transition
	  The destroyed flag was not reset properly and it's also not needed
	  as we can check osxwindow != NULL

2013-05-02 13:45:55 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/rtp/Makefile.am:
	  rtp: fix duplicated symbols with libvpx

2013-04-29 10:58:08 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/goom2k1/Makefile.am:
	  goom2k1: fix duplicated symbols with goom

2013-05-01 15:49:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: If the adapter is empty on EOS don't try to map its content
	  https://bugzilla.gnome.org/show_bug.cgi?id=699314

2013-04-30 14:36:38 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: add stream-format=raw to aac caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=699303

2013-04-30 13:07:37 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: fix and cleanup VIDIOC_EXPBUF handling
	  clear the struct, and provide a correct error message
	  https://bugzilla.gnome.org/show_bug.cgi?id=699337

2012-07-05 18:02:27 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: handle return value -ENOTTY for unimplemented VIDIOC_G_PARM
	  Newer kernels return -ENOTTY, older kernels return -EINVAL if the ioctl
	  is not implemented. With this patch, GStreamer handles both cases.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698825

2013-04-30 09:16:07 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix broken boolean expression to detect non-frame buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=699294

2013-04-29 11:07:56 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Better error message when server version is too old
	  We check for the library version at configure time, but the server
	  version can only really be checked at run-time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698768

2013-04-27 11:24:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/udp/gstudp.c:
	  udp: log WARNING debug message if UDP multicast is likely to be broken

2013-04-27 11:16:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/udp/gstudpsrc.c:
	  udpsrc: add includes to get socklen_t defined on Windows
	  https://bugzilla.gnome.org/show_bug.cgi?id=692400

2013-04-27 09:39:45 +0100  Yury Delendik <async.processingjs@yahoo.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for VP6F VP6 flash codec
	  https://bugzilla.gnome.org/show_bug.cgi?id=699010

2012-09-05 16:39:31 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: also poll for output devices
	  Note that the V4L2 API defines that for output devices POLLOUT
	  indicates that a buffer is ready to be dequeued.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698992

2012-08-20 09:52:34 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix copying of encoded buffers
	  The existence of a GstVideoFormatInfo does not guarantee, that
	  the buffer contains video frames, so the format must be checked.
	  Also, for encoded buffers the length is variable and must be set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698949

2012-07-10 15:29:40 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: add support for mpeg4 and H.263
	  https://bugzilla.gnome.org/show_bug.cgi?id=698826

2013-04-26 12:16:49 +0200  Edward Hervey <edward@collabora.com>

	* gst/monoscope/gstmonoscope.c:
	  monoscope: Fix debug statement

2013-04-25 21:50:33 +0200  Alexander Schrab <meros@meros-desktop.(none)>

	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-decode.h:
	* tests/check/Makefile.am:
	* tests/check/elements/mulawdec.c:
	  mulawdec: change base class to GstAudioDecoder
	  https://bugzilla.gnome.org/show_bug.cgi?id=698894

2013-04-25 20:59:52 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: send stream-start event.

2012-10-18 10:37:35 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: handle ENODATA return value for VIDIOC_ENUMSTD
	  In kernel v3.7-rc1, VIDIOC_ENUMSTD returns ENODATA if the current input
	  does not support the STD API.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698827

2013-04-25 13:19:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	  docs: add some pay/depayloaders
	  See https://bugzilla.gnome.org/show_bug.cgi?id=551631

2013-04-25 12:44:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/law/mulaw-encode.c:
	* tests/check/elements/mulawenc.c:
	  mulaw: Some minor memleak fixes and cleanup

2013-04-24 13:56:56 +0200  Alexander Schrab <alexas@axis.com>

	* gst/law/mulaw-encode.c:
	* gst/law/mulaw-encode.h:
	* tests/check/Makefile.am:
	* tests/check/elements/mulawenc.c:
	  mulawenc: change to gstaudioencoder base, added bitrate tags

2012-05-03 16:07:27 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: bufferpool: reset buffer size in release_buffer
	  The buffer might still be in use elsewhere when dequeuing buffers for
	  outputs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698822

2012-04-20 09:53:35 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: bufferpool: remove unused includes
	  The hacks that needed these are long gone.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698821

2013-04-25 12:12:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  (multi)udpsink: Use separate sockets for IPv4 and IPv6
	  https://bugzilla.gnome.org/show_bug.cgi?id=534243

2013-04-25 10:44:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstdynudpsink.h:
	  dynudpsink: Use separate sockets for IPv4 and IPv6
	  https://bugzilla.gnome.org/show_bug.cgi?id=534243

2013-04-25 10:43:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/Makefile.am:
	  udp: Don't include removed gstudp.h in noinst_HEADERS

2013-04-17 16:47:31 -0700  Todd Agulnick <todd@agulnick.com>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudio: Use gst_audio_channel_positions_to_mask() to create mask
	  https://bugzilla.gnome.org/show_bug.cgi?id=698807

2013-04-17 16:12:26 -0700  Todd Agulnick <todd@agulnick.com>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudio: Remove unused code

2013-04-25 09:16:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/Makefile.am:
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudp.h:
	* gst/udp/gstudpsink.h:
	* gst/udp/gstudpsrc.h:
	  udp: Remove unused enum type

2013-04-25 09:13:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/Makefile.am:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudp-marshal.list:
	  udp: Use the generic marshaller instead of generating marshallers

2013-04-25 09:07:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: Rename instance variable from host to multi_group
	  This is more consistent as it's used for the multicast-group property.

2013-04-25 09:03:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Add bind-address property
	  This is equivalent to multicast-group currently for backwards compatibility.
	  In 2.0 this should be handled separately, the former only being the multicast
	  group and the latter always being the address the socket is bound to, even if
	  a multicast group is given.

2013-04-24 16:24:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  vrawdepay: return output buffer from process
	  Return the output buffer from the process function instead of pushing
	  it ourselves. This way, the subclass can actually deal with the return
	  value of the push.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693727

2012-10-01 09:29:21 -0300  Diogo Carbonera Luvizon <diogo.luvizon@ensitec.com.br>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: save the format correctly
	  If TRY_FMT is not implemented,  gst_v4l2_object_get_nearest_size will
	  use S_FMT and will change the device's operation mode. To save the
	  old device mode we need to set the type field or else it will fail
	  to save the previous format.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685209

2013-04-24 15:38:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	  rtp: a marker bit should translate to RESYNC
	  A marker bit on an audio packet does not mean a DISCONT (in the GStreamer sense
	  of missing data) but it means that the packet is the end of a talkspurt and thus
	  a good opportunity to resync to the clock. Use the RESYNC buffer flag to note
	  this.
	  Real discontinuities are marked with DISCONT still when the seqnum has a GAP or
	  when the input buffer has the DISCONT flag set.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=627204

2013-04-22 23:51:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* MAINTAINERS:
	* README:
	* README.static-linking:
	* common:
	  Automatic update of common submodule
	  From 3cb3d3c to 5edcd85

2013-04-22 10:19:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	  rtpjpegdepay: Drop frame if it's less than 2 bytes large
	  https://bugzilla.gnome.org/show_bug.cgi?id=677560

2013-04-18 12:20:08 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	  autodetect: use _plugin_feature_rank_compare API instead of duplicating the code.

2013-04-18 09:37:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/osxaudio/gstosxaudioringbuffer.h:
	  osxaudio: Include gstaudioringbuffer.h to fix compilation in 1.0

2013-04-17 21:05:14 +0200  Philippe Normand <philn@igalia.com>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: channel-mask configuration fixes
	  Set channel-mask according to sink's layout in case of stereo layout.
	  Also initialize and reset the mask when an unrecognized channel is detected.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698224

2013-04-15 19:53:28 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Disable renegotiation in the negotiate method
	  This way, we don't block the initial negotiation.
	  Thanks to Jeremy Whiting for doing all the testing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=695981

2013-04-15 19:46:12 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  Revert "v4l2: disable renegotiation"
	  This reverts commit d1b26e1d594ab2b63324e43a36330475e98cdf18.
	  This causes the initial negotiation to never happen if a reconfigure
	  event is received after gst_base_src_start_complete() but before the loop
	  starts.
	  https://bugzilla.gnome.org/show_bug.cgi?id=695981

2013-04-17 21:12:55 +0200  Stefan Sauer <ensonic@users.sf.net>

	* ext/flac/gstflactag.c:
	  flactag: forward caps event
	  This ensures that the downstream element will get the event and negotiates. Add
	  a FIXME for updating the streamheader field on th caps.

2013-04-17 07:50:27 +0200  Stefan Sauer <ensonic@users.sf.net>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	  flac: add more logging

2013-04-17 20:24:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/osxaudio/gstosxcoreaudiocommon.h:
	  osxaudio: Fix merge conflicts

2013-04-17 10:10:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  osxaudio: Fix configure check for osxaudio plugin

2013-04-17 09:50:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	  osxaudioringbuffer: First check the type, then cast

2013-04-16 22:46:00 +0900  Takashi Nakajima <ted.nakajima@gmail.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxaudiosink.h:
	  osxaudio: use GST_IS_OSX_AUDIO_SINK in ring buffer.

2013-04-10 21:06:16 +0900  Takashi Nakajima <ted.nakajima@gmail.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	  osxaudio: call set_channel_positions() in osxaudioringbuffer acquire()

2013-04-12 12:18:04 -0700  Todd Agulnick <todd@agulnick.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	  osxaudio: use GST_AUDIO_INFO_* accessors
	  Changes include the following:
	  * Update classname references
	  * Replace GST_BOILERPLATE_FULL with G_DEFINE_TYPE
	  * Use new GstAudioInfo struct and methods
	  * Use new buffer memory allocation scheme
	  Conflicts:
	  sys/osxaudio/gstosxaudioringbuffer.c

2013-04-12 11:51:46 -0700  Todd Agulnick <todd@agulnick.com>

	* sys/osxaudio/gstosxcoreaudiocommon.h:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: adjust for changes to glib mutex api.

2013-04-10 01:21:49 +0900  Takashi Nakajima <ted.nakajima@gmail.com>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudio: try to fix up according to Sebastian's comments

2013-04-05 10:02:38 +0200  Philippe Normand <philn@igalia.com>

	* configure.ac:
	* sys/osxaudio/gstosxaudioringbuffer.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.h:
	  osxaudio: build fixes
	  Enable the osxaudio plugin build in configure.ac and fix some
	  include directive order issues.

2013-04-02 22:28:09 +0900  ted-n <ted.nakajima@gmail.com>

	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudio: fix layout for osxaudiosrc

2013-03-30 22:49:34 +0900  ted-n <ted.nakajima@gmail.com>

	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudioelement.c:
	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxaudioringbuffer.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxaudiosrc.h:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiocommon.h:
	  osxaudio: port to v.1.0

2013-04-16 19:29:48 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't unref query, we don't own it
	  Fixes double-unref bug. Bug found by Youness Alaoui

2013-04-16 20:41:10 +0200  Philippe Normand <philn@igalia.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix SCHEDULING query support
	  Chain the query up to parent before adding _BANDWIDTH_LIMITED flag,
	  so that all the other flags get set, and push mode gets added as
	  supported activation mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=693484
	  https://bugzilla.gnome.org/show_bug.cgi?id=698156

2013-03-31 12:05:49 +0200  Philippe Normand <philn@igalia.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: basic scheduling query support
	  Answer to scheduling queries with default parameters and the new
	  _BANDWIDTH_LIMITED_FLAG so that downstream is advised to minimize seek
	  operations and perform on-disk buffering if possible.
	  Bug 693484

2013-04-15 14:32:46 +0000  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix segfault accessing osxwindow when not set yet

2012-10-24 12:15:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/twolame/Makefile.am:
	  gst: Add better support for static plugins

2012-10-24 12:15:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/Makefile.am:
	  gst: Add better support for static plugins

2012-10-24 12:14:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/aalib/Makefile.am:
	* ext/cairo/Makefile.am:
	* ext/dv/Makefile.am:
	* ext/flac/Makefile.am:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/jack/Makefile.am:
	* ext/jpeg/Makefile.am:
	* ext/libcaca/Makefile.am:
	* ext/libpng/Makefile.am:
	* ext/mikmod/Makefile.am:
	* ext/pulse/Makefile.am:
	* ext/raw1394/Makefile.am:
	* ext/shout2/Makefile.am:
	* ext/soup/Makefile.am:
	* ext/speex/Makefile.am:
	* ext/taglib/Makefile.am:
	* ext/vpx/Makefile.am:
	* ext/wavpack/Makefile.am:
	* gst/alpha/Makefile.am:
	* gst/apetag/Makefile.am:
	* gst/audiofx/Makefile.am:
	* gst/audioparsers/Makefile.am:
	* gst/auparse/Makefile.am:
	* gst/autodetect/Makefile.am:
	* gst/avi/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debugutils/Makefile.am:
	* gst/deinterlace/Makefile.am:
	* gst/dtmf/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/equalizer/Makefile.am:
	* gst/flv/Makefile.am:
	* gst/flx/Makefile.am:
	* gst/goom/Makefile.am:
	* gst/goom2k1/Makefile.am:
	* gst/icydemux/Makefile.am:
	* gst/id3demux/Makefile.am:
	* gst/imagefreeze/Makefile.am:
	* gst/interleave/Makefile.am:
	* gst/isomp4/Makefile.am:
	* gst/law/Makefile.am:
	* gst/level/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/monoscope/Makefile.am:
	* gst/multifile/Makefile.am:
	* gst/multipart/Makefile.am:
	* gst/replaygain/Makefile.am:
	* gst/rtp/Makefile.am:
	* gst/rtpmanager/Makefile.am:
	* gst/rtsp/Makefile.am:
	* gst/shapewipe/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/spectrum/Makefile.am:
	* gst/udp/Makefile.am:
	* gst/videobox/Makefile.am:
	* gst/videocrop/Makefile.am:
	* gst/videofilter/Makefile.am:
	* gst/videomixer/Makefile.am:
	* gst/wavenc/Makefile.am:
	* gst/wavparse/Makefile.am:
	* gst/y4m/Makefile.am:
	* sys/directsound/Makefile.am:
	* sys/oss/Makefile.am:
	* sys/oss4/Makefile.am:
	* sys/osxaudio/Makefile.am:
	* sys/osxvideo/Makefile.am:
	* sys/sunaudio/Makefile.am:
	* sys/v4l2/Makefile.am:
	* sys/waveform/Makefile.am:
	* sys/ximage/Makefile.am:
	  gst: Add better support for static plugins

2013-04-12 19:26:11 +0000  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/goom2k1/Makefile.am:
	  goom2k1: fix duplicated symbol with goom

2013-03-10 17:17:17 +0000  Josep Torra <n770galaxy@gmail.com>

	* sys/osxaudio/gstosxaudioelement.c:
	* sys/osxaudio/gstosxcoreaudiocommon.h:
	  osxaudio: Fixes error: "GST_LEVEL_DEFAULT" redefined

2013-03-10 17:27:30 +0000  Josep Torra <n770galaxy@gmail.com>

	* sys/osxaudio/gstosxcoreaudiohal.c:
	  osxaudio: fixes implicit declaration of function 'getpid'

2013-04-14 17:55:02 +0100  Tim-Philipp Müller <tim@centricular.net>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From aed87ae to 3cb3d3c

2013-04-14 12:32:06 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: add back "iradio-mode" property to disable sending of icecast request headers
	  In 1.0 we now always send the icecast request headers by default, which
	  makes the server send icecasts metadata inserted into the stream if it
	  supports that. However, there are some use cases where this is not
	  desirable, like when just saving a radio stream to disk, so add back
	  the "iradio-mode" property to allow people to disable this.
	  https://bugzilla.gnome.org/show_bug.cgi?id=697984

2013-04-12 16:16:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtp.c:
	  rtp: register tag image types
	  The rtpgstdepay needs the type to be available in order to deserialize the
	  event.

2013-04-12 16:08:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: handle event parse failures better

2013-04-11 22:25:05 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: add TOC setter support

2013-04-12 12:31:30 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/wavenc/gstwavenc.c:
	  wavenc: small cleanups for toc handling
	  Don't add empty labl/note chunks. Always pass instance as the first param. Add more logging.

2013-04-12 12:58:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Proxy the ntp-sync property of rtpbin

2013-04-12 12:51:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Give the manager always the name "manager"
	  This allows to use the GstChildProxy interface to adjust
	  properties on it.

2013-04-11 22:53:28 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/alphacolor.c:
	* tests/check/elements/apev2mux.c:
	* tests/check/elements/id3v2mux.c:
	* tests/check/pipelines/flacdec.c:
	  tests: fix some printf format issues in debug messages

2013-04-11 19:27:15 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	  wavenc: add 'note' chunk support

2013-04-11 20:46:26 +0200  Stefan Sauer <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: add a little more docs to the audioclock

2013-04-11 15:00:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add support for NetClientClock
	  When the server suggests a GstNetTimeProvider in the SDP, set up a
	  GstNetClientClock that slaves to the remote clock and suggest this clock in
	  provide_clock.

2013-04-11 14:57:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  udpsink: avoid alloc and free in render function
	  Avoid doing alloc and free in the render function for each buffer. Instead,
	  allocate the needed arrays in _init and use those.

2013-04-10 08:36:00 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  waveparse: remove superfluous g_list_first() calls
	  The variables already point to the start of the list.

2013-04-09 23:13:18 +0100  Andreas Fenkart <andreas.fenkart@streamunlimited.com>

	* gst/rtp/gstrtpsbcdepay.c:
	  rtpsbcdepay: fix sbc frame length calculation for mono and stereo modes
	  https://bugzilla.gnome.org/show_bug.cgi?id=697463

2013-03-25 14:35:02 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: add 'note' chunk support
	  Add 'note' chunk support in TOC as GST_TAG_COMMENT
	  https://bugzilla.gnome.org/show_bug.cgi?id=696549

2013-04-08 17:53:09 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check value inside enda to set endianness

2013-04-09 21:00:12 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 04c7a1e to aed87ae

2013-04-09 17:34:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/icydemux/gsticydemux.c:
	  icydemux: avoid copy when we can

2013-04-09 16:52:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  gstpay: use bufferlist to avoid memcpy

2013-04-09 16:50:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udpsink: improve debug

2013-04-09 00:28:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/wavparse.c:
	  tests: refactor new wavparse test a little
	  Use fakesrc instead of filesrc with /dev/null.
	  https://bugzilla.gnome.org/show_bug.cgi?id=696684

2013-04-08 11:38:33 +0200  Alexander Schrab <alexas@axis.com>

	* gst/wavparse/gstwavparse.c:
	* tests/check/Makefile.am:
	* tests/check/elements/wavparse.c:
	  wavparse: error out if we receive eos before any valid data
	  https://bugzilla.gnome.org/show_bug.cgi?id=696684

2013-04-07 01:47:56 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: force deinterlacing in "interlaced" mode
	  https://bugzilla.gnome.org/show_bug.cgi?id=697467

2013-04-06 12:45:28 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	  gdkpixbufsink: Add timestamp/running-time/stream-time to the emited message

2013-04-05 14:38:43 +0200  Nicola Murino <nicola.murino@gmail.com>

	* gst/rtp/gstrtpsbcdepay.c:
	  rtpsbcdepay: fix printf format compiler warnings
	  https://bugzilla.gnome.org/show_bug.cgi?id=697343

2013-04-05 09:34:23 +0100  Todd Agulnick <todd@agulnick.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideo: include pthread.h to fix compiler warning
	  https://bugzilla.gnome.org/show_bug.cgi?id=697303

2013-04-04 22:48:45 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	  level: resync on discont
	  Drop pending data on discont and start a new cycle with a new base timestamp.
	  Cleanup some variables.

2013-04-03 23:52:47 +0100  Tom Greenwood <tgreenwood@Toms-MacBook-Pro.local>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Improve logging when vpx_codec_peek_stream_info fails
	  Decode failures and missing keyframes should get different debug
	  output.
	  https://bugzilla.gnome.org/show_bug.cgi?id=697232

2013-04-03 18:24:29 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpsbcdepay.c:
	  rtpsbcdepay: Rank as secondary
	  This way, it will be selected by decodebin
	  Bug reported by andreas.fenkart@streamunlimited.com
	  https://bugzilla.gnome.org/show_bug.cgi?id=697227

2013-04-03 19:05:38 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	* tests/check/elements/level.c:
	  level: subdivide buffers for sample accurate interval handling
	  Previously we would skip level message when processing buffers > the requested
	  interval. Also the message frequency would contain quite some jitter due to only
	  considering them at the end of buffers.
	  Cleanup the tests while we're at it.

2013-03-19 08:23:25 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/flac/gstflacenc.c:
	  flacenc: remove old since comments and update logging
	  Don't pretend that we have a timestamp on a buffer when we never set one.

2013-03-18 20:59:23 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: remove old since comment

2013-04-03 17:53:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Proxy the multicast-iface property of udpsrc

2013-04-03 11:09:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: free all queued buffers
	  Don't just loop over the first num_queued buffers but loop over
	  all the buffers and check if they need to be freed. It is possible that
	  not all buffers are queued and then the entry in our array will be NULL.
	  Those buffers that are not queued were freed in stop().
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=696651

2013-04-03 11:09:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: improve debug

2013-04-02 23:42:23 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Only forward stick events while holding the sinkpad stream lock
	  Otherwise we get a race where if the RTCP packet comes in first and while
	  it is added the pads, the segment event arrives on the RTP stream, the event
	  may be lost completely and never forwarded.

2013-04-02 23:35:06 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: No need to explicitely forward the caps
	  They are forwarded with the other events

2013-04-02 22:29:38 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	  rtpssrcdemux: Remove unused GstSegment

2013-04-02 22:26:02 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Simplify event forwarding
	  Use the gst_pad_forward() mechanic, this way we won't miss pads that are
	  added while we are pushing

2013-04-02 21:53:10 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Don't cross the internal links
	  We had the wrong condition to check for the internal links, so RTP and RTCP
	  pads got crossed!

2013-03-31 17:54:16 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix some debug messages

2013-04-02 23:36:22 +0100  Tim-Philipp Müller <tim@centricular.net>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: fix printf format compiler warning in debug message

2012-08-29 17:24:00 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: handle TrueHD audio codec id
	  https://bugzilla.gnome.org/show_bug.cgi?id=697113

2013-03-31 19:14:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	  theorapay: add delta-unit to output frames

2013-03-23 05:22:23 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: use timestamp delta as duration if possible
	  https://bugzilla.gnome.org/show_bug.cgi?id=696437

2013-03-30 09:44:41 +0100  Josep Torra <n770galaxy@gmail.com>

	* gst/rtp/gstrtpsbcdepay.c:
	  rtp: fixes debug message printf related compiler warnings in SBC depayloader

2013-03-28 16:46:36 +0000  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsbcdepay.h:
	  rtp: Add an rtpsbcdepay element
	  Pretty straightforward - takes SBC encapsulated in RTP, depayloads, and
	  pushes out SBC buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690582

2013-03-27 22:18:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/gstrtpsbcpay.c:
	  rtp: fix SBC payloader
	  Init RTP buffer on stack correctly, so mapping it works
	  without criticals and the payloader actually works.

2013-03-26 14:44:36 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Check for a subset instead of non-empty intersection in accept-caps

2013-03-26 14:39:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Properly handle the filter caps in get_caps()

2013-03-26 14:35:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Don't unnecessarily get the parent class in class_init
	  The trampoline generated by G_DEFINE_TYPE does that already.

2013-03-25 18:02:10 -0700  David Schleef <ds@schleef.org>

	* gst/avi/gstavidemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	  Use %03u for format in gst_pad_create_stream_id_printf()

2013-03-25 10:12:03 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/debugutils/gstcapssetter.c:
	  capssetter: Prevent unneeded caps copying and allocation

2013-02-01 14:33:41 +0100  Dirk Van Haerenborgh <vhdirk@gmail.com>

	* gst/debugutils/gstcapssetter.c:
	  capssetter: Pass any or filter caps upstream
	  capsetter accepts anything and just forwards different caps,
	  as such it should return ANY caps on the sinkpad.
	  https://bugzilla.gnome.org/show_bug.cgi?id=693005

2013-03-06 13:17:54 +0000  Tom Greenwood <tgreenwood@Toms-MacBook-Pro.local>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Fix for divide by zero when using 0/1 framerate
	  https://bugzilla.gnome.org/show_bug.cgi?id=695709

2013-03-24 17:55:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/wavparse/gstwavparse.c:
	  wavparse: expose CUE sheet items as tracks not chapter entries in TOC
	  https://bugzilla.gnome.org/show_bug.cgi?id=677306

2013-03-23 13:11:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ext/flac/gstflacenc.c:
	  flacenc: add more example pipelines

2013-03-23 12:59:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/wavenc/gstwavenc.c:
	  wavenc: add some example pipelines

2013-03-20 21:38:40 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	  wavenc: add TOC support
	  https://bugzilla.gnome.org/show_bug.cgi?id=680998

2013-03-23 04:56:36 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: make empty subtitle buffer recognition more robust
	  https://bugzilla.gnome.org/show_bug.cgi?id=696244

2013-03-04 15:49:06 -0800  David Schleef <ds@schleef.org>

	* ext/libpng/gstpngenc.c:
	  pngenc: unmap source frame when done

2013-03-22 15:14:15 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix test regression with one buffer streams

2013-03-05 17:00:17 -0800  David Schleef <ds@schleef.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: split large raw audio samples
	  In order to deal with a file that has samples that are 24 seconds
	  long.  Seeking still doesn't work with such files.

2013-03-22 11:54:08 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Remove documentation for dts-method

2013-03-22 13:24:33 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: deprecate dts-method property

2013-03-13 17:08:03 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix problems causing bad durations in file
	  - Fix up out-of-order incoming DTS values.
	  - Fix duration of initial sample.

2013-03-12 19:08:26 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/gstqtmux.c:
	  qtmux: fix all timestamps once first_ts is determined

2013-02-14 16:34:34 -0800  David Schleef <ds@schleef.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Use PTS/DTS from incoming buffers
	  Remove old DTS guessing code.

2013-03-18 12:30:50 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: expose mulaw caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=696052

2013-03-22 10:50:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  Require Orc >= 0.4.17
	  Orc 0.4.17 fixes a bunch crashes on i386 and RPi when orc
	  functions can't be compiled and the fallback function is
	  supposed to be used. Also fixes some issues on PowerPC.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684399
	  https://bugzilla.gnome.org/show_bug.cgi?id=693862

2013-03-22 08:47:17 +0000  Rodolfo Schulz de Lima <rodolfo@rodsoft.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix sample leak when processing private qt tags
	  https://bugzilla.gnome.org/show_bug.cgi?id=696355

2013-03-22 02:24:01 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: set stream language code from tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=696358

2013-03-21 02:55:06 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: send GAP events for subtitle streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=696244

2013-03-21 02:53:24 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: ignore empty subtitle buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=696244

2013-03-21 02:52:07 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: recognize SBTL subtype for subtitles
	  https://bugzilla.gnome.org/show_bug.cgi?id=696244

2013-03-17 16:27:03 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: add support for the toc-select event
	  Select tracks from the CUE sheet by sending a toc-select
	  event based on the uid in the TOC.
	  https://bugzilla.gnome.org/show_bug.cgi?id=540891

2013-03-19 18:09:31 -0700  Michael Smith <msmith@rdio.com>

	* gst/isomp4/gstqtmux.c:
	  mp4mux: in faststart mode, don't output up to 4 kB of garbage at the end.

2013-03-20 00:35:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/audioparsers/gstsbcparse.c:
	  sbcparse: pack multiple frames into one output buffer
	  Don't output a single buffer for every tiny SBC frame

2013-03-18 14:59:35 +0000  Bastien Nocera <hadess@hadess.net>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: fix compilation against newer kernel headers as on FC19

2013-03-14 14:12:05 +0100  Kishore Arepalli <kishore.arepalli@gmail.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix infinite loop on EOS with non-default methods or fields
	  Fixes problem of infinite loop in gst_deinterlace_reset_history.
	  Last field in the history was never deinterlaced because idx becomes negative.
	  Happens e.g. with method=scalerbob fields=bottom or
	  method=greedyl fields=top
	  https://bugzilla.gnome.org/show_bug.cgi?id=695644
	  https://bugzilla.gnome.org/show_bug.cgi?id=693173

2013-03-12 09:48:31 +0000  Kishore Arepalli <kishore.arepalli@gmail.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: don't return FALSE when dropping sink events
	  Fixes problem in conjunction with avidemux.
	  https://bugzilla.gnome.org/show_bug.cgi?id=695643

2013-03-12 00:16:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/avi/gstavimux.c:
	  avimux: change raw video caps order so that GRAY8 is last
	  People like colours.
	  https://bugzilla.gnome.org/show_bug.cgi?id=695543

2013-03-11 14:50:41 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Don't use upstream caps with peer_query_caps ()
	  Calling gst_pad_peer_query_caps () on the src pad with the caps
	  upstream can produce as a filter from gst_rtp_h264_pay_getcaps ()
	  is wrong and makes caps negotiation fail if upstream caps are not
	  NULL.
	  https://bugzilla.gnome.org/show_bug.cgi?id=695629

2013-03-10 09:10:18 +0100  Dirk Van Haerenborgh <vhdirk@gmail.com>

	* gst/avi/gstavimux.c:
	  avimux: support raw BGR
	  https://bugzilla.gnome.org/show_bug.cgi?id=695543

2013-03-10 09:25:34 +0100  Dirk Van Haerenborgh <vhdirk@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: support raw video with negative height
	  https://bugzilla.gnome.org/show_bug.cgi?id=695541

2013-03-05 14:40:56 +0100  Jonas Holmberg <jonashg@axis.com>

	* tests/check/elements/autodetect.c:
	  autodetect checktest: Do not fail without videosink
	  If there is no videosink available autovideosink will contain a
	  fakesink instead which needs special treatment in the unit test.

2013-03-09 01:18:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	* Android.mk:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* gst-plugins-good.spec.in:
	* gst/dtmf/gstdtmf.c:
	* gst/dtmf/gstdtmfcommon.h:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	  dtmf: move dtmf plugin from -bad to -good
	  https://bugzilla.gnome.org/show_bug.cgi?id=687416

2013-03-09 00:30:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Merge branch 'dtmf-moved-from-bad'
	  https://bugzilla.gnome.org/show_bug.cgi?id=687416

2013-03-05 21:22:18 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>

	* configure.ac:
	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudioelement.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: add support for iOS using the RemoteIO AudioUnit

2013-03-05 21:17:52 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiocommon.h:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	* sys/osxaudio/gstosxringbuffer.c:
	* sys/osxaudio/gstosxringbuffer.h:
	  osxaudio: add a façade for the CoreAudio API

2013-03-07 00:00:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	* common:
	  Automatic update of common submodule
	  From 2de221c to 04c7a1e

2013-03-03 11:59:31 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/lzo.c:
	  matroska: Include config.h, it's needed for _stdint.h

2013-03-03 11:53:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Fix (wrong) use of uninitialized variable compiler warning

2013-03-02 13:59:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add variant field to H.263 caps
	  avdec_h263 won't get plugged otherwise.

2013-02-22 19:06:52 +0100  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: skip disabled tracks
	  ISO/IEC 14496-12 specifies disabled tracks should be completely
	  ignored, so just do it.
	  Avoids deadlock during prerolling for some files.
	  Also prevents 'chapter' subtitle tracks from showing up.
	  https://bugzilla.gnome.org/show_bug.cgi?id=693993
	  https://bugzilla.gnome.org/show_bug.cgi?id=628790

2013-02-25 09:58:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/level.c:
	  tests: re-add suppression for GValueArray warnings to unit test as well

2013-02-28 13:25:06 +0100  Jonas Holmberg <jonashg@axis.com>

	* tests/check/elements/dtmf.c:
	  tests: use relative include for out-of-tree builds in dtmf test

2013-02-28 08:46:59 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: remove the since doc-comment from 0.10

2013-02-28 08:44:18 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	* tests/examples/level/level-example.c:
	  level: add a "post-messages" property and deprecate "message"
	  In spectrum this was changed from 0.10 to 1.0, lets do this here too.

2013-02-27 18:56:50 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/dtmf.c:
	  tests: Add tests for dtmfsrc

2013-02-27 16:15:27 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/dtmf.c:
	  tests: Fix ref leak in dtmf test

2013-02-26 14:18:20 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: streamtype is not put by all RTSP server, not make it optional
	  Specific case here is Wowza 3.5.0

2013-02-25 00:35:58 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* gst/level/gstlevel.c:
	  level: put back deprecation warnings

2013-02-24 17:00:14 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* gst/level/gstlevel.c:
	* tests/check/elements/level.c:
	  level: send last message on EOS

2013-02-23 14:34:35 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavidemux.c:
	  avidemux: push mode: handle some more 0-size buffer cases
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684944

2013-02-23 18:50:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix up example pipeline in docs

2012-11-20 12:14:07 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Update segdone periodically
	  This makes sure that we update segdone based on the read index received
	  during latency updates. As the comment notes, we make some compromises
	  to deal with the fact that segdone is a segment multiple, while the read
	  index offers finer granularity. The updates are also not very often
	  (100ms since that is how often automatic timing updates are provided).
	  All this is required for the baseaudiosink sample alignment code to work
	  at all.
	  https://bugzilla.gnome.org/show_bug.cgi?id=694257

2013-02-13 10:46:54 +0100  Paul HENRYS <visechelle@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Fix wrong code organisation in case of collision
	  change_ssrc field of RTPSession should be set before calling
	  rtp_session_schedule_bye_locked () as this function will call reconsider function
	  that will wake up rtcp_thread which will call rtp_session_on_timeout () that will
	  check change_ssrc to change the ssrc.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184

2013-02-21 11:15:23 -0500  Jean-François Fortin Tam <nekohayo@gmail.com>

	* gst/alpha/gstalpha.c:
	  alpha: improve descriptions of chroma keying-related properties and enums
	  https://bugzilla.gnome.org/show_bug.cgi?id=694374

2013-02-21 15:01:15 -0500  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Do not override the method with custom r/g/b values
	  Depending on the order g_object_set() calls aare made, the
	  target r/g/b settings will override the method if set to
	  green/blue. Change that so we do not use the target-r/g/b values
	  unless the method is set to custom.
	  https://bugzilla.gnome.org/show_bug.cgi?id=694374

2013-02-20 15:46:43 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst/auparse/gstauparse.c:
	  auparse: do not leak src_caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=694275

2013-02-20 21:03:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: only delay RTCP when we are a sender
	  Only delay the RTCP thread when we are a sender, which we can know because we
	  have a send_rtp_src pad. Otherwise we might delay the RTCP thread if we
	  are only a receiver and then there is no code path that wakes up the
	  RTCP thread and we end up without RTCP packets.

2013-02-19 11:47:20 +0100  Benjamin Gaignard <benjamin.gaignard@linaro.org>

	* configure.ac:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Add support of dmabuf
	  v4l has add a new IOCTL to export a buffer by using dmabuf.
	  This patch allow to use this new IOTCL if it has been defined in videodev2.h
	  I introduce a new IO mode (GST_V4L2_IO_DMABUF) to enable this way of working.
	  https://bugzilla.gnome.org/show_bug.cgi?id=693826

2013-02-18 20:04:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix up dodgy code that tries to fix up a broken moov atom
	  After gst_buffer_new_and_alloc() gst_buffer_copy_into() will likely
	  append to the already-existing memory instead of filling it.

2013-02-18 16:32:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix potential crash on short MOOV atom
	  Don't unmap short MOOV atom buffer twice, which happened
	  in the case where we don't fix up the MOOV atom.
	  Fixes crashes when thumbnailing partial mp4 file where
	  the MOOV atom is still incomplete.
	  https://bugzilla.gnome.org/show_bug.cgi?id=694010

2013-02-16 16:49:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ext/soup/Makefile.am:
	  souphttpsrc: set SOUP_VERSION_{MIN_REQUIRED,MAX_ALLOWED} to suppress deprecations with newer versions
	  https://bugzilla.gnome.org/show_bug.cgi?id=693911

2013-02-16 15:47:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	* ext/soup/gstsouphttpsrc.c:
	  soup: use default proxy resolver instead of deprecated GNOME proxy resolver
	  Apparently there's no reason to use it any longer. Drop libsoup-gnome
	  dependency while at it, now that we don't need anything from it any
	  more (it only consists entirely of deprecated API now anyways).
	  https://bugzilla.gnome.org/show_bug.cgi?id=693911

2013-02-15 15:43:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  tests: fix some h264 caps
	  Doesn't fix anything in particular, but is
	  still needed here for correctness.

2013-02-15 08:19:24 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: remove channel-mask from caps
	  The channel-mask is only needed for channels>2 which we don't do.

2013-02-15 16:21:21 +0100  Benjamin Gaignard <benjamin.gaignard@stericsson.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: don't check stride for encoded formats
	  Don't try to check the stride for encoded formats. Some drivers output
	  something != 0 and then we don't want to fail on that.

2013-02-15 14:11:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: use g_socket_set_option() to set buffer size with newer GLib versions
	  So we have to worry less about portability.
	  https://bugzilla.gnome.org/show_bug.cgi?id=692400

2013-02-14 14:13:27 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: remove sof-marker from template caps for now
	  Now that the subset check actually works, this breaks
	  things with demuxers that don't put a "sof-marker"
	  in their jpeg caps, and we don't have a good parser
	  to plug either yet.

2013-02-13 12:32:10 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpegenc: Put the SOF marker into the caps

2013-02-13 12:02:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpamrdepay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtp-payloading: Fix unit test caps and AMR depayloader sink template caps
	  Fields were missing from the actual caps, or too many fields
	  existed in the template caps.

2013-02-13 11:53:01 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/aacparse.c:
	  aacparse: Fix caps used in the unit test
	  The AAC caps passed were incomplete.

2013-02-13 11:49:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/wavpackenc.c:
	* tests/check/elements/wavpackparse.c:
	  wavpack: Fix unit tests, width is now called depth in the caps in 1.0

2013-02-12 23:31:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/souphttpsrc.c:
	  tests: make souphttpsrc unit test work even if http_proxy is set
	  We're testing with an http server on localhost, but don't support
	  an exception list for the http_proxy, so just unset the environment
	  variable to make sure we can run this test properly even if the
	  environment has http_proxy set.
	  Also, don't skip all tests if there is an issue with the SSL server,
	  just run the non-SSL tests then.
	  https://jenkins.qa.ubuntu.com/view/Raring/view/JHBuild%20Gnome/job/jhbuild-amd64-gst-plugins-good/

2013-02-12 12:53:52 -0800  Michael Smith <msmith@rdio.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: extract codec_data for ProRes

2013-02-08 01:02:10 +1100  Tim 'mithro' Ansell <mithro@mithis.com>

	* gst/avi/gstavimux.c:
	  avimux: Fixing buffer leak in gst_avi_mux_do_buffer
	  gst_avi_mux_do_buffer was leaking data from gst_collect_pads_pop.

2013-02-10 15:10:32 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavidemux.c:
	  avidemux: correct duration for audio VBR buffers in pull mode

2013-02-08 21:28:02 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/avi/gstavidemux.c:
	  avidemux: proper position reporting and push mode timestamping
	  ... and align current_total semantics in push and pull mode,
	  which tracks bytes for CBR and blocks for VBR.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691481

2013-02-08 17:05:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: delay RTCP until first RTP packet
	  Delay sending the first RTCP packet until we have sent the first RTP packet.
	  Otherwise we will send out a Receiver Report instead of a sender report.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=691400

2013-02-07 15:06:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: remove dead code
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355

2013-01-29 10:48:17 +0100  Paul HENRYS <visechelle@gmail.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: forward sticky events and then set caps
	  When a new src pad is added, first forward the sticky events and then
	  set the caps on the src pad
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786

2013-02-07 14:32:26 +0100  Markovtsev Vadim <v.markovtsev at samsung.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: improve debug output
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688935

2011-09-26 14:42:51 -0700  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: rework cleanup of streams
	  Move the work of cleaning up the client streams in the free_stream
	  function. This allows us to properly clean up the client streams when we
	  remove an RTP stream as well.
	  Based on patch by Sujay <sdatar@cisco.com>
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156

2013-02-07 11:40:35 +0100  Tim 'mithro' Ansell <gnome at mithis.com>

	* gst/videomixer/videomixer2.c:
	  videomixer2: avoid caps leak
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693307

2013-02-06 17:15:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: do skew estimation only for new timestamps
	  Only run the skew estimation code when we have a new RTP timestamp. If we have
	  the same RTP timestamp, we simply use the previous estimation. This works
	  because the new observation with the same RTP timestamp has to have a bigger
	  receiver time and is thus not going to influence the estimation except for
	  causing more jitter.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023

2013-02-06 13:52:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: only EOS when our source sends BYE
	  Only EOS when we receive a BYE event from the SSRC of our stream.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453

2013-02-06 13:47:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: save the stream SSRC
	  Conflicts:
	  gst/rtsp/gstrtspsrc.c

2013-02-06 13:18:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: flush connection when stopping
	  When we stop, we can flush all pending commands so that we can stop and
	  join the task.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924

2013-02-05 22:02:13 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/spectrum/README:
	  spectrum: remove outdates readme
	  Lets remove the readme from pre-0.1.0 that is completely irrelevant now.

2013-02-05 07:32:29 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: add more debug logging

2013-02-05 08:26:14 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/level/level-example.c:
	  level-example. avoid taking the arrays again for each channel for clarity
	  Also introduce some blank lines for better readability and update the comments.

2013-02-04 18:38:41 +0000  Rico Tzschichholz <ricotz@ubuntu.com>

	* gst/audioparsers/Makefile.am:
	  audioparsers: fix typo in noinst_headers

2013-02-04 11:08:23 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: further port to 1.0
	  Transformcaps is not called with caps containing single structures anymore. Also add missing filter handling. Still does not negotiate though.

2013-02-03 22:45:52 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: fix caps
	  We don't turn float into 32bit pcm. Looks like a typo from updating the caps.

2013-02-03 13:14:50 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/level/gstlevel.c:
	  level: Add missing coma between formats

2013-01-31 22:55:18 +1100  Matthew Waters <ystreet00@gmail.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: fix eos timestamp check
	  fixes hang in videotestsrc num-buffers=20 ! videomixer ! fakesink
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692935

2013-01-31 11:35:09 +0100  Dirk Van Haerenborgh <vhdirk@gmail.com>

	* gst/avi/gstavimux.c:
	  avimux: add support for raw monochrome 8-bit video
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692932

2013-01-18 21:08:12 +0400  Alexey Chernov <achernov@neosphere.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	  osxvideosink: Make GstNavigation key input events in osxvideosink compatible with x(v)imagesink ones

2013-01-29 10:30:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: avoid '...is used uninitialized'

2013-01-09 13:24:49 -0500  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: set interleaved layout correctly for LPCM audio
	  https://bugzilla.gnome.org/show_bug.cgi?id=663458

2013-01-08 20:45:21 -0500  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for LPCM fourcc (uncompressed audio in Quicktime7)
	  https://bugzilla.gnome.org/show_bug.cgi?id=663458

2013-01-08 20:42:35 -0500  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: print all debug for sound sample description v2
	  https://bugzilla.gnome.org/show_bug.cgi?id=663458

2013-01-08 20:14:17 -0500  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: sound sample description v2 doesn't override samples_per_packet
	  https://bugzilla.gnome.org/show_bug.cgi?id=663458

2013-01-08 19:57:50 -0500  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: pass stsd data to qtdemux_audio_caps()
	  We will need that later for LPCM format support. Disable
	  QDM2 parsing of stsd data which dead code before as well
	  because data was always NULL.
	  https://bugzilla.gnome.org/show_bug.cgi?id=663458

2013-01-08 19:56:46 -0500  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add len check for sound sample descriptions v1 and v2
	  https://bugzilla.gnome.org/show_bug.cgi?id=663458

2013-01-28 22:42:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpmanager: use C89-style comments

2013-01-28 18:06:15 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: Fix double-declared variable

2013-01-28 17:58:20 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtp: Fix compilation errors in previous patches

2011-04-28 22:59:28 +0200  Haakon Sporsheim <haakon.sporsheim@gmail.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Ensure MT safe event handling and plug event leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667826

2011-10-17 23:45:37 +0200  Idar Tollefsen <itollefs@cisco.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: mt-safe event-push
	  By taking a ref of the sink-pad under lock, it won't dissappear
	  while the push is taking place
	  https://bugzilla.gnome.org/show_bug.cgi?id=667816

2012-01-04 10:29:45 +0100  Pascal Buhler <pabuhler@cisco.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Safely push on pads that might be removed due to a RTCP BYE
	  https://bugzilla.gnome.org/show_bug.cgi?id=667815

2013-01-28 20:42:26 +0100  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From a942293 to 2de221c

2013-01-28 11:54:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstsbcparse.c:
	  sbcparse: init some variables to avoid bogus compiler warnings

2013-01-28 12:41:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	  rtpdepay: remove payload type restrictions
	  Remove the pt restrictions for all the depayloaders that have an
	  encoding-name. We can use this to autoplug decoders.
	  Remove the encoding-name for all the payloaders with a fixed payload
	  type.
	  We now either have an encoding-name or a pt in the sinkpad caps of
	  a depayloader.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=639292

2013-01-28 12:23:41 +0100  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	  rtp: remove payload requirements from selected depayloaders
	  encoding name is required in the caps and is a better fit for autoplugging than
	  the pt value. Hardware manufacturers have a bad habit of skimming through RFCs
	  and in this case; use unassigned numbers for encoders instead of dynamic
	  numbers.
	  In essence, this patch will add support for a lot of Bosch hardware encoders
	  without breaking autoplugging.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292

2013-01-27 10:17:59 +0530  B.Prathibha <bosslinux@cdac.in>

	* tests/examples/jack/jack_client.c:
	* tests/examples/rtp/server-alsasrc-PCMA.c:
	* tests/icles/ximagesrc-test.c:
	  tests: use g_timeout_add_seconds instead of g_timeout_add
	  https://bugzilla.gnome.org/show_bug.cgi?id=692615

2013-01-27 12:54:15 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: push mode: only parse moov 1 once
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691570

2013-01-26 22:58:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: fix compiler warning
	  gstrtpdtmfsrc.c: In function 'gst_dtmf_src_prepare_message.isra.1':
	  gstrtpdtmfsrc.c:669:3: error: 's' may be used uninitialized in this function

2013-01-25 21:06:05 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  rtpdtmfdepay: Fix missing work in doc

2013-01-24 21:00:08 -0500  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/dtmf.c:
	  tests: Add test for rtpdtmfdepay and rtpdtmfsrc

2013-01-25 20:39:33 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: Post the messages after the clock wait
	  This way, the messages will be closer in time to when the packets are sent out

2013-01-25 20:37:53 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: Only set the duration when starting to send
	  The duration depends on the clock rate, which could change due to renegotiation

2013-01-25 20:37:09 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: remove "ssrc" from caps
	  ssrc is uint and we don't have a uint range type

2013-01-24 21:08:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/isomp4/atoms.h:
	  qtmux: set language to 'undefined' instead of English by default

2013-01-23 21:35:25 -0500  Olivier Crête <olivier.crete@collabora.com>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/ximageutil.c:
	* sys/ximage/ximageutil.h:
	  ximagesrc: Set the pixel aspect ratio correctly in the caps

2013-01-08 08:56:45 +0100  Sjoerd Simons <sjoerd@luon.net>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: Re-enable prepare-format emission
	  With the port to gstreamer 1.0 the prepare-format signal stopped being
	  emitted. Start emitting this again for use in uvch264src.  While there
	  change the emission to include the caps for extra flexibility instead of
	  fource, width, height.
	  https://bugzilla.gnome.org/show_bug.cgi?id=692042

2013-01-22 18:12:10 +0100  Benjamin Gaignard <benjamin.gaignard@st.com>

	* autogen.sh:
	  autogen.sh: allow calling from out-of-tree
	  Signed-off-by: Benjamin Gaignard <benjamin.gaignard@st.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=692309

2013-01-22 19:26:09 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/audioparsers/gstsbcparse.c:
	  audioparsers: sbc: fix bogus compiler warning
	  gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame':
	  gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i

2013-01-19 13:27:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: don't error out if pa_stream_proplist_update() with new tags fails
	  Shouldn't really happen these days, but if it does, it's not really
	  a problem either.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656068

2013-01-16 18:01:23 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/souphttpsrc.c:
	  tests: skip souphttpsrc tests if there is no local http server to use
	  Skip tests if the server couldn't be started or we can't connect
	  to it for some reason (e.g. draconic build bot environments).

2013-01-16 14:32:56 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/audioparsers/gstsbcparse.c:
	  autoparsers: use appropriate printf format for gsize

2013-01-15 15:05:43 +0100  Martin Pitt <martinpitt@gnome.org>

	* tests/check/Makefile.am:
	  tests: use _1_0 variants for the various registry variables
	  These override the variants without version suffix. Makes 'make check' work
	  properly in environments that set the suffixed variant for 1.0, such as
	  jhbuild.

2013-01-11 19:24:43 +0400  Alexey Chernov <achernov@neosphere.com>

	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: Fix crash in osxvideosink with external window output

2013-01-16 12:04:59 +0400  Alexey Chernov <achernov@neosphere.com>

	* sys/osxvideo/cocoawindow.m:
	  osxvideosink: Make GstGLView propagate input events to its parent view
	  Fixes bug #691832

2013-01-16 10:19:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: update some fields in the caps to their new name
	  and to match the parser. "mode" got renamed to "channel-mode"
	  and "allocation" to "allocation-method".

2013-01-15 17:44:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	  docs: add sbcparse and rtpsbcpay to plugin docs

2013-01-15 17:38:24 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstsbcparse.h:
	* gst/audioparsers/plugin.c:
	  audioparsers: add SBC audio parser
	  From-scratch rewrite, the bluez one was useless and broken.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690582

2013-01-15 15:05:04 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From a72faea to a942293

2013-01-10 12:38:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsbcpay.h:
	  rtp: import rtpsbcpay from bluez and port to 1.0
	  Compiles, but not tested yet (sbc elements still need to be ported).
	  https://bugzilla.gnome.org/show_bug.cgi?id=690582

2013-01-09 19:59:16 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/Makefile.am:
	* gst/dtmf/gstdtmf.c:
	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfdetect.h:
	* gst/dtmf/tone_detect.c:
	* gst/dtmf/tone_detect.h:
	  dtmf/spandsp: Move dtmfdetect to use libspandsp
	  Remove our copy of the tone_detect.c file and use the original
	  from libspandsp. Also move the element to the spandsp plugin.

2011-02-13 17:51:45 -0800  Marcel Holtmann <marcel@holtmann.org>

	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: Remove workaround for compiler warnings

2010-05-19 16:59:30 +0200  Marcel Holtmann <marcel@holtmann.org>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Add pragma based workaround for GStreamer warnings

2010-01-01 17:08:17 -0800  Marcel Holtmann <marcel@holtmann.org>

	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: Update copyright information

2009-01-30 00:31:15 +0100  Marcel Holtmann <marcel@holtmann.org>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin

2009-01-01 19:33:20 +0100  Marcel Holtmann <marcel@holtmann.org>

	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: Update copyright information

2008-12-23 05:25:50 +0100  Marcel Holtmann <marcel@holtmann.org>

	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup)

2008-12-20 21:42:49 +0200  Johan Hedberg <johan.hedberg@nokia.com>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: More coding style fixes

2008-02-29 19:37:15 +0000  Luiz Augusto von Dentz <luiz.dentz@openbossa.org>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Remove possible extra memcpy for gstreamer plugin.

2008-02-28 19:38:53 +0000  Luiz Augusto von Dentz <luiz.dentz@openbossa.org>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Fix bug sending empty packages and remove a buffer copy.

2008-02-20 13:37:00 +0000  Luiz Augusto von Dentz <luiz.dentz@openbossa.org>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Fix runtime warnings of gstreamer plugin.

2008-02-19 19:49:24 +0000  Luiz Augusto von Dentz <luiz.dentz@openbossa.org>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Update gstreamer plugin to use new sbc API.

2008-02-02 03:37:05 +0000  Marcel Holtmann <marcel@holtmann.org>

	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: Update copyright information

2008-01-30 14:21:43 +0000  Luiz Augusto von Dentz <luiz.dentz@openbossa.org>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Fixes gstreamer caps and code cleanup.

2008-01-24 14:25:29 +0000  Luiz Augusto von Dentz <luiz.dentz@openbossa.org>

	* gst/rtp/gstrtpsbcpay.c:
	  rtpsbcpay: Fix gtreamer payloader sending fragmented frames.

2008-01-23 19:17:33 +0000  Luiz Augusto von Dentz <luiz.dentz@openbossa.org>

	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps.

2008-01-23 13:14:02 +0000  Luiz Augusto von Dentz <luiz.dentz@openbossa.org>

	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsbcpay.h:
	  rtpsbcpay: Make a2dpsink to act like a bin and split the payloader.

2013-01-08 16:27:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtp: small improvements

2013-01-07 15:50:33 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor handle sync code
	  Move the code that combines the last SR packet and the current jitterbuffer sync
	  values into a sync structure, into its own function. We want to reuse this bit
	  later.

2013-01-07 15:45:10 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtp: include downstream latency in SR calculations
	  When we make a mapping between an RTP timestamp and an NTP timestamp, include
	  the downstream latency applied to the sinks. This makes it possible to have
	  both sinks run with different latencies and still have correct sync on the
	  client. It also is more correct because the RTP timestamp in the SR report will
	  actually correspond more closely to the NTP time it was sent on the server.
	  For pipelines with high latency on the sender side, this actually allows a
	  GStreamer receiver to perform synchronisation instead of dropping the RTCP
	  packets.

2013-01-07 14:25:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: don't cast event functions
	  There is no need to cast the event functions and only causes problems later when
	  we change the signature later and things silently compiles wrong code.

2013-01-07 14:23:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtp: more debug

2013-01-07 14:22:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: improve debug

2013-01-02 00:03:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/udp/gstudpsrc.c:
	  udpsrc: sanity check size of available packet data for reading to avoid memory waste
	  On Windows and OS/X, _get_available_bytes() may not return the size
	  of the next pending packet, but the size of all pending packets in
	  the kernel-side buffer, which might be rather large depending on
	  configuration. Sanity-check the size returned by _get_available_bytes()
	  to make sure we never allocate more memory than the max. size for
	  a packet, if it's an IPv4 socket.
	  https://bugzilla.gnome.org/show_bug.cgi?id=610364

2013-01-04 10:03:32 +0100  Robert Krakora <rob.krakora@messagenetsystems.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: Also handle the new ENOENT return value of VIDIOC_QUERYCTRL
	  https://bugzilla.gnome.org/show_bug.cgi?id=691098

2013-01-01 19:14:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/souphttpsrc.c:
	  tests: add test for souphttpsrc error handling with data
	  https://bugzilla.gnome.org/show_bug.cgi?id=678429

2012-06-22 21:56:52 +0000  Norbert Waschbuesch <nwaschbu@opentv.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: error out properly when receiving data along with an error status
	  When receiving an error code from the http server, such as 404,
	  data might be sent along with it, like a web page. We don't want
	  to output that data in this case, and we also want to pass the
	  FLOW_ERROR return back to the base class, so it can stop properly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=678429

2013-01-01 12:20:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	* docs/plugins/gst-plugins-good-plugins.args:
	  docs: update for new rtspsrc proxy-id and proxy-pw properties

2013-01-01 12:19:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-cairo.xml:
	  docs: fix docs build and update after removal of old cairo elements

2013-01-01 12:12:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ext/cairo/Makefile.am:
	* ext/cairo/gstcairo.c:
	* ext/cairo/gstcairorender.c:
	* ext/cairo/gstcairorender.h:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttextoverlay.h:
	* ext/cairo/gsttimeoverlay.c:
	* ext/cairo/gsttimeoverlay.h:
	  cairo: remove old cairo-based text renderering element
	  They haven't worked well or at all in a very long time
	  and were rather bit-rotten, and there's no need for them
	  any more.

2013-01-01 11:52:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	* ext/cairo/.gitignore:
	* ext/cairo/Makefile.am:
	* ext/cairo/gstcairo-marshal.list:
	* ext/cairo/gstcairo.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairooverlay.h:
	* tests/examples/Makefile.am:
	* tests/examples/cairo/Makefile.am:
	* tests/examples/cairo/cairo_overlay.c:
	  cairo: port cairooverlay to 0.11
	  The other elements are not that interesting now that we're
	  using pangocairo in the pango plugin, and should probably
	  just be removed.

2012-12-31 18:59:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	* tests/examples/rtp/server-decodebin-H263p-AMR.sh:
	  examples: check for uri argument in decodebin-h264p-amr server example
	  Otherwise people get a rather confusing error message.

2012-12-31 00:22:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add "proxy-id" and "proxy-pw" properties
	  to match souphttpsrc. user/password passed via the URI
	  will still take precedence though.
	  https://bugzilla.gnome.org/show_bug.cgi?id=395427

2012-12-25 16:48:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	* sys/oss4/oss4-sink.c:
	  oss4sink: notify "volume" property on open to make apps query initial volume
	  The initial volume might not be the property default, so
	  emit a notify on the volume property to make apps get
	  an up-to-date reading of the current volume.
	  https://bugzilla.gnome.org/show_bug.cgi?id=631053

2012-12-20 17:12:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix cmd comparison
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690476

2012-12-20 17:12:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add some more debug

2012-12-20 16:44:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/raw1394/gst1394clock.c:
	  1394clock: mark our clock type as OTHER

2012-12-20 16:15:13 +0100  Jonas Holmberg <jonashg@axis.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: add jpegpay unit test
	  See also https://bugzilla.gnome.org/show_bug.cgi?id=684955

2012-12-20 15:55:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpegenc: pass flowreturn upstream

2012-09-27 15:42:56 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: handle width and height > 2040
	  If width or height is greater than 2040 set width and height to zero in
	  the rtp header and add x-dimensions to outcaps.
	  Solves #684955

2012-12-20 13:03:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: cleanup in flag define

2012-12-20 13:02:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: improve debug

2012-12-18 15:56:59 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* ext/wavpack/gstwavpackenc.c:
	  wavpack: use appropriate printf format for gsize

2012-12-18 15:55:43 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* ext/taglib/gstid3v2mux.cc:
	  taglib: use appropriate printf format for gsize

2012-12-18 15:54:08 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	  gdkpixbuf: use appropriate printf format for gsize

2012-12-18 15:51:46 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtp/gstrtpgstdepay.c:
	  rtp: use appropriate printf format for gsize

2012-12-18 15:46:56 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: use appropriate printf format for gsize

2012-12-17 16:35:56 +0100  Philippe Normand <philn@igalia.com>

	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	  interleave: set src pad caps upon last sink pad CAPS event
	  Gather caps on all sink pads before setting the src pad caps. This is
	  specially needed when the audio channel mapping is set on the sink
	  pads and the element needs to preserve it on its src pad.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690267

2012-12-17 22:55:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: skip empty tags
	  instead of trying to add tags with empty strings, which
	  causes criticals at runtime.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690358

2012-12-17 15:17:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: Make sure the caps are actually writable before changing them

2012-12-17 15:01:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: Use the peer caps for restrictions instead of the srcpad allowed caps
	  Otherwise we will intersect with the srcpad template caps and add all the caps fields
	  that the parser will ever set, no matter if downstream restricts this field or not.
	  This requires upstream to set this field on the caps to successfully negotiate.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690184

2012-12-14 22:25:08 +0000  Koop Mast <kwm@rainbow-runner.nl>

	* configure.ac:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Teach where the videodev2.h header lives on freebsd.
	  https://bugzilla.gnome.org/show_bug.cgi?id=690233

2012-12-16 23:27:41 +0000  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: set appropriate block header flag for VP8 invisible frames
	  Useful for debugging mostly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654259

2012-12-16 15:25:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* gst/rtpmanager/gstrtpdtmfmux.c:
	  docs: add rtpmux and rtpdtmfmux to plugin docs
	  https://bugzilla.gnome.org/show_bug.cgi?id=629117

2012-12-16 15:13:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtpmuxer.c:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	  rtpmanager: move rtpmux and rtpdtmfmux elements from -bad
	  https://bugzilla.gnome.org/show_bug.cgi?id=629117

2012-11-03 20:38:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpdtmfmux.h:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* gst/rtpmanager/gstrtpmuxer.c:
	* tests/check/elements/rtpmux.c:
	  rtpmux: Fix FSF address
	  https://bugzilla.gnome.org/show_bug.cgi?id=687520

2012-10-17 17:34:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Use gst_element_class_set_static_metadata()
	  where possible. Avoids some string copies. Also re-indent
	  some stuff. Also some indent fixes here and there.

2012-09-10 20:38:14 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	* tests/check/elements/rtpmux.c:
	  rtpmux: Misc fix for 0.11
	  Convert the incoming caps before proxying them
	  Clear the last_pad when going to ready
	  tests: Implement accept_caps, don't leak event

2012-07-17 16:39:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: update for RTP buffer api changes

2012-04-05 18:02:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpmuxer.c:
	  rtpmux: Update for GST_PLUGIN_DEFINE() API changes

2012-04-02 11:07:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: fix compilation

2012-03-11 19:06:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: fix for caps api changes

2012-01-26 06:58:46 -0500  Matej Knopp <matej.knopp@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Fix compiler warnings

2012-01-29 18:01:05 +0000  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Unref non-forwarded events
	  Also, don't unref forwarded ones

2012-01-28 16:57:03 +0000  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: resync iterator on resync

2012-01-27 12:08:52 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Re-push sticky events on input pad change

2012-01-25 15:43:01 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Don't leak gvalue from iterator

2012-01-25 16:46:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: more porting

2012-01-24 14:20:52 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* tests/check/elements/rtpmux.c:
	  rtpmux: port to 0.11

2011-11-04 12:22:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: make request pads take _%u

2011-04-14 14:34:26 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpdtmfmux: Add last-stop to dtmf-event upstream events
	  Add the running time of the last outputted buffer to the
	  upstream "dtmf-event" events so that the dtmf source does not
	  leave a gap.

2010-11-25 19:21:11 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Remove dead assignments

2010-10-19 13:43:14 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: add missing G_PARAM_STATIC_STRINGS flags
	  Canonicalize property names as needed.

2010-09-30 16:07:29 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Improve documentation
	  Add an example pipeline, and try to explain a bit more what it does.

2010-09-24 13:29:55 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: remove unused variable

2010-09-24 13:25:22 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: remove unused signal boilerplate

2010-09-24 13:24:48 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: no need to ref pad in _chain()

2010-08-25 22:56:03 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Unlock the right mutex
	  The mutex locked is for the 'mux' object, but we unlock the
	  pad, which means that if the rtpmux gets a flush, then the
	  object lock will stay locked forever, causing it to freeze
	  the next time it tries to take it.
	  Fixes bug #627991

2010-07-01 15:19:12 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Add support for GstBufferList
	  Factor out most of the buffer handling and implement a chain_list
	  function. Also, the DTMF muxer has been modified to just have a
	  function to accept or reject a buffer instead of having to subclass
	  both chain and chain_list.

2010-07-01 15:15:49 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Don't leak invalid buffers

2010-06-03 10:43:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpmux: fix missing debug log message argument

2010-05-10 18:37:55 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: Add some debug messages

2010-05-07 18:56:57 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpdtmfmux.h:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpdtmfmux: Remove stream-lock event handling

2010-05-07 18:54:49 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: Update doc for simplification

2010-05-07 18:40:30 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* tests/check/elements/rtpmux.c:
	  tests: Change tests to not use the priority pads instead of the events

2010-05-06 19:51:59 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpdtmfmux.h:
	  rtpdtmfmux: Drop buffers on non-priority sinks when something is incoming on the priority sink

2010-05-06 18:11:40 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpdtmfmux: Add priority sink pads

2010-05-07 17:15:47 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: Cleanup event function

2010-05-07 16:42:22 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* tests/check/elements/rtpmux.c:
	  rtpmux: Aggregate incoming segments

2010-05-06 19:09:48 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: Update documentation

2010-05-06 18:10:45 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Simplify request pad creation

2010-03-21 21:39:18 +0100  Benjamin Otte <otte@redhat.com>

	* tests/check/elements/rtpmux.c:
	  Add -Wmissing-declarations -Wmissing-prototypes to configure flags
	  And fix all warnings

2010-03-18 17:30:26 +0100  Benjamin Otte <otte@redhat.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: gst_element_class_set_details => gst_element_class_set_details_simple

2009-11-18 16:38:33 +0100  unknown <havard.graff@.eu.tandberg.int>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: update the current_ssrc from the caps
	  Fixes #604101

2009-12-09 14:42:21 +0100  Håvard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: release pads when disposing
	  Because of an allocated priv (GstRTPMuxPadPrivate), the element will
	  leak memory if not gst_rtp_mux_release_pad() is called. This would
	  previously only happen if release_request_pad() was called explicitly,
	  somthing that should not be neccesary.
	  Fixes #604099

2009-12-09 13:40:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  dtmfmux: method name cleanups

2009-10-08 19:06:26 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* tests/check/elements/rtpmux.c:
	  tests: Add test for rtpdtmfmux locking

2009-09-28 19:54:53 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* tests/check/elements/rtpmux.c:
	  tests: Add unit test for rtpmux

2009-09-28 13:36:44 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Don't ignore requested pad name

2009-07-29 17:23:31 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Remove empty finalize

2009-07-21 15:31:33 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Free the pad private data on pad release
	  Free the pad private data on pad release instead of using a weak ref,
	  which is not thread safe. Also, lock the content of the pad private using the element's
	  object lock.

2009-04-28 16:10:21 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Reject wrong caps

2009-04-28 16:03:19 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>

2009-04-28 15:58:41 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Fix leak
	  Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>

2009-04-22 18:01:07 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Fix warning

2009-04-20 20:00:15 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Set different caps depending on the input

2009-04-22 16:25:07 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Only free pad private when pad is disposed

2009-04-20 18:41:39 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Remove useless caps mangling

2009-04-20 18:36:42 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Rename variable for more clarity

2009-04-20 17:43:39 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Use GST_BOILERPLATE

2009-04-20 17:42:40 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpdtmfmux.h:
	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Do the includes locally

2009-04-15 13:23:01 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Add GST_DEBUG_FUNCPTRs

2009-04-15 13:15:55 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: Release locked pad on release_pad
	  Release the special pad if the pad is removed from the muxer.

2009-04-15 13:09:27 -0400  Laurent Glayal <spglegle@yahoo.fr>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpdtmfmux: Release special on pad dispose
	  Fixes #577690

2009-02-25 11:45:05 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	  docs: various doc fixes
	  No short-desc as we have them in the element details.
	  Also keep things (Makefile.am and sections.txt) sorted.
	  Reword ambigous returns. No text after since please.

2009-02-10 17:02:24 +0000  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmuxer.c:
	  rtpmux: Move rtpmux from gst-plugins-farsight to -bad

2009-02-20 17:45:50 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpdtmfmux.h:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* gst/rtpmanager/gstrtpmuxer.c:
	  rtpmux: Re-indent to Gst style

2009-02-10 19:11:15 +0000  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Document rtp muxer a bit

2009-02-20 13:30:49 -0500  Laurent Glayal <spglegle@yahoo.fr>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpdtmfmux.h:
	  rtpmux: Add signals before stream lock and after unlocking

2009-02-18 20:18:46 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Let ssrc through getcaps

2009-02-18 19:58:58 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Rename have_base to have_ts_base

2009-02-18 18:14:52 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Protect the seqnum with object lock in rtpmux

2009-02-18 18:07:44 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Remove unused sink_ts_base

2009-02-18 15:20:58 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Have getcaps to force the same clockrate on all pads

2009-02-18 17:05:13 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Validate RTP data in RTP Mux

2009-02-18 14:16:00 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Remove unused clock-rate property

2009-02-18 13:56:36 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpdtmfmux.h:
	  rtpmux: Clarify locking in rtpdtmfmux

2009-02-18 13:32:56 -0500  Laurent Glayal <spglegle@yahoo.fr>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Missing format parameter

2008-12-01 17:55:22 -0500  Håvard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Update seqnum base in rtp muxer
	  With help from Wim

2008-12-01 17:54:58 -0500  Håvard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Fix some more leaks

2008-12-01 17:48:29 -0500  Håvard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpmux: Fix leak

2008-09-29 15:03:05 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Don't unref caps we don't know (thanks Wim)

2008-08-12 12:48:02 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Put per-buffer debug at level LOG

2008-08-12 12:47:14 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Make debug print accurate

2008-08-12 12:46:23 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Set our caps on the buffers

2008-08-12 12:46:07 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Take the clock-base stored from the last setcaps

2008-08-12 12:41:59 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Store the clock-base on setcaps

2008-08-12 12:30:52 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Add padprivate to the request pads

2008-08-11 21:20:06 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Make indentation more correct

2008-08-11 21:05:34 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Fix typo

2008-08-11 21:03:22 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Set seqnum-base and clock-base in caps from rtpmuxer

2007-08-15 13:50:38 +0000  Zeeshan Ali <first.last@nokia.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpmux: more debug
	  20070815135038-f3f1e-9c7a5490a525c6e8753cb1b8c03354df99132b5c.gz

2007-08-20 18:50:32 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: missing comment
	  20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz

2007-07-12 19:53:36 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Make buffer writable before writing into it
	  20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz

2007-07-06 20:24:59 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Set pads active when adding them to a potentially running element
	  20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz

2007-06-07 12:01:21 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Fix multiple ref leaks (patches by SP GLE)
	  20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz

2007-05-28 15:25:05 +0000  Zeeshan Ali <first.last@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: send event to all src pads
	  20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz

2007-05-28 12:37:49 +0000  Zeeshan Ali <first.last@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: print a warning if receive an error iterating sinkpads
	  20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz

2007-05-28 12:28:08 +0000  Zeeshan Ali <first.last@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: deal with all the gst_iterator_next() return values
	  20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz

2007-05-25 12:31:16 +0000  Zeeshan Ali <first.last@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Return correct value from the event handler
	  20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz

2007-05-25 10:27:09 +0000  Zeeshan Ali <first.last@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Ville's original patch to fix the traversal of dtmf event
	  20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz

2007-03-29 13:52:50 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Set the correct ts-offset on the get_prop value
	  20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz

2007-03-29 13:52:23 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Refactorize state_change
	  20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz

2007-03-29 13:36:22 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: set SSRC on the packets
	  20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz

2007-03-29 13:19:36 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Code clean-up and more debug output
	  20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz

2007-03-28 11:22:19 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Use own clock-base
	  20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz

2007-03-23 16:31:39 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Only accept RTP streams that have the same clock-rate
	  20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz

2007-03-22 16:15:52 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpmux: Some more code-cleanups
	  20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz

2007-03-22 15:42:51 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: return newpad instead of NULL and warn if failed to create a pad
	  20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz

2007-03-22 12:41:32 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Refactorize the RTPMux code
	  20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz

2007-03-22 12:14:53 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpmux: Some more doc fixing
	  20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz

2007-03-22 11:32:28 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpmux: More Refactoring
	  20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz

2007-03-22 11:31:54 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpmux: More documentation
	  20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz

2007-03-21 16:33:11 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	  rtpmux: Refactor the event handler function
	  20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz

2007-03-21 14:52:44 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpdtmfmux.c:
	* gst/rtpmanager/gstrtpdtmfmux.h:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* gst/rtpmanager/gstrtpmuxer.c:
	  rtpmux: Add RTPDTMFMux element
	  20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz

2007-03-21 12:31:49 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable
	  20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz

2007-03-20 12:05:24 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Put more helpful description
	  20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz

2007-03-16 15:16:41 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: remove the (commented-out) code for blocking the pads
	  20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz

2007-03-16 13:14:44 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Drop buffers instead of blocking the sinkpads
	  20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz

2007-03-14 17:16:18 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Implement stream locking, needed for DTMF
	  20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz

2007-03-14 10:20:58 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: use GST_*_OBJECT instead of g_*
	  20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz

2007-03-14 10:18:54 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: No need to manage pads, parent does that for us
	  20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz

2007-03-14 09:03:58 +0000  zeenix@gmail.com <zeenix@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Fix copyright header
	  20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz

2007-03-07 08:53:07 +0000  zeeshan.ali@nokia.com <zeeshan.ali@nokia.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: The first implementation of RTP muxer
	  20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz

2012-12-15 21:27:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempo.h:
	  scaletempo: no need for a private struct

2012-12-14 15:13:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	  docs: update plugin docs

2012-12-14 15:13:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-audiofx.xml:
	  docs: add scaletempo to docs

2012-11-06 13:36:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofx.c:
	  audiofx: move scaletempo element from -bad
	  https://bugzilla.gnome.org/show_bug.cgi?id=687262

2012-10-23 14:33:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Fix event leak

2012-10-23 14:32:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Fix timestamp tracking

2012-10-23 14:06:37 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Implement LATENCY query

2012-10-23 13:39:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempo.h:
	  scaletempo: Store instance private data in the instance struct
	  Getting it over and over again via G_TYPE_INSTANCE_GET_PRIVATE()
	  is really slow.

2012-10-17 17:34:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: use gst_element_class_set_static_metadata()
	  where possible. Avoids some string copies. Also re-indent
	  some stuff. Also some indent fixes here and there.

2012-09-14 17:08:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: replace gst_element_class_set_details_simple with gst_element_class_set_metadata

2012-09-14 16:45:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: ffmpegcolorspace is no more

2012-04-05 18:02:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/gstscaletempoplugin.c:
	  scaletempo: Update for GST_PLUGIN_DEFINE() API changes

2012-03-18 18:32:55 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: port to 0.11

2011-07-07 10:52:50 -0700  Stefan Kost <ensonic@users.sf.net>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: improve the docs
	  Fix the syntax, add more explanation and xref the properties.

2011-03-22 13:46:42 +0100  Chris E Jones <chris@chrisejones.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Correctly handle newsegment events with stop==-1
	  Fixes bug #645420.

2010-10-19 13:43:14 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: add missing G_PARAM_STATIC_STRINGS flags
	  Canonicalize property names as needed.

2010-03-18 17:30:26 +0100  Benjamin Otte <otte@redhat.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple

2009-11-05 13:40:38 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: properly update new segments
	  Scaletempo was missing an update of 'stop' in
	  new segment parameters when pushing it downstream,
	  which caused files to end earlier when rate < 1.
	  Fixes #599903
	  Based on patch by: Bastian Hecht <hechtb@gmail.com>

2009-06-14 20:00:51 +0200  Maximilian Högner <pbmaxi@hoegners.de>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Explicitely cast to signed integers to fix a segfault
	  Fixes bug #585660.

2009-02-13 12:18:48 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Do not use void pointer arithmetic.

2008-10-30 12:13:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  scaletempo: Return the result of parent_class->event()
	  Original commit message from CVS:
	  * gst/audiofx/gstscaletempo.c:
	  Return the result of parent_class->event().

2008-08-31 12:20:33 +0000  Rov Juvano <rovjuvano@users.sourceforge.net>

	  Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r...
	  Original commit message from CVS:
	  Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net>
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-scaletempo.xml:
	  * examples/scaletempo/Makefile.am:
	  * examples/scaletempo/demo-gui.c: (pop_status_bar),
	  (status_bar_printf), (demo_gui_seek_bar_format), (update_position),
	  (demo_gui_seek_bar_change), (demo_gui_do_change_rate),
	  (demo_gui_do_set_rate), (demo_gui_do_rate_entered),
	  (demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled),
	  (demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause),
	  (demo_gui_do_play_pause), (demo_gui_do_open_file),
	  (demo_gui_do_playlist_prev), (demo_gui_do_playlist_next),
	  (demo_gui_do_about_dialog), (demo_gui_do_quit),
	  (demo_gui_request_set_stride), (demo_gui_request_set_overlap),
	  (demo_gui_request_set_search), (demo_gui_rate_changed),
	  (demo_gui_playing_started), (demo_gui_playing_paused),
	  (demo_gui_playing_ended), (demo_gui_player_errored),
	  (demo_gui_stride_changed), (demo_gui_overlap_changed),
	  (demo_gui_search_changed), (demo_gui_set_player_func),
	  (demo_gui_set_playlist_func), (build_gvalue_array),
	  (create_action), (demo_gui_show_func), (demo_gui_set_player),
	  (demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property),
	  (demo_gui_set_property), (demo_gui_init), (demo_gui_class_init),
	  (demo_gui_get_type):
	  * examples/scaletempo/demo-gui.h:
	  * examples/scaletempo/demo-main.c: (handle_error_message),
	  (handle_quit), (main):
	  * examples/scaletempo/demo-player.c: (no_pipeline),
	  (demo_player_event_listener), (demo_player_state_changed_cb),
	  (demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate),
	  (demo_player_scale_rate_func), (demo_player_set_rate_func),
	  (_set_state_and_wait), (demo_player_load_uri_func),
	  (demo_player_play_func), (demo_player_pause_func), (_seek_to),
	  (demo_player_seek_by_func), (demo_player_seek_to_func),
	  (demo_player_get_position_func), (demo_player_get_duration_func),
	  (demo_player_scale_rate), (demo_player_set_rate),
	  (demo_player_load_uri), (demo_player_play), (demo_player_pause),
	  (demo_player_seek_by), (demo_player_seek_to),
	  (demo_player_get_position), (demo_player_get_duration),
	  (demo_player_get_property), (demo_player_set_property),
	  (demo_player_init), (demo_player_class_init),
	  (demo_player_get_type):
	  * examples/scaletempo/demo-player.h:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/gstscaletempo.c: (best_overlap_offset_float),
	  (best_overlap_offset_s16), (output_overlap_float),
	  (output_overlap_s16), (fill_queue), (reinit_buffers),
	  (gst_scaletempo_transform), (gst_scaletempo_transform_size),
	  (gst_scaletempo_sink_event), (gst_scaletempo_set_caps),
	  (gst_scaletempo_get_property), (gst_scaletempo_set_property),
	  (gst_scaletempo_base_init), (gst_scaletempo_class_init),
	  (gst_scaletempo_init):
	  * gst/audiofx/gstscaletempo.h:
	  * gst/audiofx/gstscaletempoplugin.c: (plugin_init):
	  Add scaletempo plugin, which allows to scale the speed of audio without
	  changing the pitch by handling seeks with a rate!=1.0.
	  Integrate it into the docs and add the example application for it.
	  Fixes bug #537700.

2012-12-13 12:36:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  check: add (but disable) more rtp jitterbuffer tests
	  Tests need to be ported to 1.0 before they can be enabled but added here so they
	  don't get forgotten.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=667838

2012-01-13 01:11:31 +0100  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: bundle together late lost-events
	  The scenario where you have a gap in a steady flow of packets of
	  say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer
	  will idle up until it receives the first buffer after the gap, but will
	  then go on to produce 499 lost-events, to "cover up" the gap.
	  Now this is obviously wrong, since the last possible time for the earliest
	  lost-events to be played out has obviously expired, but the fact that
	  the jitterbuffer has a "length", represented with its own latency combined
	  with the total latency downstream, allows for covering up at least some
	  of this gap.
	  So in the case of the "length" being 200ms, while having received packet
	  500, the jitterbuffer should still create a timeout for packet 491, which
	  will have its time expire at 10,02 seconds, specially since it might
	  actually arrive in time! But obviously, waiting for packet 100, that had
	  its time expire at 2 seconds, (remembering that the current time is 10)
	  is useless...
	  The patch will create one "big" lost-event for the first 490 packets,
	  and then go on to create single ones if they can reach their
	  playout deadline.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=667838

2012-12-13 09:27:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix TCP reconnect
	  Ignore other commands when reconnecting, otherwise the loop function would pause
	  and the reconnection would not happen. Continue looping after doing a reconnect
	  so that we have a chance to actually read the new data.

2012-12-13 01:02:34 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	* sys/waveform/gstwaveformsink.h:
	  directsound, waveform: fix compilation errors caused by circular includes
	  https://bugzilla.gnome.org/show_bug.cgi?id=690124

2012-12-12 17:35:04 +0000  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackutil.h:
	* ext/libpng/gstpngenc.c:
	* ext/pulse/pulseprobe.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	* ext/vpx/gstvp8enc.c:
	* sys/oss/common.h:
	* sys/oss/gstossaudio.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-audio.h:
	  ext/sys: Fix some compilation errors caused by circular includes

2012-12-12 12:07:34 +0100  Philippe Normand <philn@igalia.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: properly set srcpad channel position
	  The src pad caps always describe a single audio channel so only the
	  first position matters if deinterleave is configured to keep channel
	  positions in its src pads.

2012-12-12 11:09:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: timeout on udpsrc is in nanoseconds

2012-12-12 11:08:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: improve timeouts
	  Make it possible to set the timeout after we went to the READY state by using
	  the timeout when checking the condition. This also makes it possible to set the
	  timeout with a higher granularity than seconds.

2012-12-11 13:00:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: add support for strides
	  Implement stride support correctly by taking it from the GstVideoFrame.
	  Propose a bufferpool upstream when not operating in passthrough.

2012-09-27 12:17:58 -0700  Aleix Conchillo Flaque <aleix@oblong.com>

	  rtspsrc: do not change state to PLAYING if currently chaning state
	  * gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be
	  happening in the application thread, so we don't change the state to
	  PLAYING in the gstrtspsrc thread unless it is safe.
	  A specific case is when chaning the state to NULL from the application
	  thread. This will synchronously try to stop the task (with the element
	  state lock acquired), but we will try a gst_element_set_state from
	  gstrtspsrc thread which will block on the element state lock causing a
	  deadlock.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684312

2012-12-10 11:44:26 +0000  Alexey Chernov <4ernov@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: Fix resizing the Cocoa window on receiving new caps
	  Fixes bug #689732.

2012-11-30 20:37:47 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* sys/v4l2/Makefile.am:
	  v4l2src: link against -lrt for clock_gettime()
	  Need to explicitly link against -lrt for clock_gettime(), which
	  we don't get in the libs any more, because core moved the
	  gmodule-no-export-2.0 bit into Requires.Private.
	  Not required for newer glibc, but for older ones, so check for that.

2012-11-30 17:22:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/shout2/gstshout2.c:
	  shout2send: accept audio/webm as well as video/webm
	  https://bugzilla.gnome.org/show_bug.cgi?id=689336

2012-11-30 17:20:18 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	* tests/check/elements/matroskamux.c:
	  webmux: fix linking with shout2send element
	  Shout2send only accepts webm format, not matroska, but due
	  to a bug in matroskamux, webmmux's source pad is also created
	  with the matroska source pad template as pad template, which
	  makes the link function think it can't link webmmux to shout2send.
	  Also add unit test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=689336

2012-11-27 11:13:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: use new option parser function

2012-11-26 15:17:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/law/mulaw-conversion.c:
	  law: fix accidental file permissions change
	  https://bugzilla.gnome.org/show_bug.cgi?id=687469

2012-11-25 16:05:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: remove unused define

2012-11-25 14:16:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid criticals if unknown fourcc has space at beginning or end
	  https://bugzilla.gnome.org/show_bug.cgi?id=682936

2012-11-24 19:32:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/videobox/gstvideobox.c:
	  videobox: fix border filling for planar YUV formats
	  We would get a green border instead of a black one, for
	  example.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684991

2012-11-24 14:27:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/law/mulaw-conversion.c:
	  mulaw: const-ify some arrays

2012-11-02 12:38:44 -0400  Roland Krikava <rkrikava@gmail.com>

	* gst/law/mulaw-conversion.c:
	  mulawdec: fix integer overrun
	  There might be more than 65535 samples in a chunk of data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=687469

2012-11-22 11:34:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: pause the task instead of spinning
	  Actually pause the loop task instead of spinning forever.

2012-11-19 03:31:37 -0500  Joshua M. Doe <oss@nvl.army.mil>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Add gray 8/16 support

2012-11-19 11:25:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	* common:
	  Automatic update of common submodule
	  From b497c4f to a72faea

2012-11-16 15:38:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle segment event
	  Make a segment event when we send a new range header to a client (first PLAY
	  request or after a seek). Send the segment event in interleaved mode.
	  Clean the segment event on cleanup
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382

2012-11-16 15:18:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix check for active streams
	  A stream can be active without a srcpad yet and we want to send
	  events on those streams as well.

2012-11-16 13:31:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: create and add pads outside of lock
	  Create and add the ghostpad for the new stream outside of the lock because it
	  is not needed and causes deadlocks.

2012-09-12 22:11:20 -0700  Aleix Conchillo Flaque <aleix@oblong.com>

	  rtspsrc: allow client to disable reconnection
	  * gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before,
	  rtspsrc always tried to reconnect to the server when the RTSP
	  connection was closed by the server. This property lets the user
	  decide whether it wants rtspsrc to reconnect or not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=683912

2012-11-16 12:16:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: clear variables before retrying
	  Else we might unref an old udpsrc twice in cleanup.

2012-11-16 12:00:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: propose ports in multicast
	  When the user configured a port-range, propose ports from this range
	  as the multicast ports. The server is free to ignore this request but if it
	  honours it, increment our ports so that we suggest the next port pair for the
	  next stream.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420

2012-11-16 11:58:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add more debug

2012-11-16 09:09:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: post messages in max-size mode as well
	  No reason not to really.

2012-11-15 14:37:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: post error before stopping

2012-11-14 00:13:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	  gst_adapter_prev_timestamp -> gst_adapter_prev_pts
	  https://bugzilla.gnome.org/show_bug.cgi?id=675598

2012-11-12 19:23:41 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Add NV12/NV21 support
	  https://bugzilla.gnome.org/show_bug.cgi?id=688225

2012-11-12 13:01:23 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Don't leak GstVideoCodecFrames that cause the creation of invisible frames
	  Fixes bug #682714.

2012-11-12 11:47:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulse: Use new GType for GThread instead of just G_TYPE_POINTER

2012-11-12 11:14:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: protect against invalid RTP packets

2012-11-12 10:44:01 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	  pngdec: Actually use the stop() vfunc implementation

2012-11-12 10:31:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Fix last commit

2012-11-12 10:10:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	  pngdec: Keep the input state in reset()
	  It's still valid after a flush and we might not get a new one.

2012-11-12 10:08:57 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Also destroy decoder in set_format() if it was created already
	  Fixes a memory leak.

2012-11-12 09:48:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Don't clear input state in reset()
	  The input state is still valid after flushing until
	  new caps arrive.
	  Fixes bug #688092.

2012-11-10 18:21:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: add support for YV12
	  We can do I420, so we can do YV12 as well.

2012-11-10 12:39:08 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: don't write stream headers with key-unit-event
	  Don't write stream headers, let upstream elements insert them in the stream if
	  all_headers=true is set in key unit events.

2012-11-09 13:27:16 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	  videocrop: Add NV12/NV21 support
	  https://bugzilla.gnome.org/show_bug.cgi?id=687964

2012-11-09 16:31:05 +0100  Debarshi Ray <rishi@gnu.org>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Don't give up so easily if failed to decode a frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=687436

2012-11-09 11:22:30 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Also clear GError

2012-11-09 11:20:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Don't error out if we get an ICMP destination-unreachable message when trying to read packets
	  See bug #529454 and #687782 and commit
	  751f2bb3646f2beff3698c9f09900dbd0ea08abb

2012-11-07 20:35:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	  configure.ac: update courtesy of autoupdate

2012-11-07 18:48:49 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	* configure.ac:
	  configure: let AG_GST_PLUGIN_DOCS check for python
	  And update common for move from AS_PATH_PYTHON to AM_PATH_PYTHON,
	  which as a side-effect should pick up newer python versions as
	  well.
	  https://bugzilla.gnome.org/show_bug.cgi?id=563903

2012-11-07 13:36:33 +0100  Christian Fredrik Kalager Schaller <uraeus@linuxrisin.org>

	* gst/rtp/Makefile.am:
	  Fix vp8rtp header names in Makefile

2012-11-06 15:03:55 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	* tests/check/elements/videocrop.c:
	  videocrop: Add support for automatic cropping
	  This change enable automatic cropping using -1 set to left, top, right or
	  bottom property. In the case both side are set to automatic cropping, the
	  croping will be done equally on both side (in the odd case, right and
	  bottom cropping will be 1 pixel more).
	  https://bugzilla.gnome.org/show_bug.cgi?id=687761

2012-11-02 16:39:28 +0100  Debarshi Ray <rishi@gnu.org>

	* ext/speex/gstspeexdec.c:
	  speexdec: Don't unmap or finish_frame an invalid GstBuffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=687464

2012-11-06 13:22:58 +0100  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: the RTCP port number is inclusive
	  The configured port number pair has its upper bound set to the maximum
	  allowed RTCP port, inclusive.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=639420

2012-11-03 20:38:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/mpg123audiodec.c:
	  Fix FSF address
	  https://bugzilla.gnome.org/show_bug.cgi?id=687520

2012-11-03 20:38:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/audiofx/gststereo.c:
	* gst/audiofx/gststereo.h:
	  Fix FSF address
	  https://bugzilla.gnome.org/show_bug.cgi?id=687520

2012-11-03 20:38:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfdetect.h:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  Fix FSF address
	  https://bugzilla.gnome.org/show_bug.cgi?id=687520

2012-11-04 00:07:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ext/aalib/gstaasink.c:
	* ext/aalib/gstaasink.h:
	* ext/cairo/gstcairo.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairooverlay.h:
	* ext/cairo/gstcairorender.c:
	* ext/cairo/gstcairorender.h:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttimeoverlay.c:
	* ext/cairo/gsttimeoverlay.h:
	* ext/dv/gstdv.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdec.h:
	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	* ext/dv/gstsmptetimecode.c:
	* ext/dv/gstsmptetimecode.h:
	* ext/flac/gstflac.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	* ext/flac/gstflactag.c:
	* ext/flac/gstflactag.h:
	* ext/gdk_pixbuf/gstgdkanimation.c:
	* ext/gdk_pixbuf/gstgdkanimation.h:
	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.h:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.h:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/gdk_pixbuf/pixbufscale.h:
	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudioclient.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	* ext/jack/gstjackringbuffer.h:
	* ext/jack/gstjackutil.c:
	* ext/jack/gstjackutil.h:
	* ext/jpeg/gstjpeg.c:
	* ext/jpeg/gstjpeg.h:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokedec.h:
	* ext/jpeg/gstsmokeenc.c:
	* ext/jpeg/gstsmokeenc.h:
	* ext/jpeg/smokecodec.c:
	* ext/jpeg/smokecodec.h:
	* ext/jpeg/smokeformat.h:
	* ext/libcaca/gstcacasink.c:
	* ext/libcaca/gstcacasink.h:
	* ext/libpng/gstpng.c:
	* ext/libpng/gstpng.h:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngdec.h:
	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	* ext/mikmod/README:
	* ext/mikmod/gstmikmod.c:
	* ext/mikmod/gstmikmod.h:
	* ext/mikmod/mikmod_types.c:
	* ext/mikmod/mikmod_types.h:
	* ext/pulse/plugin.c:
	* ext/pulse/pulseprobe.c:
	* ext/pulse/pulseprobe.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	* ext/raw1394/gst1394.c:
	* ext/raw1394/gst1394clock.c:
	* ext/raw1394/gst1394clock.h:
	* ext/raw1394/gst1394probe.c:
	* ext/raw1394/gst1394probe.h:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gstdv1394src.h:
	* ext/raw1394/gsthdv1394src.c:
	* ext/raw1394/gsthdv1394src.h:
	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	* ext/soup/gstsouphttpclientsink.h:
	* ext/speex/gstspeex.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexdec.h:
	* ext/speex/gstspeexenc.c:
	* ext/speex/gstspeexenc.h:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstapev2mux.h:
	* ext/taglib/gstid3v2mux.cc:
	* ext/taglib/gstid3v2mux.h:
	* ext/taglib/gsttaglibplugin.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	* ext/vpx/gstvp8utils.c:
	* ext/vpx/gstvp8utils.h:
	* ext/vpx/plugin.c:
	* ext/wavpack/gstwavpack.c:
	* ext/wavpack/gstwavpackcommon.c:
	* ext/wavpack/gstwavpackcommon.h:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackdec.h:
	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackenc.h:
	* ext/wavpack/gstwavpackstreamreader.c:
	* ext/wavpack/gstwavpackstreamreader.h:
	* gst-libs/gst/gettext.h:
	* gst-libs/gst/glib-compat-private.h:
	* gst-libs/gst/gst-i18n-plugin.h:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	* gst/apetag/gstapedemux.c:
	* gst/apetag/gstapedemux.h:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiochebband.h:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiocheblimit.h:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audiodynamic.h:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audioecho.h:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofirfilter.h:
	* gst/audiofx/audiofx.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.h:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioiirfilter.h:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audioinvert.h:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiokaraoke.h:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiopanorama.h:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsincband.h:
	* gst/audiofx/audiowsinclimit.c:
	* gst/audiofx/audiowsinclimit.h:
	* gst/audiofx/math_compat.h:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstamrparse.h:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstdcaparse.h:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstmpegaudioparse.h:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/audioparsers/gstwavpackparse.h:
	* gst/audioparsers/plugin.c:
	* gst/auparse/gstauparse.c:
	* gst/auparse/gstauparse.h:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosink.h:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautoaudiosrc.h:
	* gst/autodetect/gstautodetect.c:
	* gst/autodetect/gstautodetect.h:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosink.h:
	* gst/autodetect/gstautovideosrc.c:
	* gst/autodetect/gstautovideosrc.h:
	* gst/avi/avi-ids.h:
	* gst/avi/gstavi.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/cutter/gstcutter.h:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/cpureport.h:
	* gst/debugutils/gstcapsdebug.c:
	* gst/debugutils/gstcapsdebug.h:
	* gst/debugutils/gstdebug.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavigationtest.h:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstnavseek.h:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gstpushfilesrc.h:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/gsttaginject.h:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/progressreport.h:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	* gst/debugutils/tests.c:
	* gst/debugutils/tests.h:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.asm:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/greedyhmacros.h:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/plugins.h:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	* gst/deinterlace/tvtime/x86-64_macros.inc:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstaging.h:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstdice.h:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstedge.h:
	* gst/effectv/gsteffectv.c:
	* gst/effectv/gsteffectv.h:
	* gst/effectv/gstop.c:
	* gst/effectv/gstop.h:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstquark.h:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstradioac.h:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstrev.h:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstripple.h:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstshagadelic.h:
	* gst/effectv/gststreak.c:
	* gst/effectv/gststreak.h:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstvertigo.h:
	* gst/effectv/gstwarp.c:
	* gst/effectv/gstwarp.h:
	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer.h:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer10bands.h:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/equalizer/gstiirequalizer3bands.h:
	* gst/equalizer/gstiirequalizernbands.c:
	* gst/equalizer/gstiirequalizernbands.h:
	* gst/flv/amfdefs.h:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	* gst/flv/gstindex.c:
	* gst/flv/gstindex.h:
	* gst/flv/gstmemindex.c:
	* gst/flx/flx_color.c:
	* gst/flx/flx_color.h:
	* gst/flx/flx_fmt.h:
	* gst/flx/gstflxdec.c:
	* gst/flx/gstflxdec.h:
	* gst/goom/config_param.c:
	* gst/goom/convolve_fx.c:
	* gst/goom/drawmethods.c:
	* gst/goom/drawmethods.h:
	* gst/goom/filters.c:
	* gst/goom/filters_mmx.s:
	* gst/goom/flying_stars_fx.c:
	* gst/goom/goom.h:
	* gst/goom/goom_config.h:
	* gst/goom/goom_config_param.h:
	* gst/goom/goom_core.c:
	* gst/goom/goom_filters.h:
	* gst/goom/goom_fx.h:
	* gst/goom/goom_graphic.h:
	* gst/goom/goom_plugin_info.h:
	* gst/goom/goom_tools.c:
	* gst/goom/goom_tools.h:
	* gst/goom/goom_typedefs.h:
	* gst/goom/goom_visual_fx.h:
	* gst/goom/graphic.c:
	* gst/goom/gstgoom.c:
	* gst/goom/gstgoom.h:
	* gst/goom/lines.c:
	* gst/goom/lines.h:
	* gst/goom/mathtools.c:
	* gst/goom/mathtools.h:
	* gst/goom/motif_goom1.h:
	* gst/goom/motif_goom2.h:
	* gst/goom/plugin_info.c:
	* gst/goom/ppc_drawings.h:
	* gst/goom/ppc_drawings.s:
	* gst/goom/ppc_zoom_ultimate.h:
	* gst/goom/ppc_zoom_ultimate.s:
	* gst/goom/sound_tester.c:
	* gst/goom/sound_tester.h:
	* gst/goom/surf3d.c:
	* gst/goom/surf3d.h:
	* gst/goom/tentacle3d.c:
	* gst/goom/tentacle3d.h:
	* gst/goom/v3d.c:
	* gst/goom/v3d.h:
	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	* gst/icydemux/gsticydemux.c:
	* gst/icydemux/gsticydemux.h:
	* gst/id3demux/gstid3demux.c:
	* gst/id3demux/gstid3demux.h:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	* gst/interleave/deinterleave.c:
	* gst/interleave/deinterleave.h:
	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	* gst/interleave/plugin.c:
	* gst/interleave/plugin.h:
	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/atomsrecovery.h:
	* gst/isomp4/descriptors.c:
	* gst/isomp4/descriptors.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/ftypcc.h:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/gstqtmoovrecover.h:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux-doc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/gstqtmuxmap.h:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/gstrtpxqtdepay.h:
	* gst/isomp4/isomp4-plugin.c:
	* gst/isomp4/properties.c:
	* gst/isomp4/properties.h:
	* gst/isomp4/qtatomparser.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_lang.c:
	* gst/isomp4/qtdemux_lang.h:
	* gst/isomp4/qtdemux_types.c:
	* gst/isomp4/qtdemux_types.h:
	* gst/isomp4/qtpalette.h:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-decode.h:
	* gst/law/alaw-encode.c:
	* gst/law/alaw-encode.h:
	* gst/law/alaw.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-decode.h:
	* gst/law/mulaw-encode.c:
	* gst/law/mulaw-encode.h:
	* gst/law/mulaw.c:
	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	* gst/matroska/ebml-ids.h:
	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-read.h:
	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	* gst/matroska/matroska.c:
	* gst/matroska/webm-mux.c:
	* gst/matroska/webm-mux.h:
	* gst/monoscope/convolve.c:
	* gst/monoscope/convolve.h:
	* gst/monoscope/gstmonoscope.c:
	* gst/monoscope/gstmonoscope.h:
	* gst/multifile/gstmultifile.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstmultifilesrc.h:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/gstsplitfilesrc.h:
	* gst/multifile/patternspec.c:
	* gst/multifile/patternspec.h:
	* gst/multipart/multipart.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	* gst/multipart/multipartmux.c:
	* gst/multipart/multipartmux.h:
	* gst/rtp/fnv1hash.c:
	* gst/rtp/fnv1hash.h:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstasteriskh263.h:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3depay.h:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpac3pay.h:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvdepay.h:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpbvpay.h:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpchannels.c:
	* gst/rtp/gstrtpchannels.h:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvdepay.h:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpdvpay.h:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722depay.h:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg722pay.h:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723depay.h:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg723pay.h:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729depay.h:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpg729pay.h:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstdepay.h:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263depay.h:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcdepay.h:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpilbcpay.h:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kdepay.h:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpj2kpay.h:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegdepay.h:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpjpegpay.h:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp1sdepay.h:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tdepay.h:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp2tpay.h:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4adepay.h:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4apay.h:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gdepay.h:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4gpay.h:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vdepay.h:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpadepay.h:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpapay.h:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmparobustdepay.h:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvdepay.h:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtpmpvpay.h:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqcelpdepay.h:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpqdmdepay.h:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirendepay.h:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpsirenpay.h:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtpsv3vdepay.h:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheoradepay.h:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtptheorapay.h:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbisdepay.h:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvorbispay.h:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawdepay.h:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtp/gstrtpvrawpay.h:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpjitterbuffer.h:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpptdemux.h:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpsession.h:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtpdec.h:
	* gst/rtsp/gstrtsp.c:
	* gst/rtsp/gstrtsp.h:
	* gst/rtsp/gstrtspext.c:
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	* gst/smpte/barboxwipes.c:
	* gst/smpte/gstmask.c:
	* gst/smpte/gstmask.h:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmpte.h:
	* gst/smpte/gstsmptealpha.c:
	* gst/smpte/gstsmptealpha.h:
	* gst/smpte/paint.c:
	* gst/smpte/paint.h:
	* gst/smpte/plugin.c:
	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudp.c:
	* gst/udp/gstudp.h:
	* gst/udp/gstudpnetutils.c:
	* gst/udp/gstudpnetutils.h:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsink.h:
	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstaspectratiocrop.h:
	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstgamma.h:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideobalance.h:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	* gst/videofilter/gstvideomedian.c:
	* gst/videofilter/gstvideomedian.h:
	* gst/videofilter/gstvideotemplate.c:
	* gst/videofilter/plugin.c:
	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	* gst/videomixer/videomixer2pad.h:
	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	* sys/directsound/gstdirectsoundplugin.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	* sys/oss/common.h:
	* sys/oss/gstossaudio.c:
	* sys/oss/gstossdmabuffer.c:
	* sys/oss/gstossdmabuffer.h:
	* sys/oss/gstosshelper.c:
	* sys/oss/gstosshelper.h:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssink.h:
	* sys/oss/gstosssrc.c:
	* sys/oss/gstosssrc.h:
	* sys/oss4/oss4-audio.c:
	* sys/oss4/oss4-audio.h:
	* sys/oss4/oss4-property-probe.c:
	* sys/oss4/oss4-property-probe.h:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-sink.h:
	* sys/oss4/oss4-source.c:
	* sys/oss4/oss4-source.h:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxaudio/gstosxaudioelement.c:
	* sys/osxaudio/gstosxaudioelement.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxaudiosrc.h:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxringbuffer.c:
	* sys/osxaudio/gstosxringbuffer.h:
	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudio.c:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiomixer.h:
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixeroptions.c:
	* sys/sunaudio/gstsunaudiomixeroptions.h:
	* sys/sunaudio/gstsunaudiomixertrack.c:
	* sys/sunaudio/gstsunaudiomixertrack.h:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosink.h:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/sunaudio/gstsunaudiosrc.h:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2colorbalance.c:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2radio.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/gstv4l2videooverlay.c:
	* sys/v4l2/gstv4l2videooverlay.h:
	* sys/v4l2/gstv4l2vidorient.c:
	* sys/v4l2/gstv4l2vidorient.h:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tuner.h:
	* sys/v4l2/tunerchannel.c:
	* sys/v4l2/tunerchannel.h:
	* sys/v4l2/tunernorm.c:
	* sys/v4l2/tunernorm.h:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	* sys/waveform/gstwaveformplugin.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/waveform/gstwaveformsink.h:
	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	* sys/ximage/ximageutil.c:
	* sys/ximage/ximageutil.h:
	* tests/check/elements/aacparse.c:
	* tests/check/elements/ac3parse.c:
	* tests/check/elements/alphacolor.c:
	* tests/check/elements/amrparse.c:
	* tests/check/elements/apev2mux.c:
	* tests/check/elements/aspectratiocrop.c:
	* tests/check/elements/audioamplify.c:
	* tests/check/elements/audiodynamic.c:
	* tests/check/elements/audioecho.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/audiopanorama.c:
	* tests/check/elements/autodetect.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/avisubtitle.c:
	* tests/check/elements/capssetter.c:
	* tests/check/elements/deinterlace.c:
	* tests/check/elements/deinterleave.c:
	* tests/check/elements/flacparse.c:
	* tests/check/elements/flvdemux.c:
	* tests/check/elements/flvmux.c:
	* tests/check/elements/gdkpixbufsink.c:
	* tests/check/elements/icydemux.c:
	* tests/check/elements/id3demux.c:
	* tests/check/elements/id3v2mux.c:
	* tests/check/elements/imagefreeze.c:
	* tests/check/elements/interleave.c:
	* tests/check/elements/jpegdec.c:
	* tests/check/elements/jpegenc.c:
	* tests/check/elements/level.c:
	* tests/check/elements/matroskamux.c:
	* tests/check/elements/matroskaparse.c:
	* tests/check/elements/mpegaudioparse.c:
	* tests/check/elements/multifile.c:
	* tests/check/elements/parser.c:
	* tests/check/elements/parser.h:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtpbin.c:
	* tests/check/elements/rtpbin_buffer_list.c:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/shapewipe.c:
	* tests/check/elements/souphttpsrc.c:
	* tests/check/elements/spectrum.c:
	* tests/check/elements/sunaudio.c:
	* tests/check/elements/udpsink.c:
	* tests/check/elements/udpsrc.c:
	* tests/check/elements/videocrop.c:
	* tests/check/elements/videofilter.c:
	* tests/check/elements/vp8dec.c:
	* tests/check/elements/vp8enc.c:
	* tests/check/elements/wavpackdec.c:
	* tests/check/elements/wavpackenc.c:
	* tests/check/elements/wavpackparse.c:
	* tests/check/elements/y4menc.c:
	* tests/check/generic/states.c:
	* tests/check/pipelines/effectv.c:
	* tests/check/pipelines/flacdec.c:
	* tests/check/pipelines/simple-launch-lines.c:
	* tests/check/pipelines/tagschecking.c:
	* tests/check/pipelines/wavenc.c:
	* tests/check/pipelines/wavpack.c:
	* tests/examples/audiofx/firfilter-example.c:
	* tests/examples/audiofx/iirfilter-example.c:
	* tests/examples/cairo/cairo_overlay.c:
	* tests/examples/level/level-example.c:
	* tests/examples/pulse/pulse.c:
	* tests/examples/rtp/client-PCMA.c:
	* tests/examples/rtp/server-alsasrc-PCMA.c:
	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	* tests/examples/spectrum/spectrum-example.c:
	* tests/examples/v4l2/camctrl.c:
	* tests/icles/equalizer-test.c:
	* tests/icles/gdkpixbufsink-test.c:
	* tests/icles/test-oss4.c:
	* tests/icles/v4l2src-test.c:
	* tests/icles/videobox-test.c:
	* tests/icles/videocrop-test.c:
	* tests/icles/videocrop2-test.c:
	* tests/icles/ximagesrc-test.c:
	  Fix FSF address
	  https://bugzilla.gnome.org/show_bug.cgi?id=687520

2012-11-03 20:40:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ext/twolame/gsttwolamemp2enc.c:
	* ext/twolame/gsttwolamemp2enc.h:
	  Fix FSF address
	  https://bugzilla.gnome.org/show_bug.cgi?id=687520

2012-11-03 20:40:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ext/lame/gstlamemp3enc.c:
	* ext/lame/gstlamemp3enc.h:
	* ext/lame/plugin.c:
	* tests/check/pipelines/lame.c:
	  Fix FSF address
	  https://bugzilla.gnome.org/show_bug.cgi?id=687520

2012-11-02 18:47:26 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  vrawdepay: don't access rtp buffer after unmap
	  Read the marker bit before we unmap the rtp packet.

2012-11-02 09:34:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Immediately return if opening the decoder failed
	  Instead of ignoring any errors.

2012-11-01 22:02:39 +0100  Debarshi Ray <rishi@gnu.org>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Short circuit gst_vp8_dec_handle_frame if keyframe is missing
	  https://bugzilla.gnome.org/show_bug.cgi?id=687376

2012-11-02 10:53:57 +1300  Douglas Bagnall <douglas@paradise.net.nz>

	* gst/videomixer/blend.c:
	  videoconvert: Compare y offset with height, not width, when testing for overlap
	  This could have prevented images showing that should have when the
	  source height is greater than its width.
	  When width exceeds height, as is common, it probably only caused a
	  miniscule amount of unnecessary work.  I haven't tested.

2012-11-01 21:09:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp8pay.h:
	  rtpvp8: include config.h and minor style fixes

2012-11-01 20:13:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/Makefile.am:
	  rtp: fix tabs/space mess in Makefile.am

2012-11-01 20:05:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpvp8.c:
	  rtp: move VP8 payloader and depayloader from -bad
	  Spec is still in draft state, but should hopefully not
	  change much now. Besides, we announce things as VP8-DRAFT-IETF-01
	  in our caps, so even if things change in incompatible ways it
	  should not break anything.
	  https://bugzilla.gnome.org/show_bug.cgi?id=687263

2012-10-17 17:34:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8: use gst_element_class_set_static_metadata()
	  where possible. Avoids some string copies. Also re-indent
	  some stuff. Also some indent fixes here and there.

2012-09-14 17:08:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata

2012-04-05 18:02:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpvp8.c:
	  rtpvp8: update for GST_PLUGIN_DEFINE() API changes

2012-03-28 12:49:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8: update for buffer changes

2012-03-01 14:59:55 -0300  Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8; fix compatibility with the third draft
	  https://bugzilla.gnome.org/show_bug.cgi?id=671073

2012-01-25 16:20:41 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8: port some more to new memory API

2012-01-25 10:45:51 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp8pay.h:
	  rtpvp8: port to 0.11

2011-10-03 12:06:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8pay: Fix typo

2011-09-23 22:58:30 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp8pay.h:
	  rtpvp8: Update the pay/depay to the ietf-draft-01 spec

2011-09-10 11:31:20 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/dboolhuff.c:
	* gst/rtp/dboolhuff.h:
	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8: fix bitstream parsing using the wrong kind of bitreader
	  VP8 uses a probabilistic bool coder, not a straight bit coder.
	  This fixes parsing when error-resilient is set.
	  This commit includes a copy of libvpx's bool coder, BSD licensed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=652694

2011-07-12 18:03:53 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8: Reject unknown bitstream versions

2011-03-04 11:59:44 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8: Fix unitialized variable
	  Makes macosx compiler happy.

2011-01-23 17:02:38 +0000  Sjoerd Simons <sjoerd@luon.net>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpvp8depay: Accept packets with only one byte of data
	  When fragmenting partions it can happen that an RTP packet only caries 1
	  byte of RTP data.

2011-01-23 16:42:17 +0000  Sjoerd Simons <sjoerd@luon.net>

	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp8pay.h:
	  rtpvp8pay: Treat the frame header just like any other partition
	  When setting up the initial mapping just act as if the global frame
	  information is another partition. This saves special-casing it later in
	  the actual packetizing code.

2010-05-16 17:23:17 +0100  Sjoerd Simons <sjoerd@luon.net>

	* gst/rtp/dboolhuff.LICENSE:
	* gst/rtp/gstrtpvp8.c:
	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8depay.h:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvp8pay.h:
	  rtpvp8: Add simple payloaders and depayloaders for VP8
	  Minimal implementation of http://www.webmproject.org/code/specs/rtp/,
	  version 0.3.2

2012-11-01 18:42:39 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  gstpay: fix for 1.0 events
	  Caps events are sometimes not followed by a buffer but by an event. Flush any
	  pending caps before we make a packet with the event.
	  Chain up to the parent event handler before we attempt to push RTP packets, it
	  might be a segment event.

2012-11-01 18:42:24 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  gstdepay: fix small leak

2012-11-01 17:44:11 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  gstdepay: add support for events
	  Conflicts:
	  gst/rtp/gstrtpgstdepay.c

2012-11-01 17:40:31 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  rtpgstpay: add support for sending events
	  We currently only send tags and custom events. The other events
	  might interfere with the receiver timings or are otherwise handled
	  by RTP.
	  Conflicts:
	  gst/rtp/gstrtpgstpay.c

2012-11-01 15:54:58 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  gstpay: rewrite payloader
	  Use adapter to assemble the payload and make a flush function to
	  turn this payload into (fragmented) packets.
	  Conflicts:
	  gst/rtp/gstrtpgstpay.c
	  gst/rtp/gstrtpgstpay.h

2012-11-01 13:03:44 +0000  Douglas Bagnall <douglas@paradise.net.nz>

	* gst/videomixer/blend.c:
	  videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH
	  https://bugzilla.gnome.org/show_bug.cgi?id=687330

2012-11-01 13:02:16 +0000  Douglas Bagnall <douglas@paradise.net.nz>

	* gst/videobox/gstvideobox.c:
	  videbox: fix border filling for gray formats
	  Get the height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH.
	  https://bugzilla.gnome.org/show_bug.cgi?id=687330

2012-11-01 11:58:57 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  gstdepay: check for correct fragment offset
	  Make sure we only insert the rtp packet in the adapter when the
	  frag_offset matches. When the first packet of a fragment is dropped,
	  it avoids putting the remaining packets in the adapter and processing
	  the partial fragment.
	  Conflicts:
	  gst/rtp/gstrtpgstdepay.c

2012-11-01 11:54:50 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  gstpay: set C flag on all buffers of the fragment
	  Set the C flags on all the fragments instead of only those with
	  caps in them. This makes it easier in the receiver to check if there
	  is a caps in the assembled fragments just by looking at the last RTP
	  packet flags.

2012-11-01 10:55:03 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  gstdepay: use the capsversion
	  Take the caps from the input caps and store it in the slot given
	  by capsversion.

2012-11-01 10:52:25 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  gstpay: send caps inline
	  Place the capsversion on the outgoing caps so that they end up in
	  an SDP as well. Receivers need to know what capsversion a particular
	  caps is for to be able to match the caps to the CV in the RTP packets.
	  Place the caps inside the RTP packet whenever the caps change.
	  Based on patch by Andrzej Bieniek <andrzej.bieniek@pure.com>
	  Conflicts:
	  gst/rtp/gstrtpgstpay.c
	  gst/rtp/gstrtpgstpay.h

2012-10-31 16:17:48 +0000  Andrzej Bieniek <andrzej.bieniek@pure.com>

	* gst/rtp/gstrtpgstpay.c:
	  gstpay: add debug
	  Conflicts:
	  gst/rtp/gstrtpgstpay.c

2012-10-31 16:09:26 +0000  Andrzej Bieniek <andrzej.bieniek@pure.com>

	* gst/rtp/gstrtpgstdepay.c:
	  depay: correctly skip caps header size
	  Conflicts:
	  gst/rtp/gstrtpgstdepay.c

2012-09-28 00:43:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: put streamheaders on vorbis/speex/flac/theora caps to make remuxing work
	  https://bugzilla.gnome.org/show_bug.cgi?id=640589

2012-10-28 00:07:46 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: don't assert in get_time() when called after shutdown
	  Which might happen if the source gets set to NULL state before
	  the rest of the pipeline.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686985

2012-10-30 11:10:49 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/level/level-example.c:
	  tests: fix level example
	  Use the GValueArray in the message.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=687154

2012-10-30 09:27:24 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: removed unnecessary finalize function
	  https://bugzilla.gnome.org/show_bug.cgi?id=687176

2012-10-30 10:20:09 +1100  Jan Schmidt <thaytan@noraisin.net>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: Fix leaks from not chaining up in the finalize function

2012-10-27 23:22:36 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/auparse/Makefile.am:
	* gst/level/Makefile.am:
	* gst/y4m/Makefile.am:
	  gst: fix variable order in some Makefile.am
	  https://bugzilla.gnome.org/show_bug.cgi?id=687013

2012-10-27 17:27:16 -0400  Antoine Tremblay <hexa00@gmail.com>

	* ext/libcaca/Makefile.am:
	* gst/auparse/Makefile.am:
	* gst/level/Makefile.am:
	* gst/videocrop/Makefile.am:
	* gst/y4m/Makefile.am:
	  gst: add various missing GST_PLUGINS_BASE_LIBS in Makefile.am
	  Those plugins depend on either libgstaudio or libgstvideo,
	  which are in gst-plugins-base.
	  https://bugzilla.gnome.org/show_bug.cgi?id=687013

2012-10-27 13:24:24 +0100  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: mark invisible VP8 frames with the DECODE_ONLY flag
	  https://bugzilla.gnome.org/show_bug.cgi?id=654259

2012-10-26 10:55:28 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/multifile.c:
	  tests: add multifilesrc test for fix in previous commit
	  Make sure the stop-index set is honoured.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654853

2012-10-26 10:33:03 +0100  Stas Sergeev <stsp@aknet.ru>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: fix stop index handling
	  Make sure the stop index is always honoured. Avoids
	  endless loop if one wants to read and output the same
	  file N times, for example.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654853

2012-08-25 02:26:29 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: Support recursive SimpleTags
	  Fixes #682644
	  Depends on #682615

2012-08-24 13:55:41 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-read-common.c:
	  matroskademux: Expand the tag mapping.
	  * Also expose unknown tags as key=value pairs.
	  * Arrange tag map in the same order tags are listed in Matroska spec, leaving
	  unmapped tags as comments.
	  * More specific TODOs.
	  * Remove duplicate DATE define.
	  Fixes #682615
	  Depends on #682524

2012-10-26 10:09:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: Fix uninitialized variable compiler warning

2012-08-23 15:07:22 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-read-common.c:
	  matroskademux: Matroska tag TargetType support
	  * Reads TargetType and TargetTypeValue from a Tag.
	  * After Tag is completely read, processes taglist, substituting some of the
	  tags depending on target type value and the presence of video/subtitle streams.
	  * Supports reading two new simpletags - PART_NUMBER and TOTAL_PARTS
	  Depends on #682448
	  Fixes #682524

2012-08-22 15:32:41 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-read-common.c:
	  matroskademux: Per-track tags for Matroska
	  Requires Matroska file to have sane layout (track info before tag info).
	  Uses replace-merge.
	  Makes track UIDs 64-bit.
	  Fixes #682448

2012-10-25 20:18:36 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: fix typo in property description

2012-10-25 12:18:03 -0700  Michael Smith <msmith@rdio.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: read video format header fully (so we can find 'pasp' atoms) for more fourccs. Fixes aspect ratio of prores files.

2012-10-25 00:44:34 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: the new get_caps already does the filter intersection
	  It should be faster to pass the caps to intersect as the filter caps,
	  rather than using NULL and intersecting 'manually' later.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686837

2012-10-25 00:43:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: avoid assertion when using accept caps query
	  This query must receive a fixed caps, so imagefreeze should
	  fixate its framerate before sending the query downstream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686837

2012-10-25 12:33:24 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to feature development

=== release 1.0.2 ===

2012-10-25 01:01:09 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.0.2

2012-10-24 13:41:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/mpg123audiodec.c:
	  tests: fix up mpg123 test a little
	  - dist input files
	  - fix sample leak
	  - simplify check for elements
	  - only run mpg123 test if mpg123 is available and selected
	  - fix build in uninstalled setup
	  https://bugzilla.gnome.org/show_bug.cgi?id=686595

2012-10-24 12:30:10 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* tests/check/elements/mpg123audiodec.c:
	  tets: add unit test for mpg123audiodec
	  https://bugzilla.gnome.org/show_bug.cgi?id=686595

2012-10-24 00:36:42 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: added gtkdoc section
	  https://bugzilla.gnome.org/show_bug.cgi?id=686595

2012-10-24 00:22:05 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: fixed bug with last frame, disabled internal resampler & chatter
	  * The last MP3 frame wasn't being pushed when base class was draining
	  * Made sure mpg123 cannot ever use its (crude) internal resampler
	  * Disabled mpg123 stderr output
	  https://bugzilla.gnome.org/show_bug.cgi?id=686595

2012-10-24 13:50:00 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: use correct type for channel-mask bitmask
	  Fixes crash on 32-bit systems.

2012-10-24 00:21:45 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: cleaned up comments, formatting, and logging lines
	  also replaced mpg123decoder->handle != NULL checks with asserts
	  https://bugzilla.gnome.org/show_bug.cgi?id=686595

2012-10-24 11:17:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Flush the ringbuffer on GAP events without duration
	  This is required to properly start the ringbuffer and clock.

2012-10-02 20:51:29 +0200  Oleksij Rempel <bug-track@fisher-privat.net>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: set DECODE_ONLY flag on invisible AltRef frames
	  https://bugzilla.gnome.org/show_bug.cgi?id=654216

2012-10-23 16:02:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix coverart extraction if vorbis comments come after picture header
	  See sample file for bug #684701.

2012-10-23 13:45:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: ignore bad headers if we have a valid STREAMINFO header
	  If we run into any header parsing issues and we have a valid
	  STREAMINFO header already, don't error out, but just stop
	  header parsing and try to find some audio frames.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684701

2012-10-23 13:43:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: post proper error message and fix buffer leak on header parsing error
	  https://bugzilla.gnome.org/show_bug.cgi?id=684701

2012-10-22 22:32:49 -0700  Michael Smith <msmith@rdio.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: with raw audio, set a default channel-mask for multichannel audio. This doesn't actually parse 'chan' because it's absurdly complex.

2012-10-22 15:54:17 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* gst/udp/gstudpsrc.c:
	  updsrc: fix typo causing compilation error
	  gstudpsrc.c: In function 'gst_udpsrc_create':
	  gstudpsrc.c:365: error: 'ret' may be used uninitialized in this function
	  https://bugzilla.gnome.org/show_bug.cgi?id=686642

2012-10-22 11:55:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi_ fix invert function
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686550

2012-10-22 11:55:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: fix debug

2012-10-22 11:39:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: add support for 'generic' samples
	  Add support for stuffing a complete stream into 1 sample.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=686550

2012-10-20 13:01:41 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/souphttpsrc.c:
	  tests: remove superfluous g_type_init() call
	  It's deprecated in newer GLib and not needed here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686456

2012-10-20 11:32:27 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix caps leak in acceptcaps function

2012-10-19 19:24:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't leak gst_riff_strf_auds in case of MS/RIFF audio
	  https://bugzilla.gnome.org/show_bug.cgi?id=681192

2012-10-18 22:20:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: unsigned subtitle template

2012-10-18 11:32:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: in accept_caps() check if ring buffer is NULL before de-referencing
	  And sprinkle some thread-safety (take object lock for
	  accessing ring buffer, and pa main loop lock for the
	  context).
	  https://bugzilla.gnome.org/show_bug.cgi?id=683782

2012-09-13 00:10:00 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer2: Fix race condition where a src setcaps is ignored
	  If both pads receive data at the same time, they will both get their
	  sink_setcaps called which will call the src_setcaps, but there is
	  a race condition where the second one might not be called.
	  Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=683842

2011-10-31 15:43:25 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: do not use unoffical V_MJPEG codec id
	  Since it's not spec'ed, consider it a VfW compatibility
	  case. Many applications (e.g. avidemux) don't understand
	  the unofficial V_MJPEG id.
	  Fixes #659837.
	  Conflicts:
	  gst/matroska/matroska-mux.c

2012-10-17 17:34:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audiofx/gststereo.c:
	  Use gst_element_class_set_static_metadata()
	  where possible. Avoids some string copies. Also re-indent
	  some stuff. Also some indent fixes here and there.

2012-10-17 17:34:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  Use gst_element_class_set_static_metadata()
	  where possible. Avoids some string copies. Also re-indent
	  some stuff. Also some indent fixes here and there.

2012-10-17 17:03:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8enc.c:
	  jpeg, png, vpx: use gst_element_class_set_static_metadata()
	  Avoids some string copies.

2012-10-17 14:23:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	  jpegdepay: store quant tables in zigzag order

2012-10-17 13:55:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtsession: fix compiler warning

2012-10-17 13:35:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: clarify the ntp-sync option

2012-10-17 13:15:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: update caps in the source
	  Inform the source when caps changed. This was removed in the port to 1.0
	  leaving the source unaware of the clock-rate and unable to interpollate
	  rtp timestamps for SR packets.

2012-10-17 12:46:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpbin: set PTS and DTS in jitterbufffer

2012-10-17 12:24:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: disable check for ntp-sync
	  Disable the check for the ntp-sync method. It is expected that
	  a rather larger offset needs to be applied with this method.

2012-10-17 12:17:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin: use running-time for NTP time
	  When use-pipeline-clock is set, use the running-time of the
	  pipeline to calculate the NTP timestamps. This method would previously
	  only work when the base-time is set to 0 but with this change it can
	  also work with different offsets and we can also implement pause/resume
	  of the sender and receiver now.

2012-10-17 10:20:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	  videocrop: port to videofilter

2012-10-17 09:36:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: use out_info for out properties

2012-10-16 14:40:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videofilter/gstvideomedian.c:
	* gst/videofilter/gstvideomedian.h:
	  median: small cleanups

2012-10-16 13:56:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* Makefile.am:
	* gst/median/.gitignore:
	* gst/median/Makefile.am:
	* gst/median/gstmedian.c:
	* gst/median/gstmedian.h:
	* gst/median/median.vcproj:
	  median: remove now that it is in videofilter

2012-10-16 13:49:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  configure: remove median from build

2012-10-16 13:47:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstvideomedian.c:
	* gst/videofilter/gstvideomedian.h:
	* gst/videofilter/plugin.c:
	  videomedian: copy media to videomedian
	  Copy the median video filter to videofilters and rename to
	  videomedian.

2012-10-16 13:12:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/median/Makefile.am:
	* gst/median/gstmedian.c:
	* gst/median/gstmedian.h:
	  media: port to 1.0

2012-10-16 01:02:11 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: append palette data to paletted 8-bit RGB frames
	  Fixes playback of 8-bit indexed RGB videos, with fixes in -base.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686046

2012-10-15 15:36:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: And this time fix the default target-bitrate value for real

2012-10-15 15:30:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Fix default target-bitrate value

2012-10-13 00:03:29 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't assert if upstream size is not available when guessing bitrates
	  Fixes abort in push mode where the source is not seekable and the
	  size of the file is not available, as with
	  cat foo.mp4 | gst-launch-1.0 playbin uri=fd://0
	  Less noticable with releases, since we disable all
	  g_assert() there.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686008

2012-10-12 14:38:33 -0700  Michael Smith <msmith@rdio.com>

	* gst/isomp4/qtdemux.h:
	  qtdemux: allow more streams. Bump this constant to 32, which should be enough for real-world files.

2012-10-12 14:35:24 -0700  Michael Smith <msmith@rdio.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: support more different fourcc values for other ProRes variants.

2012-10-11 22:36:21 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/examples/rtp/client-H263p-AMR.sh:
	* tests/examples/rtp/client-H263p-PCMA.sh:
	* tests/examples/rtp/client-H263p.sh:
	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-H264.sh:
	* tests/examples/rtp/client-PCMA.c:
	* tests/examples/rtp/client-PCMA.sh:
	* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
	* tests/examples/rtp/server-VTS-H263p.sh:
	* tests/examples/rtp/server-alsasrc-PCMA.sh:
	* tests/examples/rtp/server-decodebin-H263p-AMR.sh:
	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  examples: update some element names for 1.0 in RTP examples
	  gstrtpbin -> rtpbin
	  ffdec_*   -> avdec_*
	  ffenc_*   -> avenc_*

2012-10-10 12:05:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove unused include

2012-10-10 10:55:28 +0200  Rasmus Rohde <rohde@duff.dk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: add multicast-iface property
	  udpsrc already has support for setting the multicast interface, which
	  is useful for multi-homed machines. This patch adds the same code to
	  the multiudpsink.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685864

2012-10-10 11:32:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: don't error on send errors but only warn
	  Don't error on send errors but simply post a warning, it's possible
	  that the next packet will be fine.

2012-10-10 10:28:24 +0200  Rasmus Rohde <rohde@duff.dk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: add force-ipv4 option
	  Add an option to the multiudpsink that makes it possible to force
	  the use of an IPv4 socket.
	  This can e.g. be used to handle the issue described in
	  https://bugzilla.gnome.org/show_bug.cgi?id=682481

2012-10-10 10:18:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: remove unused field

2012-10-10 10:10:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: use negotiated allocator or pool
	  Use the base class to allocate a buffer for us because it knows how
	  to use the negotiated allocator or bufferpool.

2012-10-10 10:09:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: post error when something goes wrong

2012-10-10 10:09:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	  spectrum: elements post element messages

2012-10-07 16:56:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development (bug fixing)

=== release 1.0.1 ===

2012-10-07 15:31:12 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.0.1

2012-10-06 14:57:10 +0100  Tim-Philipp Müller <tim@centricular.net>

	* common:
	  Automatic update of common submodule
	  From 6c0b52c to 6bb6951

2012-10-05 15:12:27 -0700  Michael Smith <msmith@rdio.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: output channels should be marked as MONO, not FRONT_LEFT, if we're not preserving input channel positions.

2012-10-04 15:13:20 -0700  Michael Smith <msmith@rdio.com>

	* gst/interleave/interleave.c:
	  interleave: use gst_audio_channel_positions_to_mask instead of a local copy of half of it. Handles some values more correctly.

2012-10-04 20:32:45 +0200  Rasmus Rohde <rohde@duff.dk>

	* gst/rtp/gstrtpgstdepay.c:
	  gstrtpdepay: don't leak input buffer
	  The rtp buffer is never unmapped in the normal code exit path
	  of gst_rtp_gst_depay_process(..) resulting in a memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=685512

2012-10-04 18:37:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Add support for NV12 and NV21

2012-10-01 15:11:05 +0200  Patricia Muscalu <patricia@axis.com>

	* gst/rtp/gstrtph264pay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtph264pay: do not push unmapped data
	  Also do not use a GstBuffer after it has been pushed into the adapter.
	  https://bugzilla.gnome.org/show_bug.cgi?id=685213

2012-10-03 10:51:45 -0700  Michael Smith <msmith@rdio.com>

	* gst/interleave/deinterleave.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/ximage/ximageutil.c:
	  meta info: threadsafe registration using g_once

2012-10-01 15:44:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: push mode; handle some initial junk before hdrl list
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685059

2012-10-01 14:03:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/icles/gdkpixbufsink-test.c:
	  tests: port gdkpixbufsink test

2012-09-29 11:59:31 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/level/gstlevel.c:
	* tests/check/elements/videocrop.c:
	  Purge references to liboil
	  https://bugzilla.gnome.org/show_bug.cgi?id=673285

2012-09-28 16:51:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/avi-ids.h:
	* gst/avi/gstavidemux.c:
	  avidemux: recognize all xsub frames as keyframes
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684977

2012-09-28 16:50:25 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: push mode: find the correct chunk for segment following seek
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684977

2012-09-27 22:17:49 +0100  Arnaud Vrac <rawoul@gmail.com>

	* gst/isomp4/qtdemux.h:
	  qtdemux: fix parsing in push mode when moov atom is at the end
	  When playing an mp4 file with the MOOV atom at the end of the file, playback
	  fails with the error message "no 'moov' atom within the first 10 MB". This is
	  due to a mistake in the upstream_size typing, making the seek to the end of
	  file never happening.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684972

2012-09-27 15:50:49 -0300  Andre Moreira Magalhaes (andrunko) <andre.magalhaes@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	  gamma: remove duplicate entries at format at caps
	  Avoids extra caps/structures processing

2012-09-27 14:13:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtpvrawdepay: negotiate pool with srcpad caps

2012-09-27 11:02:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	  dvdemux: The convert and duration queries are not supposed to change the format

2012-09-26 09:28:59 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/videomixer/videomixer2.c:
	  videomixer: clear video frame more correctly
	  Make sure not to touch memory that doesn't belong to
	  our frame, we might be one part of a side-by-side 3D
	  frame, or in a picture-in-picture scenario.

2012-09-26 00:44:59 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: minor clean-up
	  Use GstByteWriter, because we can, and g_value_take_boxed.

2012-09-10 10:27:28 +0400  Dmitriy Samonenko <dmitriy.samonenko@teligent.ru>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix speex audio decoding by creating fake stream header
	  https://bugzilla.gnome.org/show_bug.cgi?id=683622

2012-09-25 21:21:15 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/videomixer/videomixer2.c:
	* tests/check/pipelines/simple-launch-lines.c:
	  videomixer: fix warnings when using transparent background
	  gst_video_frame_map() increases the refcount, which makes
	  the buffer not writable any more technically, so calling
	  gst_buffer_memset() on it will cause nasty warnings.
	  Unit test disabled because it very rarely (for me)
	  fails, possibly negotiation-related.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684398

2012-09-25 10:43:28 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Add some useful debug logging

2012-09-25 10:41:44 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix telecine
	  This only affects behaviour in telecine cases with pattern locking
	  enabled. The default case should be untouched.
	  This works with the output from fieldanalysis at least, but the field
	  order looks swapped for telecine mixed buffers with the
	  David_slides_Schleef clip.

2012-09-25 14:43:15 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Disable GLIB deprecation warnings
	  GValueArray has been deprecated since 2.32 ... but there's no usable
	  replacement for it.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=667228

2012-09-25 14:18:35 +0200  Edward Hervey <edward@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Fix leak

2012-09-24 16:46:18 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development (bug fixing)

=== release 1.0.0 ===

2012-09-24 14:06:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.0.0

2012-09-24 11:56:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/rganalysis.c:
	  tests: remove g_printerr() that's not needed any longer
	  now that tcase_skip_broken_test() prints it as well.

2012-09-23 19:50:42 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/rganalysis.c:
	  tests: disable failing replaygain tests

2012-09-23 16:31:37 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmpte.h:
	  smpte: send stream-start event

2012-09-23 16:10:36 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartmux.c:
	* gst/multipart/multipartmux.h:
	  multipartmux: send stream-start event

2012-09-23 16:02:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: send stream-start

2012-09-23 15:57:35 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/isomp4/gstqtmux.c:
	  qtmux: send stream-start event

2012-09-23 15:48:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	  interleave: add a bunch of FIXMEs
	  Needs some more work, so stream-start, caps and tags are
	  sent in the right order.

2012-09-23 15:18:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/flv/gstflvmux.c:
	  flvmux: send stream-start event

2012-09-23 15:16:14 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/avi/gstavimux.c:
	  avimux: send stream-start event

2012-09-22 15:00:27 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  rtpdtmfdepay: Use 1.0-style caps negotiation and audio/x-raw

2012-09-22 16:08:05 +0100  Tim-Philipp Müller <tim@centricular.net>

	* common:
	  Automatic update of common submodule
	  From 4f962f7 to 6c0b52c

2012-09-21 21:54:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: answer URI query
	  Without this, something also answered the query
	  with TRUE but without setting a uri, not sure
	  what that was..

2012-09-20 17:28:47 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Make sure the caps don't have duplicated sps/pps

2012-09-20 19:58:12 +0200  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Mute stream post-connection if required
	  A bug in PulseAudio causes PA_STREAM_START_MUTED to be rejected on
	  record streams. Until this is fixed upstream, we mute the stream
	  manually at startup. Based on a patch by Alban Browaeys
	  <prahal@yahoo.com>.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684469

2012-09-20 18:00:59 -0700  Michael Smith <msmith@rdio.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: 24 bit audio here is S24LE, not S24_3LE.

2012-09-20 10:07:24 +0200  Sjoerd Simons <sjoerd@luon.net>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: handle latency query before setting up the bufferpool
	  Fixes crash if no bufferpool is set up yet.
	  https://bugzilla.gnome.org/show_bug.cgi?id=684430

2012-09-19 09:17:03 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: Specify endianness in IEC 61937 payloading
	  Corresponds to an API change in gst-plugins-base. This needs to be fixed
	  to query the expected byte order using appropriate API.
	  https://bugzilla.gnome.org/show_bug.cgi?id=678021

2012-09-19 09:15:53 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Specify endianness in IEC 61937 payloading
	  DirectSound expects native endian byte order.
	  https://bugzilla.gnome.org/show_bug.cgi?id=678021

2012-09-19 09:13:11 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Specify endianness in IEC 61937 payloading
	  Corresponds to an API change in gst-plugins-base.
	  https://bugzilla.gnome.org/show_bug.cgi?id=678021

2012-09-19 00:39:01 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Remove incorrect logic
	  I don't understand why these lines were added, they don't make sense to
	  me now and both David and I agree that removing them moves closer to
	  related logic being correct, therefore, they're being removed.
	  I've tested a few progressive, interlaced and telecine clips and they
	  all behave properly timestamp-wise and visually after these changes.

2012-09-19 00:17:49 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix field duration
	  The frame rate fraction is correctly adjusted in the cases preceding the
	  field duration calculation and so the factor of 2 is incorrect.

2012-09-18 10:34:03 -0700  Michael Smith <msmith@rdio.com>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix U/V strides for a number of cases.

2012-09-18 12:13:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: init videoinfo
	  ... to prevent random bogus caps fields.

2012-09-18 12:12:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: chain up to collectpads query function

2012-09-17 13:17:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't let GstCollectPad shadow custom sink pad query func
	  In the current implementation, the custom pad query function is not called.
	  This patch, set that query function on the GstCollectPads to avoid this
	  shadowing.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=684237

2012-09-17 18:23:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/files/Makefile.am:
	  tests: dist image.jpg for jpeg test

=== release 0.11.99 ===

2012-09-17 17:57:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.11.99

2012-09-17 16:57:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/twolame/Makefile.am:
	  Remove -DGST_USE_UNSTABLE_API

2012-09-17 16:57:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/Makefile.am:
	  Remove -DGST_USE_UNSTABLE_API

2012-09-17 16:53:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.types:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update

2012-09-17 13:30:15 +0200  Christian Fredrik Kalager Schaller <uraeus@linuxrisin.org>

	* gst-plugins-good.spec.in:
	  Fix spec file for vp8 move

2012-09-17 13:23:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* Makefile.am:
	  annodex: Add to the CRUFT_DIRS

2012-09-17 12:14:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	  docs: update

2012-09-17 09:48:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Correctly finish frames
	  Previously we would always get the same frame if multiple frames are pending,
	  leaking memory of the previous frames and breaking timestamps.

2012-09-17 09:40:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Allow changing bitrate and other parameters during playback
	  Fixes bug #648276.

2012-09-17 09:16:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	  vp8enc: Store configuration in the vpx_codec_enc_cfg_t struct instead of duplicating all variables
	  Also protect encoder with a mutex.

2012-09-16 16:03:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Update documentation to reflect new property names
	  ...and also link to the WebM encoder parameters website.

2012-09-16 15:57:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: Make some property names more readable

2012-09-16 15:47:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/.gitignore:
	  vp8: Add tests to .gitignore

2012-09-16 15:46:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/vp8enc.c:
	  vp8enc: Update patch to the new property names

2012-09-16 15:46:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/Makefile.am:
	  vpx: Integrate test into the build system too

2012-02-07 17:00:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/vp8dec.c:
	* tests/check/elements/vp8enc.c:
	  [MOVED FROM BAD 6/6] tests: fix more unit tests

2011-11-24 21:42:39 +0100  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/vp8dec.c:
	* tests/check/elements/vp8enc.c:
	  [MOVED FROM BAD 5/6] tests: update for gstcheck API change

2010-07-10 15:46:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/vp8dec.c:
	  [MOVED FROM BAD 4/6] vp8dec: Add simple unit test for vp8dec

2010-07-10 15:46:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/vp8enc.c:
	  [MOVED FROM BAD 3/6] vp8enc: Improve unit test a bit

2010-07-10 15:32:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/vp8enc.c:
	  [MOVED FROM BAD 2/6] vp8enc: Also check the output caps in the unit test

2010-07-10 15:29:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/vp8enc.c:
	  [MOVED FROM BAD 1/6] vp8enc: Add simple unit test

2012-09-16 15:43:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* ext/Makefile.am:
	  vpx: Integrate into the build system

2012-09-16 15:33:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vpx/GstVP8Enc.prs:
	* ext/vpx/Makefile.am:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	* ext/vpx/gstvp8utils.c:
	* ext/vpx/gstvp8utils.h:
	* ext/vpx/plugin.c:
	  vpx: Rename vp8 plugin to vpx
	  This is using libvpx, which can support more codecs than just VP8
	  and will likely support future codecs.

2012-09-16 15:32:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  vp8: Apply remaining changes that got lost while moving the plugin via git am thanks to merges

2012-09-16 15:25:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 134/134] vp8dec: Unref input/output states when stopping the decoder

2012-09-16 15:18:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/GstVP8Enc.prs:
	  [MOVED FROM BAD 133/134] vp8enc: Update realtime profile to the new properties

2012-09-16 10:56:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 132/134] vp8: Require latest libvpx release (1.1.0 from May 2012)
	  Fixes bug #684116 and simplifies configure checks.

2012-09-15 20:23:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 131/134] vp8enc: Use a string field for the profile in the caps
	  Just for consistency with all the other codecs.

2012-09-15 00:04:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 130/134] vp8enc: Correctly set profile in caps

2012-09-14 23:41:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 129/134] vp8: Update copyright and authors

2012-09-08 15:38:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 128/134] vp8enc: Rework encoder properties to be more in line with the libvpx tools and API
	  Also add all available properties.

2012-09-14 17:08:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 127/134] replace gst_element_class_set_details_simple with gst_element_class_set_metadata

2012-07-19 09:05:28 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 126/134] vp8dec: Call gst_video_decoder_negotiate()

2012-08-14 11:17:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8dec.h:
	  [MOVED FROM BAD 125/134] vp8dec: Add support for multiple decoding threads

2012-08-14 11:09:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 124/134] vp8dec: Add support for the MFQE postprocessing flag
	  Which is enabled by default if postprocessing is enabled.

2012-08-09 13:37:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/Makefile.am:
	  [MOVED FROM BAD 123/134] vp8: Use pkg-config file for getting the LIBS and CFLAGS

2012-08-08 17:06:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 122/134] vp8enc: Update the per-component strides for every frame too
	  This is necessary because of GstVideoAlignment

2012-07-26 19:31:14 +0200  Oleksij Rempel <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 121/134] vp8enc: initiate encoder to fix a crash.
	  Without this patch vp8enc send header before and after first
	  key frame. On second keyframe vp8dec will crash without getting
	  decoded frame. With this pipe it is easy to reproduce this issue:
	  gst-launch-1.0 videotestsrc ! vp8enc ! vp8dec ! fakesink
	  https://bugzilla.gnome.org/show_bug.cgi?id=680667

2012-07-28 00:32:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 120/134] tag: Update for taglist/tag event API changes

2012-07-23 10:35:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 119/134] ext: Update for video base classes API changes

2012-07-21 19:59:21 +0200  Oleksij Rempel <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 118/134] vp8enc: fix memory leak
	  unref frame. i hope it is correct place to do it.
	  Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>

2012-07-06 11:50:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 117/134] update for query api changes

2012-07-06 11:26:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 116/134] update for query api changes

2012-07-06 11:03:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 115/134] update for allocation query changes

2012-06-07 12:33:31 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 114/134] vp8: fix codec state leaks
	  I only tested that vp8enc ! vp8dec does not crash, as valgrind does not grok
	  at least one of the instructions used by vp8enc, preventing me from checking
	  a leak, and the lack of one after the patch.

2012-06-06 13:02:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 113/134] update for tag event change

2012-05-28 16:05:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 112/134] vp8: Port to 0.11 again

2012-05-18 12:46:55 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 111/134] vp8enc: fix target bitrate config with libvpx 1.1.0
	  libvpx 1.1.0 disallows a bitrate of 0, which was used by
	  vp8enc as a default value.
	  Instead, we use the default libvpx bitrate, scaled to our
	  video size, if no bitrate was specified.
	  This fixes encoding VP8 video with libvpx 1.1.0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=676245

2012-05-16 14:04:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 110/134] vp8enc: Update for GstVideoCodecFrame API changes

2012-04-27 18:22:42 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8dec.h:
	  [MOVED FROM BAD 109/134] vp8dec: Improve output_state handling
	  Avoid getting output_state for every buffer as that requires
	  getting the objectlock and doing reference counting. Store it locally
	  when it is created and use it.

2012-04-27 09:05:57 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 108/134] vp8dec: Use outputstate when copying output buffer data
	  Using the input state was causing a crash because the strides/offsets
	  would be wrong. Fix it by using the output as we are dealing with
	  the decoded frame.

2012-04-24 11:08:41 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 107/134] vp8: Port to -base video base classes
	  Conflicts:
	  ext/vp8/Makefile.am
	  ext/vp8/gstvp8dec.c
	  ext/vp8/gstvp8enc.c
	  Back to 0.10 state for now, need to be ported again.

2012-05-18 12:46:55 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 106/134] vp8enc: fix target bitrate config with libvpx 1.1.0
	  libvpx 1.1.0 disallows a bitrate of 0, which was used by
	  vp8enc as a default value.
	  Instead, we use the default libvpx bitrate, scaled to our
	  video size, if no bitrate was specified.
	  This fixes encoding VP8 video with libvpx 1.1.0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=676245

2012-04-05 18:02:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/plugin.c:
	  [MOVED FROM BAD 105/134] gst: Update for GST_PLUGIN_DEFINE() API changes

2012-04-04 14:41:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/Makefile.am:
	  [MOVED FROM BAD 104/134] gst: Update versioning

2012-03-06 15:21:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 103/134] vp8enc: Fix 'argument to 'sizeof' in 'memset' call is the same expression as the destination' compiler warning

2012-01-30 17:17:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 102/134] update for HEADER flag

2012-01-25 18:49:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 101/134] port some more to new memory API
	  Fixes #668677.

2012-01-24 11:22:46 +0100  Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 100/134] vp8enc: trace outgoing timestamps
	  add info level prints for outgoing timestamps.
	  Signed-off-by: Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

2012-01-04 11:05:48 +0100  Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 099/134] vp8dec: use is_alt_data option to prevent timestamp collisions
	  altref/invisible frames usually stored in container with same timestamp as
	  dependet frame. This make basevideodecoder to update timestamp for dependet
	  frame and couse TS colision on next frame:
	  ^- here is altref
	  time     : 1 2 3 4 5 6 7 8 9
	  webm ts  : 1   3 5 5   7   9
	  vp8dec ts: 1   3   7   7   9
	  Fix bug: https://bugzilla.gnome.org/show_bug.cgi?id=655245
	  Signed-off-by: Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

2012-01-02 08:28:13 +0100  Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

	* ext/vp8/GstVP8Enc.prs:
	* ext/vp8/Makefile.am:
	  [MOVED FROM BAD 098/134] vp8: add initial preset file
	  This is initial preset file, currently with only one profile
	  for realtime encoding.
	  Signed-off-by: Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

2011-11-28 13:08:27 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 097/134] various: fix pad template ref leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=662664

2011-11-25 11:36:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 096/134] vp8dec: use new basevideodecoder API to drop frames and get QoS messages posted

2011-11-10 15:13:34 +0200  Mart Raudsepp <leio@gentoo.org>

	* ext/vp8/Makefile.am:
	  [MOVED FROM BAD 095/134] mimic, opencv, vp8, acmmp3dec, linsys: Don't build static plugins
	  Pass --tag=disable-static to libtool everywhere where it's been forgotten
	  https://bugzilla.gnome.org/show_bug.cgi?id=663768

2011-11-03 14:01:41 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 094/134] vp8: Port to 0.11

2011-08-21 20:15:25 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 093/134] vp8enc: fix drop-frame property
	  Fixes #656929.

2011-08-19 19:17:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 092/134] vp8: probe for the new tuning API to keep building with older libvpx
	  https://bugzilla.gnome.org/show_bug.cgi?id=656928

2011-08-18 10:39:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 091/134] vp8enc: Remove unused and useless variable in tags handling

2011-08-12 12:08:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 090/134] vp8enc: Update for basevideoencoder ::get_caps() removal

2011-07-09 18:53:24 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 089/134] vp8enc: Add more properties

2011-06-19 16:06:46 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 088/134] vp8enc: add min/maxsection-pct option
	  This options should be good to redeuce decode CPU load.
	  for lowend hardware:
	  minsection-pct=15 maxsection-pct=400
	  for hiend hw:
	  minsection-pct=5 maxsection-pct=800
	  see example:
	  http://www.webmproject.org/tools/encoder-parameters/#2-pass_vbr_encoding_for_smooth_playback_on_low-end_hardware
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
	  Signed-off-by: David Schleef <ds@schleef.org>

2011-06-19 11:05:36 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 087/134] vp8enc: add lag-in-frames option.
	  This option set maximum of frames codec should remember,
	  to make better prediktion for alt-ref frames.
	  See example:
	  http://www.webmproject.org/tools/encoder-parameters/#2-pass_best_quality_vbr_encoding
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
	  Signed-off-by: David Schleef <ds@schleef.org>

2011-06-19 07:16:57 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 086/134] vp8enc: use multipass.cache file name as default for multipass mode.
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
	  Signed-off-by: David Schleef <ds@schleef.org>

2011-07-21 08:03:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 085/134] vp8enc: Update for GstBaseVideoEncoder::finish() signature change

2011-07-12 18:05:25 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 084/134] vp8: Fix set-but-unused warnings

2011-07-09 11:31:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 083/134] vp8enc: Use destroy notify to free the coder hook

2011-06-18 15:56:49 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 082/134] vp8enc: update for new libvpx api

2011-06-26 15:15:54 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 081/134] vp8enc: generate a timestamp for alt-ref frames.
	  It will fix handling of altref/invisible frames since matroska-mux
	  drop any fram with no timestamp.
	  see also:
	  http://www.webmproject.org/code/specs/container/
	  The encoder will currently set the AR's timestamp as close as possible
	  to the previous frame while attempting to provide a timestamp that is
	  strictly increasing. In cases where the time base given to the encoder
	  at configure time is not granular enough to allow for this the AR
	  will share the same timestamp as D, but should be
	  treated as having no duration.
	  Fixes bug #652951
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>

2011-06-18 17:47:36 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 080/134] vp8dec: add check if we have legal aspect-ratio before reset it.
	  the commit f9b552f0494e (vp8dec: set par to 1/1)
	  will fix situation where no aspect-ratio is set, but it brake
	  stream with available aspect-ratio. This patch fix it.
	  Fixes: #652902.
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>

2011-06-03 19:36:59 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 079/134] vp8dec: set par to 1/1

2011-05-18 13:27:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 078/134] vp8enc: Name max/min quantizer properties {max,min}-quantizer
	  Also improve quality property description.

2011-05-18 13:26:23 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 077/134] vp8enc: Add properties to select a maximum and minimum quantizer
	  Fixes bug #641405.

2011-05-18 13:18:58 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 076/134] vp8enc: Fix quality to (constant) quantizer mapping
	  This now allows to select all possible quantizers between
	  0 and 63.
	  See bug #641405.

2011-04-01 22:13:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 075/134] vp8dec: debug code style fixes

2011-04-01 22:13:00 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 074/134] vp8dec: propagate downstream flow return to upstream

2011-03-30 10:18:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 073/134] basevideodecoder: really and only set src pad caps whenever requested
	  ... since subclass is expected to be wise enough to know when to do so.

2011-03-29 10:41:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 072/134] basevideodecoder: invoke subclass start method at state change and use set_format
	  While this changes API slightly (e.g. actually uses set_format now), which is OK
	  for unstable API, it has following merits:
	  * symmetric w.r.t. stop at state change
	  * in line with other base class practice
	  * otherwise no subclass method at state change (global activation time)
	  Moreover, subclassese are either unaffected or trivially adjusted accordingly.

2011-03-28 08:59:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 071/134] basevideodecoder: subsume skip_frame into finish_frame

2011-03-24 14:10:07 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 070/134] basevideoencoder: provide proper upstream flow return handling

2011-03-24 13:59:35 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 069/134] vp8enc: minor optimization in setting up image buffer

2011-03-24 12:50:23 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 068/134] vp8enc: refactor frame processing

2011-03-24 11:55:41 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 067/134] vp8enc: do init at set_format time

2011-03-24 10:15:55 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 066/134] vp8enc: fix keyframe forcing

2011-03-23 09:45:20 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 065/134] basevideocodec: remove redundant caps field
	  ... as it is already at hand as the src pad's negotiated caps.

2011-03-23 08:50:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 064/134] vp8enc: use baseclass event virtual handler

2011-02-20 14:16:18 -0800  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8dec.h:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 063/134] basevideo: merge utils header into basevideocodec

2011-03-17 16:34:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/vp8/Makefile.am:
	  [MOVED FROM BAD 062/134] vp8: fix LIBADD order in Makefile.am

2011-02-04 09:08:26 +0100  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 061/134] vp8enc: Add description for bitrate units.

2010-11-30 18:43:24 -0800  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 060/134] vp8enc: Readd setting of granulepos
	  Revert parts of last patch that removed setting of granulepos.
	  oggmux still requires correct granulepos in incoming packet.

2010-11-29 20:21:31 -0800  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 059/134] vp8enc: Don't override timestamps set by base class
	  Because the base class does it correctly.
	  Fixes: #635720, #625558.

2010-11-25 18:52:47 +0100  Edward Hervey <bilboed@bilboed.com>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 058/134] vp8: Remove dead assignments

2010-10-09 17:36:07 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 057/134] basevideo: Move common fields/functions to basecodec

2010-09-18 17:28:48 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 056/134] basevideo: Move deadline to frame structure

2010-08-13 14:34:21 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 055/134] vp8dec: Set GstBaseVideoDecoder::packetized to TRUE as soon as possible
	  This fixes an infinite loop if an EOS event is received before
	  GstBaseVideoDecoder::start() is called, e.g. immediately when the
	  pads are activated.
	  Fixes bug #626815.

2010-07-10 16:52:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 054/134] vp8enc: Add support for enabling automatic insertion of alt-ref frames by the encoder

2010-07-10 16:51:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 053/134] vp8enc: Fix handling of invisible/alt ref frames

2010-07-03 17:47:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8dec.h:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	  [MOVED FROM BAD 052/134] vp8: Add initial documentation, based on the theoradec/theoraenc documentation

2010-07-03 17:34:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/Makefile.am:
	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8dec.h:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8enc.h:
	* ext/vp8/plugin.c:
	  [MOVED FROM BAD 051/134] vp8: Move structure definitions, etc to public header files for gtk-doc

2010-06-12 09:02:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 050/134] vp8enc: Implement multipass encoding
	  Fixes bug #621348.

2010-06-14 15:56:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 049/134] vp8enc: Set VP8E_SET_CPUUSED to 0
	  This setting controls how much CPU can be used by the encoder, specified
	  in fractions of 16. Negative values mean strict enforcement of this
	  while positive values are adaptive.
	  The default value is -4, which means that we're not running as fast
	  as possible and probably are wasting some quality. 0 is the recommended
	  default by libvpx upstream.

2010-06-14 15:51:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 048/134] vp8enc: Use VPX defines for REALTIME, GOOD/BEST quality deadlines instead of our own
	  These are the values used for the speed property.

2010-06-03 10:49:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 047/134] vp8enc: fix printf format warning in log message
	  gstvp8enc.c:564: error: format ‘%d’ expects type ‘int’, but argument 8 has type ‘size_t’
	  gstvp8enc.c:744: error: format ‘%d’ expects type ‘int’, but argument 8 has type ‘size_t’

2009-07-03 16:08:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/vp8/Makefile.am:
	  [MOVED FROM BAD 046/134] basevideo, vp8: guard unstable API with GST_USE_UNSTABLE_API
	  Add some guards and fat warnings to the header files with still unstable
	  API, so people who just look at the installed headers know that it
	  actually is unstable API.
	  Merging previous commit into current codebase.

2010-06-01 15:54:51 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/Makefile.am:
	* ext/vp8/gst/video/gstbasevideocodec.c:
	* ext/vp8/gst/video/gstbasevideocodec.h:
	* ext/vp8/gst/video/gstbasevideodecoder.c:
	* ext/vp8/gst/video/gstbasevideodecoder.h:
	* ext/vp8/gst/video/gstbasevideoencoder.c:
	* ext/vp8/gst/video/gstbasevideoencoder.h:
	* ext/vp8/gst/video/gstbasevideoparse.c:
	* ext/vp8/gst/video/gstbasevideoparse.h:
	* ext/vp8/gst/video/gstbasevideoutils.c:
	* ext/vp8/gst/video/gstbasevideoutils.h:
	* ext/vp8/gst/video/gstvideocompat.c:
	* ext/vp8/gst/video/gstvideocompat.h:
	  [MOVED FROM BAD 045/134] basevideo: Move base video from vp8 to gst-libs

2010-05-26 06:52:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8utils.h:
	  [MOVED FROM BAD 044/134] vp8: Use VPX_PLANE_* instead of PLANE_*

2010-05-24 11:04:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8utils.h:
	  [MOVED FROM BAD 043/134] vp8: Add compatilibity defines to work with older versions of libvpx too

2010-05-23 09:28:13 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 042/134] vp8dec: s/IMG_FMT_I420/VPX_IMG_FMT_I420/
	  This corresponds to upstream libvpx commit 6cd4a10e167203d1deb79abf60ee72599e97891b

2010-05-22 12:55:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 041/134] vp8enc: Allow a maximum keyframe distance of 0, i.e. all frames are keyframes

2010-05-22 08:45:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 040/134] vp8dec: Set decoder deadline from the QoS information

2010-05-28 16:35:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 039/134] vp8enc: Move debug output one line above where the packet is still valid

2010-05-28 15:53:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 038/134] vp8enc: Correctly ignore non-frame packets from the encoder
	  Fixes bug #619916.

2010-05-22 07:44:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gst/video/gstbasevideodecoder.c:
	  [MOVED FROM BAD 037/134] basevideodecoder: Take the frame duration into account when calculating the earliest time
	  This formula is used in many other elements too.
	  Fixes bug #619318.

2010-05-22 07:35:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gst/video/gstbasevideodecoder.c:
	  [MOVED FROM BAD 036/134] basevideodecoder: Reset QoS values when necessary

2010-05-22 09:35:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 035/134] vp8enc: Use GST_VIDEO_CAPS_YUV(I420) instead of handwritten I420 caps for the pad template
	  Fixes bug #619344.

2010-05-21 20:53:36 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/gst/video/gstbasevideodecoder.c:
	* ext/vp8/gst/video/gstbasevideodecoder.h:
	* ext/vp8/gst/video/gstbasevideoutils.h:
	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 034/134] vp8dec: drop late frames after decoding them
	  This saves a memcpy, which is always something.

2010-05-21 21:28:29 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 033/134] vp8enc: threads property
	  Increasing from 1 to 2 threads on an Thinkpad X60s decreased encode time
	  in a test from ~24 s to ~19 s, so this is quite useful.
	  Ideally we should let 0 be the default and automatically match the number
	  of CPU cores (or something).

2010-05-21 15:17:46 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 032/134] vp8enc: add mode property to switch between CBR/VBR
	  Always using CBR when bitrate is used isn't that great, VBR mode
	  can produce meaningful results too.

2010-05-21 10:54:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 031/134] vp8dec: Only enable postprocessing if the decoder supports it

2010-05-21 08:23:58 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/plugin.c:
	  [MOVED FROM BAD 030/134] vp8: typo: s/HAVE_VP8_DECODER/HAVE_VP8_ENCODER/
	  Fixup for bug #619172.

2010-05-21 08:13:06 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 029/134] vp8: move #ifdef HAVE_VP8_ENCODER/DECODER
	  Otherwise we'll try including e.g. <vpx/vp8cx.h> which doesn't exist.

2010-05-20 20:06:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 028/134] vp8enc: Write GStreamer element and version in the vorbiscomment vendor string

2010-05-20 16:49:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/plugin.c:
	  [MOVED FROM BAD 027/134] vp8: Only enable the encoder or decoder if it's available in libvpx
	  Fixes bug #619172.

2010-05-20 10:19:54 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/plugin.c:
	  [MOVED FROM BAD 026/134] vp8: exlcude dec/enc based on CONFIG_VP8_DECODER/ENCODER
	  This may not be very autotoolish, but works with libvpx in the state
	  that libvpx is actually in. Moved the debug init to the elements
	  themselves to minimize amount of #ifdefs

2010-05-20 09:24:53 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 025/134] vp8enc: Limit max-latency to 25 to match libvpx
	  From libvpx/vp8/encoder/onyx_int.h:
	  #define MAX_LAG_BUFFERS (CONFIG_REALTIME_ONLY? 1 : 25)
	  While we don't need to be tied to what libvpx does internally, it
	  doesn't make sense to pretend to support longer frame lags than are
	  actually possible.

2010-05-20 09:56:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8utils.c:
	  [MOVED FROM BAD 024/134] vp8: Undef HAVE_CONFIG_H before including libvpx headers
	  A public libvpx header includes private headers if this is
	  defined, causing compilation failures because the private headers
	  are not installed of course.

2010-05-20 08:53:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 023/134] vp8enc: Some more minor adjustments for the Ogg mapping

2010-05-19 23:02:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 022/134] vp8dec: Fix memory leak

2010-05-19 21:34:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 021/134] vp8enc: Adjust Ogg mapping for the changes

2010-05-19 18:12:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 020/134] vp8dec: Add properties to control the VP8 decoder post processing feature
	  This is disabled by default for now.

2010-05-19 17:16:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 019/134] vp8enc: Rename keyframe-interval to max-keyframe-distance
	  And use default settings for buffer sizes until we expose this
	  somehow.

2010-05-19 17:13:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/Makefile.am:
	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/gstvp8utils.c:
	* ext/vp8/gstvp8utils.h:
	  [MOVED FROM BAD 018/134] vp8: Improve error handling and debug output

2010-05-19 14:46:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 017/134] vp8: Use correct strides and plane offsets for GStreamer

2010-05-18 14:47:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 016/134] vp8enc: Implement GstTagSetter interface

2010-05-18 14:33:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 015/134] vp8enc: Fix setting of the keyframe flag on encoded frames

2010-05-18 14:30:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 014/134] vp8enc: Post an error message on the bus if encoder initialization fails

2010-05-18 14:28:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 013/134] vp8dec: Fix memory leaks and fail if initializing the decoder fails

2010-05-18 02:44:54 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 012/134] vp8enc: Set timebase
	  Also misc cleanup.

2010-05-16 10:36:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 011/134] vp8dec: Fix decoding of invisible frames

2010-05-14 14:26:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 010/134] vp8enc: Update the latency when initializing the encoder

2010-05-14 14:02:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 009/134] vp8dec: Correctly initialize stream info before peeking at the stream
	  Otherwise peeking will fail and we'll get invalid values

2010-05-14 11:01:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 008/134] vp8dec: Make sure to pass a keyframe as first frame to the decoder, copy output frames only once and require width/height/etc on the input caps

2010-05-14 10:30:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 007/134] vp8enc: Add support for invisible frames and the Ogg mapping

2010-05-14 01:14:46 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/gstvp8dec.c:
	  [MOVED FROM BAD 006/134] vp8dec: Fix reset after seeking
	  Also remove some unused code.

2010-05-13 21:19:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 005/134] vp8enc: Set frame numbers as buffer offsets

2010-05-13 21:18:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 004/134] vp8enc: Always get as many frames as possible from the encoder

2010-05-13 21:08:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 003/134] vp8enc: Fill the oldest pending frame instead of the newest

2010-05-13 20:20:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/vp8/gstvp8enc.c:
	  [MOVED FROM BAD 002/134] vp8enc: Correctly set delta unit flag for non-keyframes

2010-05-13 01:04:04 -0700  David Schleef <ds@schleef.org>

	* ext/vp8/Makefile.am:
	* ext/vp8/gst/video/gstbasevideocodec.c:
	* ext/vp8/gst/video/gstbasevideocodec.h:
	* ext/vp8/gst/video/gstbasevideodecoder.c:
	* ext/vp8/gst/video/gstbasevideodecoder.h:
	* ext/vp8/gst/video/gstbasevideoencoder.c:
	* ext/vp8/gst/video/gstbasevideoencoder.h:
	* ext/vp8/gst/video/gstbasevideoparse.c:
	* ext/vp8/gst/video/gstbasevideoparse.h:
	* ext/vp8/gst/video/gstbasevideoutils.c:
	* ext/vp8/gst/video/gstbasevideoutils.h:
	* ext/vp8/gst/video/gstvideocompat.c:
	* ext/vp8/gst/video/gstvideocompat.h:
	* ext/vp8/gstvp8dec.c:
	* ext/vp8/gstvp8enc.c:
	* ext/vp8/plugin.c:
	  [MOVED FROM BAD 001/134] vp8: Add encoder/decoder

2012-09-15 22:16:52 +0200  Christian Fredrik Kalager Schaller <uraeus@linuxrisin.org>

	* gst-plugins-good.spec.in:
	  Update spec file with F18 name change and add deinterlacer

2012-09-15 19:06:06 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	  use gst_element_factory_get_metadata to replace obsolete API

2012-09-14 17:55:16 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* sys/osxaudio/gstosxaudiosink.c:
	  replace _get_caps_reffed with _get_caps

2012-09-14 17:08:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audiofx/gststereo.c:
	  replace gst_element_class_set_details_simple with gst_element_class_set_metadata

2012-09-14 17:08:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	  replace gst_element_class_set_details_simple with gst_element_class_set_metadata

2012-09-14 17:07:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* tests/check/elements/qtmux.c:
	  replace gst_element_class_set_details_simple with gst_element_class_set_metadata

2012-09-14 13:30:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* gst/multipart/multipartmux.c:
	* gst/rtp/README:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/y4m/gsty4mencode.c:
	* tests/examples/equalizer/demo.c:
	* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
	* tests/examples/rtp/server-VTS-H263p.sh:
	* tests/examples/rtp/server-decodebin-H263p-AMR.sh:
	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	* tests/icles/gdkpixbufsink-test.c:
	  fix more caps

2012-09-14 02:57:44 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	  Back to development

=== release 0.11.94 ===

2012-09-14 02:48:43 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	* configure.ac:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.11.94

2012-09-14 01:50:44 +0100  Tim-Philipp Müller <tim@centricular.net>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update translations

2012-09-14 01:46:14 +0100  Tim-Philipp Müller <tim@centricular.net>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update docs

2012-09-14 00:47:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/wavpackenc.c:
	  tests: push stream-start and segment events in wavpackenc test

2012-09-13 10:56:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2: remove unused properties

2012-09-13 10:15:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: disable reconfigure
	  See https://bugzilla.gnome.org/show_bug.cgi?id=683902

2012-09-10 22:09:59 -0700  Jan Schmidt <thaytan@noraisin.net>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Don't treat every custom-downstream event as EOS
	  Don't fall through to the EOS handling after receiving a
	  custom-downstream event.

2012-09-12 21:05:44 +0200  Stefan Sauer <ensonic@users.sf.net>

	* ext/cairo/gsttextoverlay.c:
	* gst/avi/gstavimux.c:
	* gst/flv/gstflvmux.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstqtmux.c:
	* gst/matroska/matroska-mux.c:
	* gst/multipart/multipartmux.c:
	* gst/smpte/gstsmpte.c:
	* gst/videomixer/videomixer2.c:
	  collectpads: remove gst_collect_pads_add_pad_full
	  Rename gst_collect_pads_add_pad_full() to gst_collect_pads_add_pad() and fix all
	  invocations.

2012-09-12 17:14:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udp: add include for IPPROTO_*

2012-09-12 16:39:08 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udp: properly match braces and cpp directives
	  Fixes compilation where IPV6_TCLASS not defined.

2012-09-12 14:42:07 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: Use default query handler where needed
	  And clean up get_caps code while I'm at it

2012-09-12 13:28:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: improve framerate transform
	  Handle G_MAXINT in the framerates better. If we cannot double or divide the
	  framerate, clamp to the smallest/largest possible value we can express instead
	  of failing.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683861

2012-09-12 13:17:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: small cleanup

2012-09-07 17:20:57 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/videomixer2.c:
	  videomixer2: Adding nv12 and nv21 support
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683841

2012-09-12 10:18:53 +0200  Michael Smith <msmith@rdio.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: add support for prores
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683839

2012-09-12 00:16:31 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/rganalysis.c:
	  tests: fix most of the rganalysis unit tests
	  Before the element would post messages on the bus itself, now
	  the sinks do that based on the tag events they receive. But
	  since we don't have proper sink elements in these unit tests,
	  but just dangling pads, we have to post the tag messages the
	  test checks for ourselves.
	  Down from 52/55 failing to 7/52 failing.

2012-09-11 17:36:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/wavparse/gstwavparse.c:
	  ext, gst: only activate in pull mode if upstream is seekable

2012-09-11 15:38:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: disable renegotiation
	  We can't yet wait for the bufferpool to DRAIN before starting renegotiation so
	  disable it for now.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=682770

2012-09-11 12:48:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  tests: rtpbin: port to the new GLib thread API

2012-09-11 12:36:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: port to the new GLib thread API

2012-09-11 11:59:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't reset segment
	  Don't reset the segment because we need the values for accumulation. the segment
	  is reset at start and after a flushing seek. Fixes some problems with files with
	  quicktime segments.

2012-09-10 17:14:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/id3demux.c:
	  tests: fix id3demux test

2012-09-10 14:31:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/amfdefs.h:
	* gst/flv/gstflvdemux.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	  gst: adjust comment style

2012-09-10 14:30:42 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: remove defunct commented code

2012-09-10 13:35:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: consider stream alive when not connected yet
	  When we start and renegotiate, there is a moment where the stream is created but
	  not yet connected. Make sure all functions deal with this situation correctly
	  instead of erroring out.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681247

2012-09-10 12:15:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: don't fail when not negotiated yet
	  When get_time is called but we are not yet negotiated, return 0 instead of
	  posting an error. It's possible that the base class is still negotiating when
	  our get_time is called.

2012-09-10 11:32:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-source.c:
	  update for audio base src api change

2012-09-10 00:42:52 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/avi/gstavimux.c:
	* gst/isomp4/qtdemux.c:
	  video/x-3ivx and video/x-xvid -> video/mpeg,mpegversion=4
	  If it ever turns out that we really must use thoe specific
	  fourccs and not the generic one, we can still add a flavor
	  field to the caps later.

2012-09-07 16:15:42 +0200  Daniela <daniela.muzzu@selexelsag.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid leak
	  When setup fails, make sure to cleanup afterwards.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673509

2012-09-07 15:23:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpamrdepay.c:
	  rtpamrdepay: unmap rtp buffer
	  ... thereby plugging a memleak.

2012-09-07 14:13:17 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: adjust to modified bufferlist semantics
	  ... now implemented by buffer memory blocks.

2012-09-07 14:11:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: avoid crashing on NULL access in debug message

2012-09-07 14:11:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: plug caps leak

2012-09-06 17:09:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: remove redundant _set_allocation call

2012-09-06 17:05:00 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  tests: deinterlace: do not leak deinterlace pads

2012-09-06 17:04:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: plug some leaks

2012-09-06 16:49:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: reuse core function for GCD

2012-09-06 16:31:00 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: support filter in getcaps

2012-09-06 16:30:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: do not leak getcaps result

2012-09-06 16:23:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: add support for bufferpool
	  Add bufferpool support to avoid a memcpy in the videosink when actively
	  interlacing.
	  Remove some commented obsolete code.

2012-09-06 13:38:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: proxy allocation query in passthrough
	  We can let the allocation query pass when we are operating in passthrough mode.

2012-09-06 13:23:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: use default event functions
	  instead of blindly forwarding unknown events.

2012-09-06 13:23:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: small cleanups

2012-09-06 12:56:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: call default query handlers
	  Call the default query handler instead of forwarding the query blindly. Fixes
	  issues of strides because of proxying the allocation query wrongly.

2012-09-06 10:42:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: remove unused code.

2012-09-06 10:42:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulse: improve debug

2012-09-05 11:50:05 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	  dvdemux: remove obsolete update newsegment handling code

2012-09-04 12:35:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: avoid deadlock
	  _update_properties takes the object lock and should not be called when the
	  object lock is already taken.

2012-09-03 12:46:03 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: extract interlaced-ness of video track from interlace-mode field
	  instead of the old boolean "interlaced" field.

2012-09-03 02:51:24 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/avi/gstavimux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* tests/check/elements/avimux.c:
	  video/x-xvid -> video/mpeg,mpegversion=4

2012-09-02 02:50:50 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  text/plain + text/x-pango-markup -> text/x-raw

2012-09-02 01:31:53 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/soup/gstsouphttpsrc.c:
	* gst/matroska/matroska-demux.c:
	  gst_message_new_duration -> gst_message_new_duration_changed

2012-08-30 22:07:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: also stop probatation on existing sources
	  Receiving an RTCP packet should also stop probation on sources we have seen
	  before.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683065

2012-08-22 16:36:21 -0700  Aleix Conchillo Flaque <aleix@oblong.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtp: make rtp packet probation configurable (bug #682512)

2012-08-30 12:21:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbuf: adjust to modified video overlay composition API

2012-08-30 11:30:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fixup 0.11 port of suspect frame checking
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=682959

2012-08-28 18:56:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: avoid invalid H264 bytestream codec_data
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681369

2012-08-28 19:00:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: port segment event creation to 0.11

2012-08-28 16:28:13 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: release extra event ref when replacing pending newsegment event

2012-07-03 17:50:24 +0200  David Corvoysier <david.corvoysier@orange.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_types.c:
	  isomp4: add DASH tfdt box support
	  MPEG DASH has defined a set of new boxes to specify duration, indexes and
	  offsets of ISOBMFF fragments.
	  The Track Fragment Base Media Decode Time (tfdt) Box can in particular be
	  included inside a traf box to specify the absolute decode time, measured on the
	  media timeline, of the first sample in decode order in the track fragment.
	  This information can be used by the isomp4 demux to find out the current position of
	  an MP4 fragment in the timeline.
	  This patch adds code to isomp4 to:
	  - parse the tfdt box
	  - adjust the time/position member of the new segment sent when playback starts
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677535

2012-08-26 22:39:55 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/aalib/gstaasink.c:
	* ext/cairo/gstcairorender.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttimeoverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libpng/gstpngdec.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/progressreport.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/equalizer/gstiirequalizernbands.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/interleave/deinterleave.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/README:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/wavparse/gstwavparse.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	* tests/examples/cairo/cairo_overlay.c:
	* tests/examples/rtp/client-H263p-AMR.sh:
	* tests/examples/rtp/client-H263p-PCMA.sh:
	* tests/examples/rtp/client-H263p.sh:
	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-H264.sh:
	* tests/examples/rtp/client-PCMA.sh:
	* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
	* tests/examples/rtp/server-VTS-H263p.sh:
	* tests/examples/rtp/server-alsasrc-PCMA.sh:
	* tests/examples/rtp/server-decodebin-H263p-AMR.sh:
	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/icles/gdkpixbufsink-test.c:
	* tests/icles/videocrop-test.c:
	  docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert

2012-08-26 22:32:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/flac/gstflacdec.c:
	* gst/videomixer/videomixer2.c:
	  docs: gst-launch-0.11 -> gst-launch-1.0

2012-08-26 22:08:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/deinterlace/gstdeinterlace.c:
	* tests/check/elements/deinterlace.c:
	  deinterlace: the field in caps is "interlace-mode" not "interlace-method"
	  Fix deinterlace unit test. Need to set right field on output caps.
	  Also remove right field (not old 0.10 "interlaced" boolean field)
	  from caps in unit test before comparing old and new.

2012-08-26 21:45:44 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/icydemux.c:
	  tests: fix icydemux unit test
	  Was waiting for a tag message on the bus, which would never
	  come, because elements don't post those themselves any more
	  but let sinks post them from tag events. Only that there are
	  no sinks in this unit test.

2012-08-26 21:27:00 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/videocrop.c:
	  tests: fix videocrop crop_to_1x1 unit test for GRAY8 format
	  Update table with pixel values with the value actually produced
	  by videotestsrc.

2012-08-27 09:00:45 +0200  Sjoerd Simons <sjoerd@luon.net>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Only print caps if they're provided

2012-08-24 19:43:08 +0100  Michael Rubinstein <mrubinstein@rai-dev.com>

	* gst/videomixer/blend.c:
	  videomixer: fix endianness check on systems where non-glib endianness defines are not set
	  On Windows LITTLE_ENDIAN without the G_ in was not defined,  so the
	  test comes out wrong.

2012-08-22 17:23:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udpsink: don't crash on NULL error
	  Check if there is an error before retrieving its message.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=682481

2012-08-22 13:30:19 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 668acee to 4f962f7

2012-08-22 13:18:00 +0200  Stefan Sauer <ensonic@users.sf.net>

	* configure.ac:
	  configure: bump gtk-doc req to 1.12 (mar-2009)
	  This allows us to e.g. unconditionally use gtkdoc-rebase.

2012-08-22 11:21:38 +0200  Martin Ertsaas <mertsas@cisco.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: Make osxvideosink use the non-deprecated threading api from glib.
	  https://bugzilla.gnome.org/show_bug.cgi?id=682446

2012-08-14 15:40:31 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Handle negotiation events
	  This makes sure that we:
	  a) Destroy an existing stream if a negotiate() request comes in: this is
	  required when receiving a downstream renegotiation request after a
	  stream has been created.
	  b) Create a new stream on prepare(): this is required since we do a
	  setcaps() in negotiate(), which causes the stream to be dropped by a
	  ringbuffer release() call (this does not happen during first negotiation
	  since the release is only done on a running ringbuffer). The subsequent
	  call to ringbuffer acquire() fails because the stream was lost on
	  release().
	  https://bugzilla.gnome.org/show_bug.cgi?id=681247

2012-08-14 15:38:27 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseutil.c:
	  pulse: Clear unpositioned flag when setting positions
	  If converting a PA channel map to gst channel positions results in a
	  valid set of channel positions, we clear the unpositioned flag from the
	  ringbuffer spec.

2012-08-14 09:37:45 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Remove redundant channel-mask setting for stereo case
	  The gstaudio helper libraries already take care of this case for us.

2012-08-14 09:36:30 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Don't use memset to set invalid channel positions
	  This itereates over the GstAudioInfo to set invalid channel positions
	  rather than use memset() which works right now because it assumes that
	  GST_AUDIO_CHANNEL_POSITION_INVALID is -1.

2012-08-22 10:30:04 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	  gdkpixbufsink: minor docs improvement

2012-08-22 10:23:24 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
	  gdkpixbuf: re-enable already-ported gdkpixbufsink

2012-08-22 10:08:08 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
	  gdkpixbuf: port gdkpixbufoverlay element to 0.11

2012-08-22 00:00:46 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	* ext/gdk_pixbuf/gstgdkpixbufdec.h:
	* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
	  gdkpixbuf: re-enable already-ported gdkpixbuf element as gdkpixbufdec
	  Not sure why it as disabled exactly given that it had already
	  been ported (though without metas or baseclass).
	  Move plugin_init bits into separate source file, and rename
	  decoder element to gdkpixbufdec.

2012-08-21 23:25:47 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/gdk_pixbuf/gst_loader.c:
	  gdkpixbuf: remove old and unused gst_loader source file
	  Once upon a time used to load GStreamer vids via GdkPixbuf API.

2012-08-16 16:51:16 -0700  Aleix Conchillo Flaque <aleix@oblong.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: make jitterbuffer drop-on-latency available (fix #682055)
	  Conflicts:
	  gst/rtsp/gstrtspsrc.h

2012-08-21 19:47:45 +0800  Huacai Chen <chenhc@lemote.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: make gst_v4l2_fill_lists() adapt to kernel 3.3+
	  When do v4l2_ioctl() with VIDIOC_ENUMINPUT fails on some devices,
	  kernels before 3.3.0 return EINVAL, but newer kernels return ENOTTY.
	  This patch make those devices work well on kernel 3.3+.
	  Related kernel commit:
	  http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=commit;h=07d106d0a33d6063d2061305903deb02489eba20
	  Signed-off-by: Huacai Chen <chenhc@lemote.com>
	  Signed-off-by: Rui Wang <wangr@lemote.com>
	  Signed-off-by: Jie Chen <chenj@lemote.com>

2012-08-20 23:30:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* docs/plugins/inspect/plugin-matroska.xml:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  video/x-dvd-subpicture -> subpicture/x-dvd

2012-08-17 20:52:42 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: fix example pipeline in docs

2012-08-17 14:59:57 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* tests/check/elements/equalizer.c:
	  equalizer: enable presets for the n-band equalizer
	  Add a test for saving and restoring the preset.

2012-08-14 01:20:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix not-negotiated errors on variable or missing framerate in input caps
	  Remove some bogus code I added during porting that would error out
	  on missing or variable framerates in input caps. Handle this like
	  we do in 0.10
	  Fixes test_mode_disabled_passthrough unit test check.

2012-08-12 13:16:32 +0200  Sjoerd Simons <sjoerd@luon.net>

	* gst/law/alaw-decode.c:
	* gst/law/mulaw-decode.c:
	  law: Filter layout caps field
	  The layout caps field shouldn't be passed through to the sink pad
	  of {mu,a}lawdec.
	  https://bugzilla.gnome.org/show_bug.cgi?id=681677

2012-08-09 19:41:34 +0300  Anton Belka <antonbelka@gmail.com>

	* ext/flac/gstflacenc.c:
	  flacenc: allow a TOC with single alternative top-level entry
	  Allow a TOC that has a single alternative top-level entry
	  with multiple sequence sub-entries
	  https://bugzilla.gnome.org/show_bug.cgi?id=540891

2012-08-09 11:48:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: Give MARGINAL rank to the mpg123 decoder element

2012-08-09 10:31:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: And fix the GTK check to use the correct pkg-config package name

2012-08-09 10:25:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Fix GTK required version variable name

2012-08-09 08:35:23 +0100  Matthias Clasen <mclasen@redhat.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: fix build with recent kernels, the v4l2_buffer input field was removed
	  This was unused apparently and removed in the kernel in commit:
	  From 2b719d7baf490e24ce7d817c6337b7c87fda84c1 Mon Sep 17 00:00:00 2001
	  From: Sakari Ailus <sakari.ailus@iki.fi>
	  Date: Wed, 2 May 2012 09:40:03 -0300
	  Subject: [PATCH] [media] v4l: drop v4l2_buffer.input and V4L2_BUF_FLAG_INPUT
	  Remove input field in struct v4l2_buffer and flag V4L2_BUF_FLAG_INPUT which
	  tells the former is valid. The flag is used by no driver currently.
	  https://bugzilla.gnome.org/show_bug.cgi?id=681491
	  Conflicts:
	  sys/v4l2/gstv4l2bufferpool.c

2012-08-08 17:25:36 -0700  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtph264pay: Make it actually work after cleanups

2012-08-08 17:40:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	  gst: Set alignment at the correct place of GstAllocationParams

2012-08-08 17:39:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* gst/matroska/matroska-demux.c:
	* gst/multipart/multipartmux.c:
	* gst/videomixer/videomixer2.c:
	  gst: Set alignment at the correct place of GstAllocationParams

2012-08-08 16:25:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  Back to development

=== release 0.11.93 ===

2012-08-08 15:22:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.11.93

2012-08-08 15:17:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	* win32/MANIFEST:
	* win32/common/tuner-enumtypes.c:
	* win32/common/tuner-enumtypes.h:
	* win32/common/tuner-marshal.c:
	* win32/common/tuner-marshal.h:
	  win32: add generated tuner-marshal/enumtypes files for v4l2src and update
	  And gst-indent the right rtp marshal files; add missing files to MANIFEST.

2012-08-08 15:10:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/tvtime-dist.c:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videomixer/blendorc-dist.c:
	  gst: update disted orc files

2012-08-08 12:58:50 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/mpg123/Makefile.am:
	  mpg123: dist header file

2012-08-08 11:31:59 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/wavpack/gstwavpackdec.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* sys/oss4/oss4-audio.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  Silence some 'variable may be used uninitialized' compiler warnings
	  When compiling with -DG_DISABLE_ASSERT

2012-08-08 10:56:51 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* tests/icles/ximagesrc-test.c:
	  No code with side-effects inside g_assert() please

2012-08-07 11:14:21 -0700  Olivier Crête <olivier.crete@collabora.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Return FLUSHING instead of ERROR on unlock
	  If the base class asks multiudpsink to unlock, then it should return
	  FLUSHING, not ERROR

2012-07-26 16:19:57 +0300  Anton Belka <antonbelka@gmail.com>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flacenc: add TOC support
	  Add TOC as embedded cuesheets in flac files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=54089

2012-08-07 12:12:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: generate empty vorbiscomment for complete streamheaders if needed
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681335

2012-08-06 18:02:50 -0700  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Block pad while it is announced.
	  Block the RTP pad and associated RTCP pads while they are being
	  announced. This it to prevent a race where one is announced and
	  before the callback has connected it, the other one gets a buffer.
	  We can't use the "padlock" of ssrcdemux because it causes deadlocks.

2012-08-06 15:00:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  common: un-do accidental common update revert in commit 7b5925b5

2012-08-06 14:50:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: set correct data_size for generated dummy frame
	  ... which prevents getting stuck in a loop if such one is needed.

2012-08-06 14:50:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: improve and fix debug statement
	  ... so it really informs about next rather than past frame.

2012-08-06 12:34:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: update available bytewriter space when repositioning
	  ... and add some more assert to catch potential surprises early on.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680558

2012-08-04 12:47:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	* ext/dv/gstdvdemux.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	  gst: Add stream-id to stream-start events

2012-08-04 12:54:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Chain up to the parent class' query handler if no pad is provided

2012-08-02 01:48:29 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: add a better detection for the main run loop

2012-07-27 16:13:49 +0200  Xavi Artigas <xartigas@fluendo.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Do not overwrite the DS buffer when testing for AC3 support
	  https://bugzilla.gnome.org/show_bug.cgi?id=680706
	  Conflicts:
	  sys/directsound/gstdirectsoundsink.c

2012-08-05 16:39:23 +0100  Tim-Philipp Müller <tim@centricular.net>

	* common:
	  Automatic update of common submodule
	  From 94ccf4c to 668acee

2012-08-03 16:13:52 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Release lock before signalling new pad
	  This prevents a deadlock where something would try to push an event
	  through the SSRC demux from the callback, causing the pads to be iterated
	  and the lock taken.

2012-08-04 16:13:36 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/lame/gstlamemp3enc.c:
	  gst_tag_list_free -> gst_tag_list_unref

2012-08-04 16:10:16 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/debugutils/gsttaginject.c:
	* gst/flv/gstflvdemux.c:
	* gst/icydemux/gsticydemux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/wavparse/gstwavparse.c:
	* tests/check/elements/apev2mux.c:
	* tests/check/elements/icydemux.c:
	* tests/check/elements/id3demux.c:
	* tests/check/elements/id3v2mux.c:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/pipelines/tagschecking.c:
	  gst_tag_list_free -> gst_tag_list_unref

2012-08-03 13:43:31 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: map input buffer in READ mode, not WRITE mode
	  Makes things actually work.

2012-08-03 11:50:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/mpg123/gstmpg123audiodec.c:
	  mpg123: query supported output formats at run-time
	  Fixes stuff. We use a string here since we can't be bothered
	  with GValue.

2012-08-03 14:10:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: manage race between connection closing and flushing
	  ... where the former can happen in task thread and the latter in mainloop
	  upon downward state change.

2012-08-03 14:02:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: improve and relax audio frame parsing
	  ... so as to properly recognize first audio frame.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681077
	  Conflicts:
	  ext/flac/gstflacdec.c

2012-08-03 11:48:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/mpg123/Makefile.am:
	  mpg123: hook up to build system

2012-08-03 11:13:48 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>

	* ext/mpg123/gstmpg123audiodec.c:
	* ext/mpg123/gstmpg123audiodec.h:
	  mpg123: add new libmpg123-based mp3 decoder plugin
	  Needs a bit of cleaning up.
	  https://bugzilla.gnome.org/show_bug.cgi?id=681003

2012-08-01 12:16:41 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix double unref of private tag buffer

2012-07-30 17:54:51 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: create TOC as needed
	  Avoid creating the toc if the wav has no or empty cue chunk.
	  Also a small code cleanup.

2012-07-28 11:26:01 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/wavparse/gstwavparse.c:
	  wavparse: update for TOC API changes

2012-07-28 11:22:43 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-read-common.c:
	  matroska: update for TOC API changes

2012-07-28 11:20:08 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: update for TOC API changes

2012-07-28 00:19:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflactag.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/wavpack/gstwavpackdec.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/debugutils/gsttaginject.c:
	* gst/flv/gstflvdemux.c:
	* gst/icydemux/gsticydemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-read-common.c:
	* gst/multipart/multipartdemux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/wavparse/gstwavparse.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rgvolume.c:
	  tag: Update for taglist/tag event API changes

2012-07-27 12:05:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/isomp4-plugin.c:
	* gst/isomp4/qtdemux.c:
	  qt(de)mux: pass private blob tags in a sample
	  ... rather than a buffer, and the detailed info in the sample info
	  rather than caps.

2012-07-27 11:31:13 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: Don't return NULL from _transform_caps
	  If _transform_caps () returns NULL, the basetransform _transform_caps
	  tries to call gst_caps_is_subset () with a NULL subset which hits an
	  assertion.

2012-07-27 11:26:18 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: obtain image type from the sample info

2012-07-27 11:25:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: remove extraneous _unref
	  ... since we did not obtain a buffer ref from the GstSample.

2012-07-27 10:14:23 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: Update to use GstSample tag setting API

2012-07-26 16:34:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: modify buffer data rather than buffer itself

2012-07-26 16:28:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: avoid leaking bytewriter instance

2012-07-26 16:04:23 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix timestamp adjustment and caps

2012-07-26 16:03:57 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix/simplify telecine state checks

2012-07-26 12:08:58 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Improve debug output

2012-07-26 12:08:36 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix low-latency pattern locking

2012-07-24 16:19:53 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: RFF should be ignored in deinterlace
	  RFF only occurs on progressive frames in telecine sequences. For
	  deinterlace, we don't want these repeated fields as we will simply be
	  pushing the progressive frame and then moving on.
	  However, we need to consider RFF in order to correctly identify patterns
	  and adjust the timestamps.

2012-07-24 14:59:47 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Improve process logic
	  The logic now works better if we filter orphans, then progressive, then
	  telecine interlaced fields which need to be woven and fall through to
	  interlace. Telecine interlaced fields will be regularly deinterlaced if
	  there is no pattern lock for us to be sure that we have a telecine
	  pattern.
	  Telecine sequences that aren't 24fps progressive with RFF flags can't
	  really be tested until fieldanalysis is ported.

2012-07-25 16:02:34 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: only set complete output caps once
	  ... so as to avoid downstream complaints about missing streamheaders.

2012-07-25 15:29:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: also support S24_32 output

2012-07-25 15:28:14 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: pass correct parameters to encoder lib

2012-07-25 14:57:13 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: adjust to modified audioencoder getcaps helper API

2012-07-25 12:50:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: go and stay in the loop function on PLAY
	  When we have a PLAY request, go into the LOOP function next. When we are
	  looping, keep on looping until we are told otherwise.
	  This fixed rtsp and TCP connections.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680551

2012-07-25 12:49:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: set caps after activating the pad

2012-07-25 12:49:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: small cleanups

2012-07-25 10:08:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/gstrtpxqtdepay.c:
	  xqtdepay: fix buffer refcount error
	  After pushing the buffer into the adapter, we should not let the baseclass push
	  it out anymore. This error was introduced while porting to 0.11.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=680540

2012-07-24 21:41:53 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: remove obsolete liboil comment

2012-07-24 21:11:18 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: push mode: increase segment accuracy following seek
	  Conflicts:
	  gst/matroska/matroska-demux.c

2012-07-24 16:41:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: perform proper KEY_UNIT seek also in push mode
	  Conflicts:
	  gst/matroska/matroska-demux.c

2012-07-24 19:04:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: don't crash dereferencing NULL error when leaving multicast group on shutdown
	  Strangely enough, if we do pass an error variable to be filled, we
	  no longer get an error on leaving.

2012-07-24 15:55:12 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: rearrange some checks to avoid NULL use

2012-07-24 15:38:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: use same fourcc to determine caps in determining uncompressed-ness
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673898
	  Conflicts:
	  gst/avi/gstavidemux.c

2012-07-24 15:36:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  Revert "avidemux: Don't consider 0 fcc_handler as uncompressed."
	  This reverts commit c6b9f5b25ab435669816a07049b0e5a8f01e09ca.
	  fourcc GST_RIFF_rgb = 0 still leads to raw uncompressed rgb caps.
	  See also https://bugzilla.gnome.org/show_bug.cgi?id=673898

2012-07-24 12:10:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix up example pipeline some more
	  No more ffmpegcolorspace

2012-07-20 16:30:00 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Fix the example gst-launch pipeline.

2012-07-24 12:33:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: avoid NULL access when checking subtitle
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680388

2012-07-24 12:22:08 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Reset parser when we have caps without codec_data
	  This ensures the detection (and proper downstream caps settings) will
	  actually happen when we have new incoming caps without codec_data.
	  This was easily triggered by streams from matroskademux which initially
	  provided caps with a constructed codec_data, but then pushed new caps
	  without the codec_data once it detected the stream was adts.

2012-07-24 09:17:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	* gst/videomixer/blendorc.orc:
	  videomixer: prefix orc functions with video_mixer_orc_

2012-07-24 09:13:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videobox/gstvideoboxorc.orc:
	  videobox: prefix orc functions with video_box_orc_

2012-07-23 18:51:00 +0200  Christian Fredrik Kalager Schaller <uraeus@linuxrisin.org>

	* gst-plugins-good.spec.in:
	  Update spec file with latest changes

2012-07-23 17:37:58 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: generate correct segment stream time
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680275

2012-07-23 16:42:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kdepay.h:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpj2kpay.h:
	  rtp: always use buffer lists

2012-07-23 15:24:17 +0200  Patricia Muscalu <patricia@axis.com>

	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	  rtpmp4vpay: always enable buffer-lists

2012-07-23 15:22:24 +0200  Patricia Muscalu <patricia@axis.com>

	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpjpegpay.h:
	  rtpjpegpay: always enable buffer-lists

2012-07-23 15:49:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: get frame flags correctly
	  Also move the deinterlace plugin to ported status

2012-07-23 15:33:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: proper parse recovery after seek
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680427

2012-07-23 12:39:05 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: clear old segment event when requesting new one
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680283

2012-07-23 10:32:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	  ext: Update for video base classes API changes

2012-07-23 08:49:07 +0200  Alban Browaeys <prahal@yahoo.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: convert all non GST_FORMAT_BYTES to format bytes.
	  Convert all non GST_FORMAT_BYTES to format bytes:
	  fixes:
	  GStreamer-CRITICAL **: gst_query_set_duration: assertion `format ==
	  g_value_get_enum (gst_structure_id_get_value (s, GST_QUARK (FORMAT)))'
	  failed
	  when playing more than one wav stream.
	  gst-plugins-base/tests/icles/playback/test7 uri1.wav uri2.wav

2012-07-23 09:25:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Don't fail if more data then needed is available when parsing cue chunks
	  Fixes bug #680328.

2012-07-23 09:22:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Some minor cleanup to the cue/labl parsing

2012-07-23 08:45:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 98e386f to 94ccf4c

2012-07-19 14:55:45 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  deinterlace: Port to 1.0
	  This requires the additional INTERLACED buffer flag recently added to
	  -base

2012-07-20 15:18:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/interleave/interleave.c:
	  interleave: convert the output segment to time
	  Convert the stored input segment to time before pushing it out.
	  Conflicts:
	  gst/interleave/interleave.c

2012-07-20 13:12:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	  interleave: try to fix segment handling
	  Conflicts:
	  gst/interleave/interleave.c

2012-07-20 15:28:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Non-update seeks should still make sure that reverse playback status is reset
	  Conflicts:
	  gst/matroska/matroska-demux.c

2012-07-20 15:18:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Properly initialize from_offset and from_time

2012-07-20 14:25:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: We need an index and index entry for reverse playback
	  Reverse playback does not work with index-less files yet.

2012-07-20 14:10:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: clean up push mode segment handling
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680277

2012-07-20 13:35:29 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: properly transform incoming segment event
	  ... which is really useful for proper push mode seeking.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680278

2012-07-20 11:07:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: Fix reverse playback for seeks without stop position
	  Conflicts:
	  gst/matroska/matroska-demux.c
	  gst/matroska/matroska-demux.h

2012-07-20 10:48:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Only take the stream_start_time into account for SET seeks
	  For other seeks the stream_start_time is already added to the
	  segment values.
	  Conflicts:
	  gst/matroska/matroska-demux.c

2012-07-08 20:36:22 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: Add TOC support
	  Add support for:
	  * Cue Chunk
	  * Associated Data List Chunk
	  * Label Chunk
	  https://bugzilla.gnome.org/show_bug.cgi?id=677306

2012-05-09 15:58:16 +0200  Maria Giovanna Chiossa <mariagiovanna.chiossa at selexelsag.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: also set UDP buffer size in multicast
	  Also set the UDP buffer size in multicast mode.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675448

2012-07-18 23:43:59 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/avi/gstavidemux.c:
	  avidemux: fix header parsing in push mode
	  Fix 'break' that got warped to the wrong place,
	  probably as part of a merge. Fixes GST_IS_BUFFER
	  criticals in parse_idit() when being accidentally
	  passed a NULL buffer because of the missing break.
	  gst-launch-1.0 playbin uri=http://docs.gstreamer.com/media/sintel_trailer-480i.avi

2012-07-18 22:47:22 +0200  Alban Browaeys <prahal@yahoo.com>

	* configure.ac:
	* ext/soup/gstsouphttpsrc.c:
	  soup: deprecated soup_message_headers _get -> _get_one
	  https://bugzilla.gnome.org/show_bug.cgi?id=680206

2012-07-18 18:27:40 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	  jpeg/png: Call video_decoder_negotiate()

2012-07-18 17:57:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/debugutils/gstpushfilesrc.c:
	  update for ghostpad changes

2012-07-18 11:36:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Pass seek rate to upstream seek events in push mode
	  Fixes bug #679435.
	  Conflicts:
	  gst/matroska/matroska-demux.c

2012-07-17 16:39:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  update for RTP buffer api changes

2012-07-17 16:38:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtsp/gstrtpdec.c:
	  update for RTP buffer api changes

2012-07-16 11:07:44 +0200  Patricia Muscalu <patricia@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: use buffer lists
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679994

2012-07-17 10:01:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Fix parsing of ISRC from the cuesheets

2012-07-05 14:15:25 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: add TOC support
	  Add support embedded cuesheets in flac files.
	  Parsing METADATA_BLOCK_CUESHEET as TOC.
	  https://bugzilla.gnome.org/show_bug.cgi?id=540891

2012-07-13 14:43:31 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: avoid some more frame misparsing by additional header sanity check
	  ... using a required constant blocking_strategy bit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=679807

2012-07-13 13:51:48 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	  demux: Push STREAM_START event when needed

2012-07-11 13:10:07 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/isomp4/gstqtmux.c:
	  qtmux: avoid warning if both ts are equal

2012-07-11 12:28:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: check the right size when warning about too large udp packets
	  What matters is the total size, not the size of any of the
	  individual memory chunks that make up the packet.

2012-07-10 14:38:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosink.h:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosink.h:
	  autodetect: proxy ts-offset properties
	  Proxy the ts-offset property in the audio*sink elements.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679343

2012-07-09 16:27:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	  fix for allocator API changes

2012-07-09 12:22:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/matroska/matroska-demux.c:
	* gst/wavparse/gstwavparse.c:
	  update for riff field rename

2012-05-21 13:54:51 +0200  Mathias Hasselmann <mathias@openismus.com>

	* tests/check/Makefile.am:
	  tests: drop redundant elements_level_LDADD line
	  https://bugzilla.gnome.org/show_bug.cgi?id=676302

2012-07-08 13:30:34 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/jpegdec.c:
	  tests: minor jpegdec clean-ups and fixes
	  Fix race condition in eos checking and a leak. And
	  build pipeline without parse_launch.

2012-05-21 13:53:54 +0200  Mathias Hasselmann <mathias@openismus.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/jpegdec.c:
	* tests/files/image.jpg:
	  tests: Add some basic tests for jpegdec
	  https://bugzilla.gnome.org/show_bug.cgi?id=676302

2012-07-08 00:08:55 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/dtmf/gstdtmfsrc.c:
	  dtmfsrc: pass unhandled non-custom events to the base class
	  https://bugzilla.gnome.org/show_bug.cgi?id=666626

2012-07-06 19:11:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: avoid some relocations

2012-07-06 14:49:18 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpmp4vpay.c:
	  rtpmp4vpay: remove deprecated send-config property
	  Use config-interval instead.

2012-07-06 14:42:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: remove deprecated "byte-stream" and "access-unit" properties
	  These will be picked automatically based on downstream caps now, so
	  if you want the depayloader to output a specific format, make sure
	  the element downstream advertises that preference or use a capsfilter
	  after the depayloader to force it.

2012-07-06 14:13:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: remove deprecated and non-functional "profile-level-id" property
	  This is now optionally taken from downstream caps, so can be
	  specified via a capsfilter after the payloader.

2012-07-06 15:07:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: perform additional sanity check before confirming ADTS format
	  ... and tweak confusing debug message.

2012-07-06 15:29:14 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: remove unhelpful stray debug message

2012-07-06 13:16:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: remove deprecated and unused "ntp-ns-base" property

2012-07-06 12:57:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/gstqtmux-doc.c:
	  docs: update isomp4 docs for gppmux -> 3gppmux change as well

2012-07-06 12:54:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	* tests/check/pipelines/tagschecking.c:
	  isomp4: remove gppmux, which was deprecated in favour of 3gppmux

2012-07-06 12:49:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/smpte/gstsmpte.c:
	  smtp: remove deprecated "fps" property

2012-07-06 12:46:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: remove deprecated and unused "autoscan" property
	  Replaced by boundary=NULL.

2012-07-06 09:07:41 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/gstrtph263ppay.c:
	* tests/check/elements/rtp-payloading.c:
	  rtph263ppay: accept any h263 input unless downstream forces specific requirements
	  rtph263ppay should accept any input compatible with its sink template
	  caps if it just outputs to e.g. udpsink or fakesink.
	  rtph263ppay ! rtph263pdepay should also work with any compatible input.
	  This would fail before with not-negotiated errors because the get_caps
	  function would see the encoding-name in the depayloader's template caps
	  and default to baseline H.263 because there's no profile/level information
	  in those caps, which is the right thing to do if downstream has filtercaps
	  from an SDP, but not if those fields are absent because they can be
	  anything like with the depayloader's template caps. Makes
	  videotestsrc ! avenc_h263p ! rtph263ppay ! rtph263pdepay ! fakesink
	  work.

2012-07-05 22:57:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/rtp-payloading.c:
	  tests: fix h263p payload ! depayload unit test
	  Need to add h263version field to input caps since the
	  payloader sink get_caps function will contain it in the
	  the caps, and the stricter caps subset check requires
	  this to be present in the input caps as well then.

2012-07-06 11:50:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngenc.c:
	* sys/v4l2/gstv4l2sink.c:
	  update for query api changes

2012-07-06 11:26:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/dv/gstdvdec.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* sys/v4l2/gstv4l2src.c:
	  update for query api changes

2012-07-06 11:02:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngenc.c:
	* sys/v4l2/gstv4l2sink.c:
	  update for allocation query changes

2012-07-05 15:14:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/rgvolume.c:
	  tests: fix rgvolume unit test event handling
	  Must flush after EOS before sending more buffers or
	  another EOS event, or the event or buffer will be
	  rejected. Also send a SEGMENT event at the start
	  of each stream for good measure.

2012-07-05 13:13:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/wavparse/gstwavparse.c:
	  gst: Implement segment-done event

2012-07-05 12:35:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Remove the TOC query handling

2012-07-04 19:52:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-read-common.c:
	  matroska: Update for new GstToc API
	  TOC support in matroskamux is disabled for now as it was broken anyway.

2012-07-04 23:57:18 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/rganalysis.c:
	  tests: fix rganalysis unit test event handling
	  Must flush after EOS before sending more buffers or
	  another EOS event, or the event or buffer will be
	  rejected. Also send a SEGMENT event at the start
	  of each stream for good measure.

2012-07-04 18:58:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: clear 0 DTS on buffers output, as sinks will prefer DTS over PTS for syncing
	  Since the initial decoded still image buffer will have dts=pts=0, and
	  we only set PTS on buffers we push out, all buffers pushed out would
	  have a DTS of 0. Sinks, however, will prefer DTS over PTS if both are
	  set, and will therefore always see a timestamp of 0 no matter what
	  the PTS is set to.
	  Fixes unit test too.

2012-07-04 20:59:03 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix query function implementation; more debugging

2012-07-04 19:41:52 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix spec stuff in directsoundsink

2012-05-31 19:22:47 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: fix access to invalid pointer in set_volume

2012-06-13 12:12:39 +0200  Sebastian Dr=C3=B6ge <sebastian.droege@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix caps leaks

2012-05-29 11:37:59 +0000  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: fix acceptcaps check

2012-05-25 10:14:57 +0000  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: use helper function to check for spdif formats

2012-05-25 10:19:09 +0000  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: add support for DTS

2012-05-08 16:23:42 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: force 48000 kHz force AC-3 over spdif

2012-07-04 17:42:49 +0400  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: add support for ac-3 over spdif

2012-07-04 12:37:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  tests: disable deinterlace test for now, element still needs to be ported
	  But leave it active and print a FIXME. Porting is in progress.

2012-07-03 19:38:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/interleave/deinterleave.c:
	  deinterleave; downgrade caps change failure debug message
	  Add some more info and downgrade to warning, so
	  it doesn't look like the unit test failed.

2012-07-03 17:52:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: fix negotiation and unit test
	  Must remove a possibly-fixed channel-mask field if
	  we're going to set unfixed channels on the structure,
	  or a different channel count.

2012-07-03 17:26:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Only push the TOC event, the message is handled by the sinks

2012-07-03 12:47:58 +0900  Javier Jardón <jjardon@gnome.org>

	* tests/examples/equalizer/demo.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/icles/gdkpixbufsink-test.c:
	  tests: do not use deprecated gtk+ symbols
	  https://bugzilla.gnome.org/show_bug.cgi?id=679301

2012-07-03 09:27:17 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	  configure: require Gtk+ 3.0 for tests/examples

2012-07-03 12:57:18 +0900  Javier Jardón <jjardon@gnome.org>

	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	  rtp: remove some outdated comments
	  https://bugzilla.gnome.org/show_bug.cgi?id=679301

2012-06-29 11:51:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: default to force-aspect-ratio=true

2012-06-28 20:03:05 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: add push mode support
	  https://bugzilla.gnome.org/show_bug.cgi?id=656317

2012-06-28 11:29:55 +0200  David Corvoysier <david.corvoysier@orange.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: Try to seek upstream before processing seek push event
	  When it receives a seek in push mode, the qtdemux should first try to push the event upstream, and only if upstream fails fall back to
	  its own seek logic.

2012-06-28 11:47:20 +0200  David Corvoysier <david.corvoysier@orange.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: Allow duration queries to be forwarded upstream
	  When receiving a duration query for TIME format, try to query upstream, and only if upstream fails fall back to qtdemux duration handling.

2012-06-28 11:59:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: cleanups
	  Use the caps properties for alignment and format.
	  Remove some old properties, we always want to use bufferlists when we can now.

2012-06-28 11:32:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  h264pay: prefer AVC, it's easier to parse etc

2012-06-27 09:09:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: mark all output frames as keyframes

2012-06-26 18:48:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  matroska: update for GstToc API additions

2012-06-26 17:04:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska: set interlace-mode

2012-06-26 13:19:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: improve debug

2012-06-26 13:02:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  Revert "v4l2: free kernel buffers before allocating new ones"
	  This reverts commit 1b09bc609a578e731f0dbc8f6e698e25d8f4c5f8.
	  Seems to make libv4l2 complain, maybe because we call REQBUFS with 0 buffers
	  before we allocated buffers.

2012-06-26 12:07:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: free kernel buffers before allocating new ones
	  See https://bugzilla.gnome.org/show_bug.cgi?id=670257

2012-06-26 12:07:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: improve debug

2012-06-26 11:14:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: setup strides and offsets for all planes

2012-06-25 20:11:53 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: update for GstTocSetter changes

2012-06-25 13:31:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Return FALSE from queries if we can't answer POSITION/DURATION queries

2012-06-21 17:15:11 +0300  Anton Belka <antonbelka@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Return FALSE from TOC query if no TOC exists instead of an empty TOC

2012-06-24 22:51:16 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-read-common.c:
	  matroska: update for GstToc API changes

2012-06-23 14:57:28 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: update for gst_element_make_from_uri() changes

2012-06-20 12:31:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/flvdemux.c:
	* tests/check/elements/flvmux.c:
	* tests/check/elements/id3demux.c:
	  update for bus api changes

2012-06-20 10:33:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/flv/gstflvdemux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/wavparse/gstwavparse.c:
	  update for task api change

2012-06-20 09:59:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  update for clock api changes

2012-06-19 12:15:33 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxringbuffer.c:
	* sys/osxaudio/gstosxringbuffer.h:
	  osxaudiosink: respect the prefered channel layout
	  In OSX is allowed to configure the default audio output device,
	  prefered channel layout and speaker positions through the tool
	  "Audio MIDI Setup".

2012-04-30 22:59:58 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Send gap events for subtitle streams

2012-06-17 01:00:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: fix up docs for 0.11

2012-06-16 23:29:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: small uri handler fixup and some more docs
	  Get URI location using gst_uri_get_location(), so any
	  escaped bits get unescaped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=609049

2012-06-17 00:59:21 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: re-port to 0.11

2012-06-16 19:06:25 +0100  Bastien Nocera <hadess@hadess.net>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: Implement splitfile:// URI scheme
	  https://bugzilla.gnome.org/show_bug.cgi?id=609049
	  Conflicts:
	  gst/multifile/gstsplitfilesrc.c

2012-06-14 10:43:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	  theoradepay: fix buffer memory
	  The memory was added to the input buffer instead of the output buffer.

2012-06-13 13:36:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't reset time in flush-stop
	  Don't reset the time in flush-stop. Live sources can do this flush in the
	  playing state and so the pipeline will never have a chance to update the
	  base_time of the elements, which only happens when going from paused to
	  playing.

2012-06-12 12:42:31 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxringbuffer.c:
	* sys/osxaudio/gstosxringbuffer.h:
	  osxaudiosink: Add support for SPDIF output
	  A big refactoring to allow passthrough AC3/DTS over SPDIF.
	  Several random cleanups and minor fixes.

2011-09-01 15:41:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: send QoS messages when dropping a frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=657941

2012-06-12 16:05:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Rework the async state handling
	  Always send the flushing events to the udp elements now that basesrc supports
	  this. This makes sure a segment event is sent correctly after a flush.
	  Keep track of the currently executing command and make it possible to specify
	  what command you want to cancel when starting a new async command.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=677905

2012-06-11 18:24:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/videomixer/videomixer2.c:
	  childproxy: update api use

2012-06-11 12:54:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: always perform full seek if seek is flushing
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677838

2012-06-11 11:20:18 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: printf format fix for long -> int change

2012-06-08 20:38:34 +0200  Hans de Goede <hdegoede@redhat.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Don't probe UVC devices for being interlaced
	  UVC devices are never interlaced, and doing VIDIOC_TRY_FMT on them
	  causes expensive and slow USB IO, so don't probe them for interlaced.
	  This shaves 2 seconds of the startup time of cheese with a Logitech
	  Webcam Pro 9000.
	  Signed-off-by: Hans de Goede <hdegoede@redhat.com>
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677722

2012-06-09 16:53:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/rndbuffersize.c:
	  debug: change rndbuffersize properties from long to int
	  These should all be int instead of long, to avoid bugs
	  when passing these as varargs with g_object_set(), and
	  there was no reason to use long in the first place here.
	  Fixes FIXME.

2012-06-08 15:54:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/rtsp/gstrtpdec.c:
	  elements: Use gst_pad_set_caps() instead of manual event fiddling

2012-06-08 15:04:59 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 03a0e57 to 98e386f

2012-06-08 10:11:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/audioparsers/gstwavpackparse.c:
	* sys/oss4/oss4-audio.c:
	* tests/check/elements/interleave.c:
	  update for audio api change

2012-06-07 16:12:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  Back to development

=== release 0.11.92 ===

2012-06-07 16:12:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.11.92

2012-06-07 16:11:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2012-06-07 15:03:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: improve clock handling
	  Post the notify outside of the pa_lock to avoid a deadlock caused by basesrc
	  calling get_time with the object lock.
	  Reset the clock on connect.
	  Post clock-lost and clock-provide messages.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673977
	  Conflicts:
	  ext/pulse/pulsesrc.c

2012-04-12 13:21:17 +0300  Mohammed Sameer <msameer@foolab.org>

	* ext/pulse/pulsesrc.c:
	  Better GstClock for pulsesrc
	  This clock uses the actual stream time (pa_stream_get_time) to get a more accurate timestamp.
	  Conflicts:
	  ext/pulse/pulsesrc.c

2012-06-07 11:16:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	  png: fix video state leaks

2012-06-07 11:16:37 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix video state leak

2012-06-07 12:11:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: only reset the manager object when we did a seek
	  Only reset the manager object when we used a Range header, ie. when we did a
	  seek. Otherwise we just paused and we can resume just fine.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677475

2012-06-06 16:13:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  tests: add test for rtpsession cleanup

2012-06-06 18:18:41 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 1fab359 to 03a0e57

2012-06-06 14:17:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Update for TOC event API change

2012-06-06 13:02:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflactag.c:
	* ext/soup/gstsouphttpsrc.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/debugutils/gsttaginject.c:
	* gst/flv/gstflvdemux.c:
	* gst/icydemux/gsticydemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-read-common.c:
	* gst/multipart/multipartdemux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/wavparse/gstwavparse.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rgvolume.c:
	  update for tag event change

2012-06-06 13:00:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* tests/check/elements/aspectratiocrop.c:
	* tests/check/elements/videocrop.c:
	  fix Y800 format

2012-06-01 01:19:35 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* configure.ac:
	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideo: straightforward port to 0.11

2012-05-31 18:39:25 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/libpng/gstpngdec.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	  Some printf variable format fixes
	  The osx compiler complains about those

2012-06-05 09:18:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: Fix GstBaseParse::get_sink_caps() implementations
	  They should take the filter caps into account and always return
	  the template caps appended to the actual caps. Otherwise the
	  parsers stop to accept unparsed streams where upstream does not
	  know about channels, rate, etc.
	  Fixes bug #677401.

2012-06-04 16:17:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: set colorimetry on output info

2012-06-04 08:10:15 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxaudio/gstosxringbuffer.c:
	  osxaudiosink: Handle endianness correctly

2012-06-01 16:37:00 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxringbuffer.c:
	  osxaudiosink: Add support for int audio

2012-06-01 10:28:53 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From f1b5a96 to 1fab359

2012-05-31 13:36:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: set the palette size correctly

2012-05-31 10:15:43 +0200  Michael Jones <michael.jones@matrix-vision.de>

	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2vidorient.h:
	  v4l2: add missing G_END_DECLS
	  G_BEGIN_DECLS didn't have matching G_END_DECLS
	  https://bugzilla.gnome.org/show_bug.cgi?id=677165

2012-05-31 13:08:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 92b7266 to f1b5a96

2012-05-31 10:26:27 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	  osxvideosink: Really fix the build on 10.5
	  The API that we use to run the Cocoa loop in another
	  thread does not exist in 10.5 or earlier.

2012-05-26 12:21:18 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix race in starting the runloop thread
	  Block gst_osx_video_sink_run_cocoa_loop until the loop thread has started and
	  finished initializing NSApp. Fixes occasional warnings/crashes due to two
	  threads going inside NSApp before finishLaunching had completed.

2012-05-30 16:03:55 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	  osxvideosink: Fix last commit to actually work
	  MAC_OS_X_VERSION_10_6 is obviously not defined on 10.5.

2012-05-30 13:51:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/osxvideo/Makefile.am:
	  osxvideosink: Put the right flags in the right variable

2012-05-30 13:24:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Fix GST_OBJCFLAGS

2012-05-30 12:45:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From ec1c4a8 to 92b7266

2012-05-30 12:43:37 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/osxvideo/osxvideosink.h:
	  osxvideosink: NSWindowDelegate is available in all OSX versions newer than 10.6

2012-05-30 12:40:57 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	  osxvideosink: Fix build with older OSX versions

2012-05-30 11:09:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* sys/osxvideo/Makefile.am:
	  configure: Add OBJC specific compiler flags
	  See bug #643939.

2012-05-30 11:23:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 3429ba6 to ec1c4a8

2012-05-29 17:50:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videocrop/gstvideocrop.c:
	  video: remove duplicate format

2012-05-29 16:52:02 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Post error message if EOS before pads were created
	  Happens with some files with only headers

2012-05-28 15:22:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngdec.h:
	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	  png: Port to 0.11 again

2012-05-14 12:46:57 +0200  Jens Georg <mail@jensge.org>

	* ext/soup/gstsouphttpsrc.c:
	  soup: Drop transferMode.dlna.org header
	  Leave it to the application to decide on the header. No header at all
	  is better than having the wrong header as DLNA mandates that a missing
	  header has to be tolerated while a wrong header is an error.
	  https://bugzilla.gnome.org/show_bug.cgi?id=676020

2012-04-07 09:52:09 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngdec.h:
	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	  png: Port to base video classes
	  Conflicts:
	  ext/libpng/gstpngdec.c
	  ext/libpng/gstpngdec.h
	  ext/libpng/gstpngenc.c
	  ext/libpng/gstpngenc.h
	  Reverted to 0.10, needs to be ported again.

2012-05-27 00:02:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/matroska/matroska-read-common.c:
	  flv, matroska: don't use GstStructure API on tag lists

2012-05-26 11:57:16 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtp/gstrtpmp2tdepay.c:
	  rtpmp2tdepay: Only output integral mpeg-ts packets
	  From RFC 2250
	  2. Encapsulation of MPEG System and Transport Streams
	  ...
	  For MPEG2 Transport Streams the RTP payload will contain an integral
	  number of MPEG transport packets.  To avoid end system
	  inefficiencies, data from multiple small MTS packets (normally fixed
	  in size at 188 bytes) are aggregated into a single RTP packet.  The
	  number of transport packets contained is computed by dividing RTP
	  payload length by the length of an MTS packet (188).
	  ....
	  Since it needs to contain "an integral number of MPEG transport packets", a
	  simple fix is to check that's the case, and strip off any leftover data.
	  Fixes #676799
	  Conflicts:
	  gst/rtp/gstrtpmp2tdepay.c

2012-05-24 20:43:16 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: make sure all selectors are performed on the same thread
	  When we are using a dedicated thread to run the main run loop we
	  must make sure that all selectors are performed on this same thread.
	  For instance if performSelectorOnMainThread is called from the real
	  main thread, it will not go through the message queue and will be
	  executed from the real main thread. By forcing the target thread,
	  we ensure that all functions will be called either from the real
	  main thread when the main run loop is running or from our thread
	  spinning the main loop.

2012-05-24 16:09:54 +0200  Mathias Hasselmann <mathias.hasselmann at gmx.de>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: remove framerate
	  The jpeg decoder doesn't need/care about the framerate to so it should
	  not be in the caps.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676302

2012-05-24 13:08:35 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: start the loop before calling [gstview haveSuperview]
	  ...as haveSuperview requires the mainloop to be running

2012-05-24 13:08:13 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix indentation

2012-05-22 16:47:36 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* sys/osxvideo/Makefile.am:
	  osxvideosink: enable running the cocoa main runloop in a thread

2012-05-22 16:45:28 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: add code to optionally run the cocoa main runloop in a separate thread
	  Add a little hack to run the cocoa main runloop from a separate thread _when_
	  the main runloop is not being run (which means that the app doesn't use cocoa).
	  Runloops are thread specific, so the hack boils down to getting the runloop for
	  the main thread and setting it as the runloop for our dedicated thread.

2012-05-22 16:32:53 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: reset app_started to FALSE when shutting down

2012-05-22 14:49:17 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: rename cocoa runloop helper funcs

2012-05-22 14:26:13 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: don't create application menus

2012-05-16 21:52:45 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: reset the embed property for backward compatilibity

2012-05-16 21:12:22 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix navigation when force-aspect-ratio is activated

2012-05-16 18:52:45 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: add force-aspect-ratio property

2012-05-14 18:01:02 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: start internal window if no view is provided

2012-05-14 14:27:58 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: implement the navigation interface

2012-05-11 18:24:08 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osvideosink: create, destroy, resize and draw from the main thread

2012-04-19 08:37:28 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: increase NEWSEGMENT accuracy after seeking
	  demux->common.segment is populated during seek handling with the target
	  start/stop positions. Don't override them when sending out a NEWSEGMENT.
	  Conflicts:
	  gst/matroska/matroska-demux.c

2012-04-19 08:31:00 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: don't discard the incoming seek segment on push based seeking
	  The incoming seek segment was being discarded leading to push based seeking
	  being potentially inaccurate.

2012-05-23 18:12:24 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* common:
	  common: Update so the plugin scanner changes are included
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676674

2012-05-23 18:07:35 +0200  Sebastian Rasmussen <sebrn@axis.com>

	* configure.ac:
	  configure: suppress some warnings when debug is disabled
	  Warnings about unused variables should be suppressed if core has the
	  debug system disabled.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676671

2012-05-24 09:29:25 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtph264pay.c:
	  rtp: fix build issue in gstrtph264pay.c

2012-05-21 12:17:35 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Add unrestricted caps
	  If there are no profile restrictions downstream, return caps with
	  profile=constrained-baseline in the first structure and append
	  unrestricted caps as the last structure.
	  Fixes bug #672019

2012-05-24 09:57:31 +0200  Maria Giovanna Chiossa <mariagiovanna.chiossa at selexelsag.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: add the Scale header when needed
	  Setting GST_SEEK_FLAG_SKIP when sending a seek event in rtspsrc should
	  set the "Scale" field in the rtsp PLAY header.
	  Because the boolean "src->skip" is set after the call, "Speed" instead
	  of "Scale" is always set. Move the assignment before issuing the _play
	  request.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676618

2012-05-17 16:23:59 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix the sample pipeline.

2012-05-22 12:35:04 +0400  Anton Novikov <random.plant@gmail.com>

	* gst/icydemux/gsticydemux.c:
	  icydemux: warning if setting srcpad caps fails

2012-05-22 12:35:29 +0400  Anton Novikov <random.plant@gmail.com>

	* gst/icydemux/gsticydemux.c:
	  icydemux: activate srcpad before setting caps
	  Before gst_pad_set_active() is called, the pad has
	  FLUSHING flag set, so setting the caps fails

2012-05-22 13:46:27 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* ext/Makefile.am:
	* ext/libmng/Makefile.am:
	* ext/libmng/gstmng.c:
	* ext/libmng/gstmng.h:
	* ext/libmng/gstmngdec.c:
	* ext/libmng/gstmngdec.h:
	* ext/libmng/gstmngenc.c:
	* ext/libmng/gstmngenc.h:
	  mng: remove ext/libmng
	  Port to 0.10 was never finished.
	  Interest was lost.
	  https://bugzilla.gnome.org/show_bug.cgi?id=324364

2012-05-18 16:37:04 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/avi/gstavimux.c:
	  avimux: fix assertion when handling a date tag as a string
	  Date tags are GDate, not strings. Add a special case to convert
	  it to the exif date format representation in string to avoid
	  the assertion

2012-05-21 11:47:07 +0200  Sjoerd Simons <sjoerd@luon.net>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Listen to source output events, not sink input

2012-05-18 12:53:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp2tpay.c:
	  rtpmp2tpay: respect mtu and packet boundaries
	  See #659915.

2012-05-18 11:10:46 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpeg: Remove dead code
	  Conflicts:
	  ext/jpeg/gstjpegdec.c

2012-05-18 11:05:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Fix compilation

2012-05-18 11:02:52 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: When dropping frames on EOS, flush out data
	  Cleaner way of handling stray data

2012-05-17 09:34:03 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: Remove unused variable
	  Conflicts:
	  ext/jpeg/gstjpegdec.c

2012-05-17 09:33:18 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Only parse for SOI when we didn't see it before

2012-05-17 09:31:41 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Remember if we saw SOI and handle stray data on EOS

2012-05-15 20:58:25 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Allow U and V components to use different quant tables if they contain the same data
	  This allows some cameras (Logitech C920) that specify different quant
	  tables but both with the same data, to work.
	  Bug reported by Robert Krakora

2012-05-14 15:51:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix possible data corruption after seeking
	  Consider a downstream element that may issue seeks in very short
	  succession (e.g. queue2), depending on the access pattern of
	  the downstream element (e.g. qtdemux with audio/video chunks
	  interleaved so that there's always a sizeable gap between the
	  current chunks for each stream). In this case, queue2 will maintain
	  two ranges, and even when it serves a chunk from memory, it will
	  switch ranges and make souphttpsrc seek to the end of the available
	  data for that range, assuming that that's where we'll want to
	  continue reading from next.
	  This may lead to the following seek request pattern:
	  - source reading position A
	  - seek to B
	  - now reading position still A, requested_postion is B
	  - streaming thread to be restarted to continue from B
	  - seek to A, before streaming thread had time to do the seek
	  - do_seek() now sees reading position == seek position and
	  returns early.
	  - however, requested position is still B from the earlier
	  seek request
	  - streaming thread starts up, sees that a seek to B is pending
	  and requests data from B from the server, while the GstBaseSrc
	  segment has of course been updated/reset to position A, which
	  was the last seek request.
	  - we will now send data for position B and pretend that's the
	  data from position A (via the newsegment event, etc.)
	  - this causes data corruption
	  Reproducible doing seek-emulated fast-forward/backward on 006648.

2012-05-16 09:12:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Require core/base 0.11.91

2012-01-13 18:09:50 -0500  Matej Knopp <matej.knopp@gmail.com>

	* .gitignore:
	  .gitignore: add visual studio IDE files and OS X .DS_Store files
	  https://bugzilla.gnome.org/show_bug.cgi?id=667899

2012-05-03 09:32:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpeg: Port to 0.11 again

2012-04-06 12:13:24 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpeg: Port jpegdec/jpegenc to base video classes
	  Conflicts:
	  ext/jpeg/gstjpegdec.c
	  ext/jpeg/gstjpegdec.h
	  ext/jpeg/gstjpegenc.c
	  ext/jpeg/gstjpegenc.h
	  Reverted to 0.10 versions for now, next port again.

2012-05-13 19:21:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-annodex.xml:
	* ext/Makefile.am:
	* ext/annodex/Makefile.am:
	* ext/annodex/gstannodex.c:
	* ext/annodex/gstannodex.h:
	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmldec.h:
	* ext/annodex/gstcmmlenc.c:
	* ext/annodex/gstcmmlenc.h:
	* ext/annodex/gstcmmlparser.c:
	* ext/annodex/gstcmmlparser.h:
	* ext/annodex/gstcmmltag.c:
	* ext/annodex/gstcmmltag.h:
	* ext/annodex/gstcmmlutils.c:
	* ext/annodex/gstcmmlutils.h:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/cmmldec.c:
	* tests/check/elements/cmmlenc.c:
	  annodex: remove annodex plugin and CMML elements
	  This never really took off and is most likely completely
	  unused. If there is still a need for this, it should
	  probably be done differently, perhaps inside oggdemux/mux.

2012-05-13 16:59:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  Back to development

=== release 0.11.91 ===

2012-05-13 16:31:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* common:
	* configure.ac:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.11.91

2012-05-13 16:30:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2012-05-13 15:56:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From dc70203 to 3429ba6

2012-05-09 15:14:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: only send flush-stop if it was a flushing seek

2012-05-09 12:54:11 +0200  Peter Seiderer <ps.report@gmx.net>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: fix v4l2_std_id logging
	  input.std is of type v4l2_std_id which is defined as 64-bit unsigned integer.
	  Casting to uint means the higher bits, wich are used for the private video
	  standards of the TI video capture/display driver for example, are lost.

2012-05-09 12:24:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: must send flush-stop after acquiring the stream lock
	  Otherwise the streaming thread might just keep on going and we
	  might never get the stream lock.

2012-05-09 11:15:21 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: port seeking code to 0.11

2012-05-08 19:07:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: add support for seeks
	  Useful for e.g. filesrc ! rndbuffersize ! queue2 ! ...

2012-05-08 18:45:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: send SEGMENT event before pushing buffers
	  Conflicts:
	  gst/debugutils/rndbuffersize.c

2012-05-09 11:15:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/interleave/interleave.c:
	  interleave: fix compilation again

2012-01-13 10:49:43 +0100  Pascal Buhler <pabuhler@cisco.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: creation should be signaled before validation
	  https://bugzilla.gnome.org/show_bug.cgi?id=667850

2012-05-04 15:20:47 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: do not proxy our filter caps downstream on caps queries
	  Downstream likely won't accept video/x-raw and the caps query
	  will return EMPTY caps. Instead, create a copy of the caps that
	  has all structure names replaced by 'image/jpeg'
	  Simple pipeline that shows the problem:
	  gst-launch-1.0 videotestsrc num-buffers=1 ! "video/x-raw, \
	  width=(int)640, height=(int)480" ! videoscale ! jpegenc ! \
	  "image/jpeg, width=(int)800, height=(int)600" ! filesink \
	  location=/tmp/image.jpg

2012-05-02 21:17:43 +0200  Alban Browaeys <prahal@yahoo.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: set layout=interleaved on raw audio caps
	  This fixes a not-negotiated error at least on mov files with
	  twos audio with two channels and video dvcp. As playbin and gst-launch
	  sample coming from the qtdemux.c file uses audioconvert and the latter
	  require format interleaved.
	  https://bugzilla.gnome.org/show_bug.cgi?id=675326

2012-05-02 21:49:56 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* sys/waveform/Makefile.am:
	  waveform: No more gstinterfaces
	  Fixes #675319

2012-05-02 20:14:24 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* sys/directsound/Makefile.am:
	  directsound: No more gstinterfaces
	  Fixes #675319

2012-05-01 18:58:03 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: change sink pad template name from sink_%d to sink_%u

2012-04-30 11:00:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/interleave/interleave.c:
	  interleave: handle EOS on all pads
	  When all pads go to EOS immediately, we are not negotiated and our collected
	  function is called (without any available data). Handle this case gracefully.
	  Conflicts:
	  gst/interleave/interleave.c

2012-04-30 10:59:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/interleave/interleave.c:
	  interleave: improve debugging

2012-05-01 13:31:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Update for basesrc API changes

2012-04-30 23:57:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: don't set up stuff before the input and output formats are known
	  Fixes crash on startup.

2012-04-30 14:09:23 +0200  Peter Seiderer <ps.report@gmx.net>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: don't write stream header twice for first file

2012-04-30 13:32:41 +0200  Peter Seiderer <ps.report@gmx.net>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: fix buffer list size calculation in render_list
	  Fix uninitialized 'size' variable in call to gst_buffer_list_foreach().

2012-04-30 21:58:00 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/multifile/gstmultifilesrc.c:
	  multifile: unnecessary size check

2012-04-30 21:30:56 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/avi/gstavidemux.c:
	  avi: fix build errors
	  fix redundant declarations
	  and also style/indent issues

2012-04-26 12:47:27 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: implement forward snapping keyframe seeking
	  Requires an index.

2012-04-26 12:46:11 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: implement forward snapping keyframe seeking
	  In pull mode with an index.

2012-04-28 23:14:24 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/matroskamux.c:
	  tests: fix matroskamux unit test after media type changes

2012-04-28 19:57:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	  matroska: update for media type changes

2012-04-24 16:08:47 +0200  idc-dragon <idc-dragon at gmx.de>

	* gst/rtp/gstrtpceltdepay.c:
	  celtdepay: calculate size correctly
	  The summation was done wrong, causing the de-payloader to exit its loop too
	  early, before all frames are processed.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674472

2012-04-24 15:57:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: improve debug

2012-04-24 15:34:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: start unmuted when requested
	  When we explicitely set the mute property to FALSE, connect to pulseaudio with
	  the PA_STREAM_START_UNMUTED flag set, otherwise pulseaudio will use its
	  previously used value (which might start the stream muted).
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=672401

2012-04-25 09:41:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: improve timestamp code
	  Sample the pipeline clock and device clock closer to eachother to reduce jitter.
	  Don't subtract the frame duration from the timestamp when we can use the device
	  timestamps.
	  Assume a delay of 1 frame in read-write mode.

2012-04-24 12:37:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: use driver timestamps
	  Use the drive timestamps for timestamping outgoing buffers.

2012-04-23 18:01:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Improve buffer management
	  Query the amount of available buffers when doing set_config(). This allows us to
	  configure the parent bufferpool with the number of buffers to preallocate.
	  Keep track of the provided allocator and use it when we need to allocate a
	  buffer in RW mode.
	  When we are can not allocate the requested max_buffers amount of buffers, make
	  sure we keep 2 buffers around in the pool and copy them into an output buffer.
	  This makes sure that we always have a buffer to capture into. We also need to
	  detect those copied buffers and unref them when they return to the pool.

2012-04-23 16:51:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: free the queued buffers
	  Only free the queued buffers that we keep track of in our buffer array. for rw
	  io-mode, we do allocate buffers but we don't keep track of them in the buffer
	  array.

2012-04-23 16:10:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: mark memory as no-share
	  We don't support sharing our mmapped memory so mark it as NO_SHARE.

2012-04-23 16:09:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2: remove old unused file

2012-04-23 13:32:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2: remove unused function

2012-04-11 12:42:17 +0100  Bastien Nocera <hadess@hadess.net>

	* ext/soup/gstsouphttpsrc.c:
	  soup: Handle icy and icyx URI schemes
	  As handled by QuickTime (for icy), and Orban/Coding Technologies
	  AAC/aacPlus Player (for icyx). See also:
	  https://bugzilla.gnome.org/show_bug.cgi?id=394207
	  https://bugzilla.gnome.org/show_bug.cgi?id=403285
	  https://bugzilla.gnome.org/show_bug.cgi?id=673899

2012-04-23 10:03:19 +0300  Mart Raudsepp <mart.raudsepp@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  docs: Add Since tag for new GstV4l2Src::prepare-format signal

2012-04-23 10:07:12 +0200  Chris Pankow <kain2396@gmail.com>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Fix time-domain convolution for multichannel input
	  Fixes bug #674025.

2012-04-21 11:08:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* po/POTFILES.in:
	  po: remove some more non-existent files from the list

2012-04-21 10:05:45 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* po/POTFILES.in:
	  po: Remove non-existent potfiles from the list
	  Fixes #674518

2012-04-20 18:13:15 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/icles/test-oss4.c:
	  tests: oss4: limit test scope

2012-04-20 18:13:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* sys/oss4/Makefile.am:
	* sys/oss4/oss4-audio.c:
	* sys/oss4/oss4-audio.h:
	* sys/oss4/oss4-mixer-enum.c:
	* sys/oss4/oss4-mixer-enum.h:
	* sys/oss4/oss4-mixer-slider.c:
	* sys/oss4/oss4-mixer-slider.h:
	* sys/oss4/oss4-mixer-switch.c:
	* sys/oss4/oss4-mixer-switch.h:
	* sys/oss4/oss4-mixer.c:
	* sys/oss4/oss4-mixer.h:
	* sys/oss4/oss4-property-probe.c:
	* sys/oss4/oss4-property-probe.h:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-sink.h:
	* sys/oss4/oss4-source.c:
	* sys/oss4/oss4-source.h:
	  oss4: port to 0.11

2012-04-20 18:12:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* sys/oss/Makefile.am:
	* sys/oss/gstossaudio.c:
	* sys/oss/gstosshelper.c:
	* sys/oss/gstosshelper.h:
	* sys/oss/gstossmixer.c:
	* sys/oss/gstossmixer.h:
	* sys/oss/gstossmixerelement.c:
	* sys/oss/gstossmixerelement.h:
	* sys/oss/gstossmixertrack.c:
	* sys/oss/gstossmixertrack.h:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss/gstosssrc.h:
	  oss: port to 0.11

2012-04-20 16:49:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: first activate pad then set caps

2012-04-20 13:35:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: set caps on srcpad
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674219

2012-04-19 14:16:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: update for video api change

2012-04-19 12:38:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix compilation on older v4l2
	  Fix compilation on systems where the H264 format is not defined.

2012-04-19 12:20:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdec.c:
	* ext/raw1394/Makefile.am:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/y4m/gsty4mencode.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  video: Update for libgstvideo API changes

2012-04-19 08:27:01 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: Allow mpeg-ts cameras to negociate format
	  This removes an ugly hack until the reason for the hack can be documented

2012-04-19 09:50:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: Fix merge

2012-04-19 09:40:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: Rename pre-set-format signal to prepare-format

2012-04-16 22:08:21 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: Add H264 encoded stream support to the caps
	  This is not enough to properly support H264 cameras, but it will
	  allow an H264 stream to be generated by v4l2src using the default
	  settings of the camera. If used with the pre-set-format signal, the
	  H264 encoder can be fully configured.
	  Conflicts:
	  sys/v4l2/gstv4l2object.c

2012-04-16 22:06:21 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* sys/v4l2/.gitignore:
	* sys/v4l2/gstv4l2-marshal.list:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: Adding a pre-set-format signal
	  In order to support UVC H264 encoding cameras, an H264 Probe&Commit
	  must happen before the normal v4l2 set-format. This new signal is
	  meant to allow an external application or bin to do it.
	  It also serves to expose the file descriptor used by v4l2src in case
	  some custom ioctls need to be called.
	  Conflicts:
	  sys/v4l2/Makefile.am
	  sys/v4l2/gstv4l2src.c
	  sys/v4l2/v4l2src_calls.c

2012-04-18 17:09:45 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* ext/raw1394/gst1394probe.c:
	* ext/raw1394/gst1394probe.h:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	  dv1394: port to 0.11

2012-04-17 15:14:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttextoverlay.h:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	* gst/multipart/multipartmux.c:
	* gst/multipart/multipartmux.h:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmpte.h:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	* gst/videomixer/videomixer2pad.h:
	  collectpads2: rename to collectpads

2012-04-16 16:37:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/flv/gstflvmux.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstqtmux.c:
	* gst/matroska/matroska-mux.c:
	* gst/smpte/gstsmpte.c:
	* gst/videomixer/videomixer2.c:
	  misc: chain up to collectpads event handler

2012-04-16 09:09:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 6db25be to dc70203

2012-04-15 22:49:47 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/shout2/gstshout2.c:
	  shout2: update for ogg media type changes

2012-04-13 16:54:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmpte.h:
	  smpte: use some more boilerplate

2012-04-13 16:54:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flx/gstflxdec.c:
	  flxdec: improve segment handling
	  ... to send a proper TIME segment downstream.

2012-04-13 16:54:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* gst/flx/gstflxdec.c:
	* gst/flx/gstflxdec.h:
	  flxdec: port to 0.11

2012-04-13 16:54:42 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: adjust to deprecated GMutex setup

2012-04-13 16:54:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: port to 0.11

2012-04-13 16:54:31 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/smpte/gstsmptealpha.c:
	  alpha, smpte: adjust to removed color-matrix caps field

2012-04-13 16:27:34 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* sys/v4l2/Makefile.am:
	  v4l2: ensure autogenerated files are created
	  The tuner marshal and enumtypes are autogenerated, and they need
	  to be created before the compilation of gstv4l2tuner.c
	  This patch adds the automake instruction for ensuring the
	  autogeneration of those files previous the compilation.

2012-04-13 13:41:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* autogen.sh:
	* configure.ac:
	  configure: Modernize autotools setup a bit
	  Also we now only create tar.bz2 and tar.xz tarballs.

2012-04-13 13:37:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 464fe15 to 6db25be

2012-04-13 13:04:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* ext/pulse/Makefile.am:
	* ext/pulse/plugin.c:
	* ext/pulse/pulsemixer.c:
	* ext/pulse/pulsemixer.h:
	* ext/pulse/pulsemixerctrl.c:
	* ext/pulse/pulsemixerctrl.h:
	* ext/pulse/pulsemixertrack.c:
	* ext/pulse/pulsemixertrack.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	* gst/rtsp/Makefile.am:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/gstv4l2videooverlay.c:
	* sys/v4l2/gstv4l2videooverlay.h:
	* sys/v4l2/tuner-marshal.list:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tuner.h:
	* sys/v4l2/tunerchannel.c:
	* sys/v4l2/tunerchannel.h:
	* sys/v4l2/tunernorm.c:
	* sys/v4l2/tunernorm.h:
	* tests/check/Makefile.am:
	* tests/examples/pulse/Makefile.am:
	* tests/icles/Makefile.am:
	* tests/icles/v4l2src-test.c:
	  Update everything for the removal of the interface library and mixer/tuner interfaces

2012-04-12 15:50:16 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtp: Use unchecked variant of GstByteWriter where applicable
	  The size was checked before

2012-04-12 15:49:44 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-demux.c:
	  matroska: Check return value of GstByteReader/Writer

2012-04-12 15:48:57 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/isomp4/atoms.c:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	  isomp4: Check return value of GstByteWriter
	  And use unchecked variant of GstByteReader where applicable

2012-04-12 15:48:00 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Use unchecked variant of GstByteReader
	  We know there's at least 7 bytes (checked above)

2012-04-12 15:47:49 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avi: Check return value of GstByteWriter

2012-04-12 15:47:24 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: Check return value of GstBitReader/GstByteReader

2012-04-12 11:57:59 +0100  uraeus <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  Add interleave plugin to spec file

2012-04-12 11:19:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  Back to development

=== release 0.11.90 ===

2012-04-12 10:27:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* gst/deinterlace/tvtime-dist.c:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videomixer/blendorc-dist.c:
	* win32/common/config.h:
	  Release 0.11.90

2012-04-12 10:26:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2012-04-11 00:19:30 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* ext/jpeg/gstjpegenc.c:
	  Fix format string
	  Fixes #673859

2012-04-11 00:19:16 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* sys/waveform/gstwaveformsink.c:
	  Remove unused variable
	  Fixes #673859

2012-04-10 11:57:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  Merge remote-tracking branch 'origin/0.10'
	  Conflicts:
	  gst/flv/gstflvdemux.c
	  gst/matroska/matroska-demux.c

2012-04-10 11:37:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: some more segment handling tweaking

2012-04-10 00:51:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairorender.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttimeoverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libmng/gstmngdec.c:
	* ext/libmng/gstmngenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/mikmod/gstmikmod.c:
	* ext/pulse/pulsemixer.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/gstcapsdebug.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/equalizer/gstiirequalizernbands.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/interleave/deinterleave.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	* gst/median/gstmedian.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/spectrum/gstspectrum.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/videomixer/videomixer2.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* gst/y4m/gsty4mencode.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstossmixerelement.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-mixer.c:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	  Use new gst_element_class_set_static_metadata()

2012-04-10 00:47:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/twolame/gsttwolamemp2enc.c:
	  Use new gst_element_class_set_static_metadata()

2012-04-10 00:47:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  Use new gst_element_class_set_static_metadata()

2012-04-09 12:55:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/pipelines/simple-launch-lines.c:
	  tests: disable simple smokeenc/dec launch lines test
	  Disable test for smoke elements, which aren't ported yet
	  (and maybe shouldn't be ported).

2012-04-09 00:14:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	* tests/check/elements/interleave.c:
	  interleave: make channel-poisitions property a GValueArray again
	  Or perhaps it should just be a guint64 channel mask, which would
	  be nicer in C, but more awkward for bindings (even more so since
	  we can't add a flags type for it, since that only supports guint
	  size flags). Fixes wavenc unit test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=669643

2012-04-06 16:03:47 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: cleanly initialize and set needed segment
	  Fixes #673165.

2012-04-05 17:17:22 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Fix threading issue in index handling

2012-04-06 09:13:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't use static variables to hold index associations
	  This not really threadsafe in any way.

2012-04-05 19:17:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/flvmux.c:
	* tests/check/elements/interleave.c:
	  tests: make few tests more valgrind-friendly

2012-04-05 19:17:42 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* tests/check/elements/deinterleave.c:
	  (de)interleave: fix ported unit test and enable as ported

2012-04-05 19:17:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/cmmldec.c:
	  tests: cmmldec: adjust to tag events no longer posted on bus by element

2012-04-05 19:17:29 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  updsrc: clear error

2012-04-05 18:42:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 7fda524 to 464fe15

2012-04-05 18:02:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/gststereo.c:
	  gst: Update for GST_PLUGIN_DEFINE() API changes

2012-04-05 18:02:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/dtmf/gstdtmf.c:
	  gst: Update for GST_PLUGIN_DEFINE() API changes

2012-04-05 17:40:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/twolame/gsttwolamemp2enc.c:
	  gst: Update for GST_PLUGIN_DEFINE() API changes

2012-04-05 17:40:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/plugin.c:
	  gst: Update for GST_PLUGIN_DEFINE() API changes

2012-04-05 17:36:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/annodex/gstannodex.c:
	* ext/cairo/gstcairo.c:
	* ext/dv/gstdv.c:
	* ext/flac/gstflac.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/jack/gstjack.c:
	* ext/jpeg/gstjpeg.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libmng/gstmng.c:
	* ext/libpng/gstpng.c:
	* ext/mikmod/gstmikmod.c:
	* ext/pulse/plugin.c:
	* ext/raw1394/gst1394.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsoup.c:
	* ext/speex/gstspeex.c:
	* ext/taglib/gsttaglibplugin.c:
	* ext/wavpack/gstwavpack.c:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audiofx.c:
	* gst/audioparsers/plugin.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautodetect.c:
	* gst/avi/gstavi.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/gstdebug.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/effectv/gsteffectv.c:
	* gst/equalizer/gstiirequalizer.c:
	* gst/flv/gstflvdemux.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/interleave/plugin.c:
	* gst/isomp4/isomp4-plugin.c:
	* gst/law/alaw.c:
	* gst/law/mulaw.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska.c:
	* gst/median/gstmedian.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifile.c:
	* gst/multipart/multipart.c:
	* gst/replaygain/replaygain.c:
	* gst/rtp/gstrtp.c:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtsp/gstrtsp.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/plugin.c:
	* gst/spectrum/gstspectrum.c:
	* gst/udp/gstudp.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/videofilter/plugin.c:
	* gst/videomixer/videomixer2.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* gst/y4m/gsty4mencode.c:
	* sys/directsound/gstdirectsoundplugin.c:
	* sys/oss/gstossaudio.c:
	* sys/oss4/oss4-audio.c:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudio.c:
	* sys/v4l2/gstv4l2.c:
	* sys/waveform/gstwaveformplugin.c:
	* sys/ximage/gstximagesrc.c:
	  gst: Update for GST_PLUGIN_DEFINE() API changes

2012-04-05 13:26:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Update version to 0.11.89.1

2012-04-04 20:06:58 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: ensure initialized test buffer memory

2012-04-04 14:41:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/dtmf/Makefile.am:
	  gst: Update versioning

2012-04-04 14:38:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/twolame/Makefile.am:
	  gst: Update versioning

2012-04-04 14:38:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/Makefile.am:
	  gst: Update versioning

2012-04-04 14:33:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/version.entities.in:
	* ext/aalib/Makefile.am:
	* ext/cairo/Makefile.am:
	* ext/dv/Makefile.am:
	* ext/flac/Makefile.am:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/jack/Makefile.am:
	* ext/jpeg/Makefile.am:
	* ext/libcaca/Makefile.am:
	* ext/libpng/Makefile.am:
	* ext/pulse/Makefile.am:
	* ext/raw1394/Makefile.am:
	* ext/soup/Makefile.am:
	* ext/speex/Makefile.am:
	* ext/taglib/Makefile.am:
	* ext/wavpack/Makefile.am:
	* gst-plugins-good.spec.in:
	* gst/alpha/Makefile.am:
	* gst/apetag/Makefile.am:
	* gst/audiofx/Makefile.am:
	* gst/audioparsers/Makefile.am:
	* gst/auparse/Makefile.am:
	* gst/avi/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debugutils/Makefile.am:
	* gst/deinterlace/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/equalizer/Makefile.am:
	* gst/flv/Makefile.am:
	* gst/icydemux/Makefile.am:
	* gst/id3demux/Makefile.am:
	* gst/interleave/Makefile.am:
	* gst/isomp4/Makefile.am:
	* gst/law/Makefile.am:
	* gst/level/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/multifile/Makefile.am:
	* gst/replaygain/Makefile.am:
	* gst/rtp/Makefile.am:
	* gst/rtpmanager/Makefile.am:
	* gst/rtsp/Makefile.am:
	* gst/shapewipe/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/spectrum/Makefile.am:
	* gst/videobox/Makefile.am:
	* gst/videocrop/Makefile.am:
	* gst/videofilter/Makefile.am:
	* gst/videomixer/Makefile.am:
	* gst/wavenc/Makefile.am:
	* gst/wavparse/Makefile.am:
	* gst/y4m/Makefile.am:
	* pkgconfig/Makefile.am:
	* pkgconfig/gstreamer-plugins-good-uninstalled.pc.in:
	* sys/directsound/Makefile.am:
	* sys/oss/Makefile.am:
	* sys/oss4/Makefile.am:
	* sys/osxaudio/Makefile.am:
	* sys/osxvideo/Makefile.am:
	* sys/sunaudio/Makefile.am:
	* sys/v4l2/Makefile.am:
	* sys/waveform/Makefile.am:
	* sys/ximage/Makefile.am:
	* tests/check/Makefile.am:
	* tests/examples/audiofx/Makefile.am:
	* tests/examples/cairo/Makefile.am:
	* tests/examples/pulse/Makefile.am:
	* tests/examples/spectrum/Makefile.am:
	* tests/icles/Makefile.am:
	  gst: Update versioning

2012-04-04 12:10:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	  Merge remote-tracking branch 'origin/0.10'
	  Conflicts:
	  gst/matroska/matroska-demux.c
	  gst/matroska/matroska-mux.c
	  gst/matroska/matroska-read-common.c
	  gst/matroska/matroska-read-common.h

2012-04-03 18:36:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: plug template caps leak

2012-04-03 11:50:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: avi only knows about DTS
	  Only set DTS on outgoing buffers unless we have a keyframe and then we can set
	  the PTS to DTS as well.

2012-04-02 23:35:43 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/matroska/matroska-read-common.c:
	  mkv: port toc changes to 0.11

2012-04-02 23:18:00 +0200  Stefan Sauer <ensonic@users.sf.net>

	  Merge branch '0.10'
	  Conflicts:
	  gst/matroska/matroska-demux.c
	  gst/matroska/matroska-mux.c
	  gst/matroska/matroska-read-common.c
	  gst/matroska/matroska-read-common.h

2012-03-29 23:22:28 +0400  Alexander Saprykin <xelfium@gmail.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroska: add GstToc support for muxer

2012-03-29 23:12:13 +0400  Alexander Saprykin <xelfium@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroska: add support for GstToc in demuxer

2012-03-29 23:05:14 +0400  Alexander Saprykin <xelfium@gmail.com>

	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: add chapter support in GstMatroskaReadCommon

2012-04-02 13:00:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/goom2k1/lines.c:
	  goom2k1: Fix 'may be used uninitialized in this function' compiler warning

2012-04-02 11:13:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	  use transform_ip_on_passthrough

2012-03-31 15:43:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/videomixer/videomixer2.c:
	* tests/check/elements/equalizer.c:
	* tests/examples/equalizer/demo.c:
	* tests/icles/equalizer-test.c:
	  update for child proxy api change

2012-03-30 18:13:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/atoms.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsrc.c:
	* gst/y4m/gsty4mencode.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/ximage/ximageutil.c:
	* tests/check/elements/deinterleave.c:
	* tests/check/elements/interleave.c:
	  update for buffer api change

2012-03-30 12:53:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	* ext/speex/gstspeexenc.h:
	  speexenc: Use new gst_audio_encoder_set_headers() API

2012-03-30 12:18:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	* ext/speex/gstspeexenc.c:
	* ext/wavpack/gstwavpackenc.c:
	  ext: Update for GstAudioEncoder API changes

2012-03-29 23:22:28 +0400  Alexander Saprykin <xelfium@gmail.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroska: add GstToc support for muxer

2012-03-29 23:12:13 +0400  Alexander Saprykin <xelfium@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroska: add support for GstToc in demuxer

2012-03-29 23:05:14 +0400  Alexander Saprykin <xelfium@gmail.com>

	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: add chapter support in GstMatroskaReadCommon

2012-03-29 17:22:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/pipelines/wavpack.c:
	  tests: wavpack: fewer buffers are also adequate and more convenient

2012-03-29 17:22:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/videocrop.c:
	  tests: videocrop: unmap video frame and unref caps

2012-03-29 17:22:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/audiowsincband.c:
	  tests: audiowsincband: unmap examined output buffers

2012-03-29 17:21:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: plug ref leak

2012-03-29 17:21:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: fix supported template caps and sample processing

2012-03-29 17:21:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: plug structure leak

2012-03-29 16:04:26 +0100  uraeus <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  Update spec file with latest ported plugins

2012-03-29 15:03:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	  Merge remote-tracking branch 'origin/0.10'
	  Conflicts:
	  configure.ac

2012-03-28 16:26:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  tests: tagschecking: muxers need TIME format

2012-03-28 16:26:15 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/pipelines/flacdec.c:
	  tests: flacdec: needs flacparse nowadays

2012-03-28 14:49:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackenc.c:
	  wavpackenc: query downstream for BYTE seeking support

2012-03-28 14:48:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: query downstream for BYTE seeking support

2012-03-28 14:46:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: clean up obsolete log statement

2012-03-28 12:49:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/mikmod/gstmikmod.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/avi/gstavimux.c:
	* gst/flv/gstflvmux.c:
	* gst/icydemux/gsticydemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/y4m/gsty4mencode.c:
	* tests/check/elements/parser.c:
	  update for buffer changes

2012-03-28 12:16:45 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/audiodynamic.c:
	  tests: audiodynamic: correctly port original test to mind in place transform

2012-03-28 11:05:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	  audiofx: more adjustment to changed semantics of audiofilter _setup method

2012-03-28 11:10:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/audiofirfilter.c:
	  tests: audiofirfilter: negotiate the intended raw audio format

2012-03-27 18:41:45 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audioparsers/gstwavpackparse.c:
	  wavpackparse: init datastructure

2012-03-27 17:18:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstwarp.c:
	  effectv: fix strides

2012-03-27 16:41:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-encode.c:
	* gst/matroska/matroska-demux.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/gstsmpte.c:
	* sys/oss/gstosssink.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/ximage/gstximagesrc.c:
	* tests/check/elements/qtmux.c:
	  caps: improve caps handling
	  Avoid caps copy and leaks

2012-03-27 14:04:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/icydemux.c:
	  tests: icydemux: activate internal test helper src pad

2012-03-27 12:44:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: update for get_param
	  Remove const from the GstCaps.
	  Plug some GstStructure leaks

2012-03-27 00:02:08 +0300  Raimo Järvi <raimo.jarvi@gmail.com>

	* configure.ac:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	  udp: Fix compiling with mingw.
	  https://bugzilla.gnome.org/show_bug.cgi?id=672880

2012-03-26 18:31:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rgvolume.c:
	  tests: replaygain: misc compatibility fixes
	  Discard caps event when checking for and counting various tag events,
	  and remove all testing of 8 bits depth in 16 bits width format since
	  it no longer exists.

2012-03-26 18:28:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtpbin.c:
	  tests: rtp: misc compatibiliy fixes
	  ... such as always setting pad caps and providing needed caps fields.

2012-03-26 18:26:40 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/videofilter.c:
	  tests: videofilter: ensure initial segment event

2012-03-26 18:25:28 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	  shapewipe: proper video info and frame management
	  ... particularly since each incoming pad has a distinct format.

2012-03-26 18:24:08 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: ensure output caps are set when pushing output data
	  ... even if some SPS/PPS has not passed by yet.

2012-03-26 18:22:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	  videofilter: avoid holding object lock when calling basetransform function

2012-03-26 18:22:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix some lock management
	  ... to avoid trying to take a non-recursive lock twice.

2012-03-26 18:21:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	  rtpL16(de)pay: fix raw audio format in template caps

2012-03-26 18:20:40 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/replaygain/gstrganalysis.c:
	  replaygain: also still post the results of the analysis

2012-03-26 15:59:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: don't error in shutdown
	  Don't log with the ERROR category when we are stopping because we are shutting
	  down.
	  Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=672824

2012-03-26 15:51:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: fix latency

2012-03-26 15:30:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: called base class start
	  Chain up to the base class start method so that metadata is properly tagged.
	  Remove an unused variable.
	  fixes: https://bugzilla.gnome.org/show_bug.cgi?id=672813

2012-03-26 12:12:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Replace master with 0.11

2012-03-25 00:00:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	  gdkpixbufoverlay: add "alpha" property to set alpha of overlay image
	  .. or turn the overlay off by setting alpha to 0.0

2012-03-24 09:51:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: plug caps leak

2012-03-23 18:47:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/imagefreeze.c:
	  tests: imagefreeze: remove extraneous _unref

2012-03-23 18:47:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/avimux.c:
	  tests: avimux: adjust to modified sink pad template name

2012-03-23 18:46:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: cleanup element sooner
	  ... to avoid stray refs in sticky caps events.

2012-03-23 18:45:56 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/audiowsincband.c:
	* tests/check/elements/audiowsinclimit.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/qtmux.c:
	  tests: arrange for sending an initial segment event
	  ... which is needed nowadays since various gst_segment_to_...
	  no longer automatically set the format to the specified one
	  (from _UNDEFINED).

2012-03-23 18:44:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: immediately return GST_FLOW_EOS
	  ... rather than _OK since we will not be caring about subsequent buffer
	  anyway.

2012-03-23 18:43:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: fix query and _getcaps handling

2012-03-23 18:42:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	  audiofx: adjust to changed semantics of audiofilter _setup method
	  ... in that it will now call subclass with info on proposed audio format
	  without having set that info already in base class.  As such,
	  subclass can not rely on audio format info being available there.

2011-07-14 16:23:49 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set
	  This allows outputting streams in AVC format even if the SPS/PPS are sent inside
	  the RTP stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654850
	  Ported from master

2012-01-29 18:39:54 +0000  Olivier Crête <olivier.crete@collabora.com>

	* gst/udp/gstmultiudpsink.c:
	  udpsink: Unlock on error

2012-03-22 18:27:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: use sink pad template caps rather than src

2012-03-22 18:23:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  Merge branch 'master' into 0.11

2012-03-22 18:21:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmpte.h:
	* gst/smpte/gstsmptealpha.c:
	* gst/smpte/gstsmptealpha.h:
	  smpte: port to 0.11

2012-03-22 16:10:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: intersect downstream allowed peer caps with sink pad template

2012-03-22 15:55:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  back to development

=== release 0.11.2 ===

2012-03-22 15:51:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	* win32/common/config.h:
	* win32/common/gstudp-marshal.c:
	  Release 0.11.2

2012-03-22 11:55:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2012-03-22 11:53:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  unport gdkpixbuf
	  not merged: https://bugzilla.gnome.org/show_bug.cgi?id=654850
	  Conflicts:
	  docs/plugins/Makefile.am
	  docs/plugins/gst-plugins-good-plugins-docs.sgml
	  docs/plugins/gst-plugins-good-plugins-sections.txt
	  docs/plugins/gst-plugins-good-plugins.hierarchy
	  docs/plugins/inspect/plugin-avi.xml
	  docs/plugins/inspect/plugin-png.xml
	  ext/flac/gstflacdec.c
	  ext/flac/gstflacdec.h
	  ext/libpng/gstpngdec.c
	  ext/libpng/gstpngenc.c
	  ext/speex/gstspeexdec.c
	  gst/audioparsers/gstflacparse.c
	  gst/flv/gstflvmux.c
	  gst/rtp/gstrtpdvdepay.c
	  gst/rtp/gstrtph264depay.c

2012-03-22 11:45:11 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/smpte/gstsmpte.c:
	  smpte: only start collectpads2 at state change rather than init

2012-03-21 13:22:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/audioamplify.c:
	* tests/check/elements/audiodynamic.c:
	* tests/check/elements/audioecho.c:
	* tests/check/elements/audiopanorama.c:
	* tests/check/elements/rtp-payloading.c:
	  tests: update for memory api changes

2012-03-20 10:24:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  update for memory api changes

2012-03-19 12:01:40 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: perform additional frame crc check if applicable
	  ... such as a frame header parsing throwing some suspicious warnings.
	  So we can be a bit more convinced we determine the right frame end.

2012-03-19 11:58:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: avoid indefinite extended search for frame end if possible
	  ... which is particularly useful if locked on to the wrong frame start
	  and/or corrupt frame being crc checked.

2012-03-16 18:23:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: improve error handling and resilience
	  ... by noting that one occurred in the first place, and then appropriately
	  ignoring some transient ones.

2012-03-19 10:33:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: negotiate an allocator on the srcpads
	  We do an ALLOCATION query to find out an allocator and parameters on the
	  srcpads. This way decoders (and sinks) can specify the memory and parameters
	  they want us to write into.

2012-03-17 20:53:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	  docs: update docs for new properties and add gdkpixbufoverlay element
	  Somewhat at least. No idea why it doesn't pick up the description
	  or example pipeline.

2012-03-18 00:11:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: make most properties controllable and flag them as mutable-playing

2012-03-17 23:41:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	  gdkpixbufoverlay: add properties for positioning and sizing

2012-03-17 20:18:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
	  gdkpixbuf: add gdkpixbufoverlay element
	  Still lacks features such as positioning or resizing, or
	  animations, but it's usable already, and supports lots of
	  formats.

2012-03-16 22:52:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	  don't poke into basetransform internals
	  But use the methods

2012-03-16 21:47:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-parse.c:
	* gst/wavparse/gstwavparse.c:
	  don't pass random pointers to pull_range

2012-03-15 22:15:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/monoscope/gstmonoscope.c:
	  updarte for bufferpool changes

2012-03-15 22:11:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/dv/gstdvdec.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  update for bufferpool changes

2012-03-15 20:37:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/dv/gstdvdec.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  update for allocation query changes

2011-07-14 16:23:49 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set
	  This allows outputting streams in AVC format even if the SPS/PPS are sent inside
	  the RTP stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654850

2012-03-15 14:06:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  update for bufferpool api change

2012-03-15 13:38:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  update for memory api changes

2012-03-15 13:37:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	  update for memory api changes

2012-03-15 13:36:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/flac/gstflacdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* gst/interleave/deinterleave.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/matroska/matroska-demux.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multipart/multipartmux.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/videomixer/videomixer2.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* tests/check/elements/audiochebband.c:
	* tests/check/elements/audiocheblimit.c:
	  update for memory api changes

2012-03-14 21:36:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  update for memory api changes

2012-03-14 19:55:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/dv/gstdvdec.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  take padding into account

2012-03-14 17:07:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: port to 0.11

2012-03-14 15:45:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reply FALSe on serialized queries

2012-03-13 23:08:38 +0100  Andrej Gelenberg <andrej.gelenberg@udo.edu>

	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	  pngenc: add support for 8- and 16-bit gray images
	  Add support for direct encoding of 8- and 16-bit big endian gray images.
	  https://bugzilla.gnome.org/show_bug.cgi?id=672025

2012-03-14 11:21:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4vpay.c:
	  mp4vpay: we can also handle x-divx

2012-03-14 10:39:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackenc.c:
	  wavpackenc: do not set output caps directly
	  ... but use base class function instead.

2012-03-13 21:31:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4vdepay.c:
	  mp4vdepay: fix buffer handling
	  Don't always output the payload subbuffer, use a separate variable to
	  make things clearer and without the error.

2012-03-13 20:49:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udpsink: make buffer-size work again

2012-03-13 20:36:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix SO_RCVBUF handling

2012-03-13 19:26:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: don't leak the address

2012-03-13 19:26:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: unmap on empty packet

2012-03-13 18:07:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: do DTS and PTS correctly

2012-03-13 17:54:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: set DTS and PTS on output buffers
	  Set PTS and DTS on output buffers instead of just the PTS. In streaming cases
	  you want to synchronized encoded data based on the DTS because that is
	  monotonically increasing.

2012-03-13 17:54:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux: debug additional sdtp flag

2012-03-13 17:27:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	  rtp: fix unmap calls

2012-03-13 13:25:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.h:
	  pulse: fix formats, we can not handle S8 but only U8

2012-03-13 12:40:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: fix streamheaders
	  Fix the caps of flacenc, the reference encoder only support 24 bits in
	  32 bits.
	  Set streamheader on output caps.

2012-03-12 17:17:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/monoscope/gstmonoscope.c:
	  update for caps api changes

2012-03-12 16:43:27 +0200  Sreerenj Balachandran <sreerenj.balachandran@intel.com>

	* configure.ac:
	  configure.ac : bump GLib requirement to 2.31.14
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=671911

2012-03-12 15:27:27 +0100  Ross Burton <ross at burtonini.com>

	* ext/flac/gstflacenc.c:
	  flacenc: generate seektables every 10 sec by default
	  Since this is what the command line tool does as well, it seems like
	  a better default.

2012-03-10 13:44:08 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: only unlock pad when it was locked
	  This fixes the mutex being unlocked too much and ending up allowing
	  other threads when they should not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=671776

2012-03-07 13:39:50 +0100  Andrej Gelenberg <andrej.gelenberg@udo.edu>

	* ext/libpng/gstpngdec.c:
	  pngdec: add support for video/x-raw-gray formats
	  pngdec can now decode gray 8- and 16-bit images without alpha channel
	  direct to video/x-raw-gray format. 16-bit gray images have big-endian
	  format, because it's native PNG endianness. Gray images with alpha
	  channel still converted to RGBA.
	  Signed-off-by: Andrej Gelenberg <andrej.gelenberg@udo.edu>

2012-03-08 17:07:51 +0100  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  gstrtspsrc: disable RTSP keep-alive on request

2012-03-12 14:48:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/smpte/gstsmpte.c:
	  smpte: fix stride handling

2012-03-12 12:23:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* tests/check/elements/videocrop.c:
	* tests/check/elements/videofilter.c:
	  fix for caps _normalize changes

2012-03-12 11:47:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	* gst/matroska/matroska-demux.c:
	  fix for caps api change

2012-03-12 10:43:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	* gst/matroska/matroska-demux.c:
	* sys/oss4/oss4-audio.c:
	  fix for _do_simplify changes

2012-03-12 08:48:32 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/flv/gstflvmux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/matroska/matroska-mux.c:
	  gst: Fix some query leaks

2012-03-11 19:06:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  fix for caps api changes

2012-03-11 19:06:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/pulse/pulsesrc.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/videomixer/videomixer2.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/ximage/gstximagesrc.c:
	  fix for caps api changes

2012-03-10 10:51:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* gst/alpha/gstalphacolor.c:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/auparse/gstauparse.c:
	* gst/goom2k1/gstgoom.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	  fix template caps refcount

2012-03-09 15:53:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: fix use of AC_LANG_PROGRAM
	  No need to include the int main () { } bits, the body is enough.

2012-03-09 15:25:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: fix autogen.sh warnings
	  configure.ac:410: warning: AC_LANG_CONFTEST: no AC_LANG_SOURCE call detected in body

2012-03-08 13:06:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/aalib/gstaasink.h:
	  aasink: propose videometa uptream
	  subclass from videosink.
	  Propose videometa upstream because we can handle it with the video api.

2012-03-08 01:53:50 -0500  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: do not unref sample caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=671534

2012-03-08 11:36:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/autodetect.c:
	* tests/check/elements/videocrop.c:
	  tests: improve more tests

2012-03-08 11:20:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/capssetter.c:
	* tests/check/elements/gdkpixbufsink.c:
	  tests: fix some more tests

2012-03-07 15:22:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: improve cleanup
	  Reuse cleanup methods to make sure we remove all pads correctly

2012-03-07 15:00:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: set caps without the lock
	  Release the lock before setting the caps on the srcpad, which triggers an event,
	  which could eventually call back into us and cause a deadlock.

2012-03-07 14:55:08 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpptdemux.c:
	  ptdemux: set caps after activating the pad
	  Set the caps after we activated the pad or else it will just fail.

2012-03-07 14:54:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/law/alaw.c:
	* gst/law/mulaw.c:
	  law: add layout to audio caps

2012-03-07 14:51:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-decode.h:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-decode.h:
	  law: use GstAudioInfo
	  Use GstAudioInfo to generate output caps.

2012-03-07 04:20:00 -0500  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtdemux: covert art tag type is GstSample not GstBuffer now
	  https://bugzilla.gnome.org/show_bug.cgi?id=671534

2012-03-07 10:28:58 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/POTFILES.in:
	  po: fix POTFILES.in for new wavpackparse location in source tree

2012-03-06 21:44:36 -0800  David Schleef <ds@schleef.org>

	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	  udp: Change the default port to 5004
	  udpsrc/udpsink are almost always used with RTP, so let's use an
	  RTP port as the default port.  It's unclear why 4951 was used, it
	  goes back to early commits in CVS.

2012-03-06 21:36:02 -0800  David Schleef <ds@schleef.org>

	  Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11

2012-03-06 15:58:20 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: use base class tag handling helper
	  ... so as to ensure these to be handled and sent at proper time.

2012-03-06 14:25:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/wavpack/gstwavpackstreamreader.c:
	  wavpack: Fix possible underflow of unsigned integer variable

2012-03-06 14:22:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Fix 'comparison of unsigned expression >= 0 is always true'
	  This variable can never be below zero anyway.

2012-03-06 14:18:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use correct enum for return values

2012-03-06 14:16:21 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpdvdepay.c:
	  dvdepay: Fix 'comparison of unsigned expression >= 0 is always true' compiler warning
	  This was an actual bug as it could've caused reading from
	  invalid memory areas when the input is broken.

2012-03-06 13:21:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/greedyh.asm:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopTop.inc:
	  deinterlace: Fix 'variable 'oldbx' is uninitialized when used here' compiler warnings

2012-03-06 13:19:24 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix 'implicit conversion from enumeration type 'GstDeinterlaceFields' to different enumeration type 'GstDeinterlaceMode'' compiler warning

2012-03-05 15:29:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.h:
	  gdk: cleanups and fix rowstride
	  Fix the output rowstride, we need to take the stride of the output video frame.
	  Since we are also dealing with planes, take the plane data and stride.
	  Don't store the same info twice in different variables.

2012-03-05 13:31:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	  gdkpixbuf: fix event handling

2012-03-05 12:20:07 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/wavpackdec.c:
	* tests/check/elements/wavpackenc.c:
	* tests/check/elements/wavpackparse.c:
	* tests/check/pipelines/wavpack.c:
	  tests: port wavpack tests to 0.11

2012-03-05 13:36:39 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackdec.h:
	  wavpackdec: port to 0.11

2012-03-05 12:17:39 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackcommon.c:
	* ext/wavpack/gstwavpackcommon.h:
	* ext/wavpack/gstwavpackenc.c:
	  wavpackenc: port to 0.11

2012-03-05 13:34:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* ext/wavpack/Makefile.am:
	* ext/wavpack/gstwavpack.c:
	* ext/wavpack/gstwavpackparse.c:
	* ext/wavpack/gstwavpackparse.h:
	  wavpack: remove legacy wavpackparse

2012-03-05 12:15:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/audioparsers/gstwavpackparse.h:
	* gst/audioparsers/plugin.c:
	  audioparsers: port wavpackparse to 0.11

2012-03-05 13:29:59 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/wavpack/gstwavpackparse.c
	  sys/v4l2/gstv4l2bufferpool.c
	  sys/v4l2/gstv4l2bufferpool.h
	  sys/v4l2/gstv4l2videooverlay.c

2012-03-05 12:43:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  x-raw-bayer -> x-bayer

2012-03-05 11:17:30 +0100  Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

	* sys/v4l2/gstv4l2xoverlay.c:
	  v4l2sink: don't use deprecated XKeycodeToKeysym
	  https://bugzilla.gnome.org/show_bug.cgi?id=671299
	  Signed-off-by: Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

2012-03-05 12:03:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/ximage/Makefile.am:
	* sys/ximage/gstximagesrc.c:
	  ximage: use new style caps

2012-03-05 10:49:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: allow some timestamp tolerance to arrange for perfect timestamping
	  ... which also happens to make some more unit tests pass.

2012-03-05 10:47:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: fix copying output data

2012-03-05 10:46:51 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackenc.c:
	  wavpackenc: restore legacy buffer offset decorating somewhat
	  ... at least sufficiently to aid in recognizing rewritten header buffer
	  making unit test pass.

2012-03-05 10:51:33 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/audioparsers/gstwavpackparse.c:
	  wavpackparse: initialize header to silence older gcc versions

2012-03-05 10:45:46 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/wavpack/gstwavpackparse.c:
	  wavpackparse: remove empty lines in varable declarations caused by old indent

2012-03-05 10:44:54 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/jack/gstjack.h:
	  jack: fix obvious wrong definition for the master flag

2012-03-04 19:55:26 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: change the transport-mode enum into flags
	  One can use (or not use) master and slave mode independently.

2012-03-02 11:49:02 -0500  Antoine Tremblay <hexa00@gmail.com>

	* gst/avi/gstavimux.c:
	  avimux: support up to 6 channels of AC-3
	  https://bugzilla.gnome.org/show_bug.cgi?id=671220

2012-03-03 13:04:48 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: clear DISCONT flag when recycling buffers into the buffer pool
	  The base class may have set the DISCONT flag on the first buffer pushed
	  out. We need to clear that when recycling buffers back into the buffer
	  pool, otherwise we constantly push out buffers with the discont flag
	  set, which might upset downstream elements, esp. for compressed
	  formats like mpeg-ts.

2012-03-01 14:15:29 +0100  Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2src: fix v4l2_munmap() for compressed formats
	  Make sure we always call munmap() with the same size we called mmap()
	  with before.
	  Current v4l2src uses the same structure for VIDIOC_QUERYBUF, VIDIOC_QBUF
	  and v4l2_munmap calls. The problem is that the video buffer size (length)
	  may vary for compressed or emulated bufs. VIDIOC_QBUF will change it if
	  we pass the pointer of a v4l2_buffer. This is why we should avoid using
	  same variable for mmap and video buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=671126

2012-03-02 11:17:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/flv/gstindex.c:
	  gst: Update for the gstmarshal.[ch] removal

2012-03-02 10:13:08 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsemixerctrl.h:
	* gst/videofilter/gstvideobalance.c:
	* sys/v4l2/gstv4l2colorbalance.h:
	  mixer/colorbalance: Update for API changes

2012-03-01 17:15:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	  aasink: fix stride

2012-03-01 11:36:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/plugin.c:
	  audioparsers: disable non-ported wavpackparse

2012-03-01 11:29:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/wavpack/gstwavpackenc.c
	  tests/check/elements/audioiirfilter.c
	  tests/examples/v4l2/probe.c

2012-02-29 22:31:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	  gdkpixbufsink: remove deprecated property

2012-02-29 22:30:56 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	  gdkpixbufscale: remove deprecated property

2012-02-29 22:28:01 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.h:
	  gdkpixbufsink: port to 0.11

2012-02-29 22:25:23 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/gdk_pixbuf/pixbufscale.h:
	  gdkpixbufscale: port to 0.11

2012-02-29 22:24:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.h:
	  gdkpixbufdec: port to 0.11

2012-02-29 17:26:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/ximage/ximageutil.c:
	* sys/ximage/ximageutil.h:
	  update for metadata API changes

2012-02-28 13:51:10 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstwavpackparse.c:
	* gst/audioparsers/gstwavpackparse.h:
	* gst/audioparsers/plugin.c:
	  audioparsers: add baseparse based wavpackparse

2012-02-28 11:38:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/ximage/ximageutil.c:
	  update for metadata tags

2012-02-27 23:46:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackdec.h:
	* tests/check/elements/wavpackdec.c:
	  wavpackdec: adjust to audio format limitations
	  ... which does not allow expressing arbitrary depth in a GstAudioFormat.
	  Also adjust unit test to modified behaviour.

2012-02-27 23:46:08 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	  wavpackdec: determine depth from bytes per sample
	  ... rather than from bits per sample, since spec states values are already
	  left justified w.r.t. bits per sample but not w.r.t. bytes per sample
	  (and so the latter determines the normalization, or indicated depth).

2012-02-27 23:46:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackdec.h:
	  wavpackdec: port to audiodecoder

2012-02-27 23:45:54 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackenc.h:
	* tests/check/elements/wavpackenc.c:
	  wavpackenc: port to audioencoder
	  Also adjust unit test to slightly modified behaviour.

2012-02-27 14:47:25 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/annodex/gstannodex.c:
	* ext/annodex/gstcmmlparser.c:
	* ext/annodex/gstcmmltag.c:
	* ext/pulse/pulseprobe.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/interleave/interleave.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* sys/oss4/oss4-audio.c:
	* sys/oss4/oss4-property-probe.c:
	* sys/v4l2/gstv4l2object.c:
	* tests/check/elements/audiofirfilter.c:
	* tests/check/elements/audioiirfilter.c:
	* tests/check/elements/cmmldec.c:
	* tests/check/elements/interleave.c:
	* tests/check/pipelines/wavenc.c:
	* tests/examples/audiofx/firfilter-example.c:
	* tests/examples/audiofx/iirfilter-example.c:
	* tests/examples/pulse/pulse.c:
	* tests/examples/rtp/server-alsasrc-PCMA.c:
	* tests/examples/v4l2/probe.c:
	* tests/icles/test-oss4.c:
	  Suppress deprecation warnings in selected files, for g_value_array_* mostly

2012-02-27 13:09:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: chain up to parent event handler

2012-02-27 13:05:33 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: fix event handling
	  Fix dodgy segment event handling
	  Chain up to parent event handler

2012-02-27 09:14:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: use public api
	  instead of poking into the private structures of the base class

2012-02-27 06:35:01 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* ext/lame/Makefile.am:
	  amrwbdec, lame, mad: link to libgstbase

2012-02-27 01:09:11 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/matroska/matroska-mux.c:
	  flvmux, matroskamux, qtmux: if in doubt about downstream seekability default to streaming=true
	  If downstream didn't answer our SEEKING query and told us
	  it's seekable, default to streaming=true. We couldn't do
	  this in 0.10 for backwards compatibility reasons, but we
	  can in 0.11. Play it safe.

2012-02-27 01:00:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11
	  Conflicts:
	  gst/audioparsers/gstmpegaudioparse.c

2012-02-27 00:56:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge commit 'f9207722ca8fd8dcc1e7215d8af85efe4debfdf4' into 0.11

2012-02-27 00:55:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: fix up after merge

2012-02-27 00:48:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge commit '38516ad367128d83f9e156529018adb4433cd328' into 0.11
	  Conflicts:
	  ext/pulse/pulseaudiosink.c
	  gst/audioparsers/gstmpegaudioparse.c

2012-02-26 20:39:52 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/goom2k1/gstgoom.c:
	  goom2k1: fix compiler warning

2012-02-26 20:30:24 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: fix compiler warning

2012-02-25 15:55:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: create streamable output if downstream is not seekable
	  Ignore the "streamable" property setting and create streamable
	  output if downstream is known not to be seekable (as queried
	  via a SEEKABLE query).
	  Fixes pipelines like qtmux ! appsink possibly creating seemingly
	  corrupted output if streamable has not been set to true.

2012-02-25 15:48:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: create streamable output if downstream is not seekable
	  Ignore the "streamable" property setting and create streamable
	  output if downstream is known not to be seekable (as queried
	  via a SEEKABLE query).
	  Fixes pipelines like flvmux ! appsink possibly creating seemingly
	  corrupted output if streamable has not been set to true.

2012-02-25 15:40:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: create streamable output if downstream is not seekable
	  Ignore the "streamable" property setting and create streamable
	  output if downstream is known not to be seekable (as queried
	  via a SEEKABLE query).
	  Fixes pipelines like webmmux ! appsink creating seemingly
	  corrupted output if streamable has not been set to true.

2012-02-24 11:03:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstvideoflip.c:
	  update for basetransform change

2012-02-24 10:26:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/ximage/ximageutil.c:
	  update for metadata change

2012-02-23 08:42:25 -0800  David Schleef <ds@schleef.org>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/inspect/plugin-efence.xml:
	* gst/debugutils/Makefile.am:
	* gst/debugutils/efence.c:
	* gst/debugutils/efence.h:
	* gst/debugutils/efence.vcproj:
	  efence: remove plugin
	  Valgrind is much more useful these days.

2012-02-23 12:05:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* NEWS:
	* RELEASE:
	  Update NEWS and RELEASE as well

2012-02-23 11:07:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Bump version after release

2012-02-23 12:03:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/audioecho.c:
	* gst/audiofx/audioecho.h:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.h:
	  audiofx: remove transform lock usage

2012-02-23 11:16:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	* gst/videofilter/gstvideobalance.c:
	  update for basetransform lock removal

2012-02-22 23:36:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/Makefile.am:
	  debugutils: disable efence plugin properly
	  We don't want it built if mmap isn't available either..

2012-02-22 17:39:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: fix get_caps function some more so that all structures have channel info
	  Set channels and channel-layout on the right structure; that is, the
	  structure we are going to append to the caps we are building, and not
	  the structure we are using as a template for all the structures. Fixes
	  first structure of the returned caps not having any channel info set
	  on it.

2012-02-22 17:09:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: microoptimisation: avoid unnecessary list and string copies

2012-02-22 17:03:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: audio caps have a *list* of formats, not an array of formats
	  A list of things in caps is something where one is picked in the
	  course of negotiation. An array is always something that only makes
	  sense as a whole in that order.

2012-02-22 18:02:27 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: remove post-port bogus _unref

2012-02-22 17:00:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: remove bogus pad locking that causes deadlocks
	  It's not clear why the pad object lock is taken here. But
	  gst_pad_{has,get}_current_caps() will try to take the lock
	  as well and deadlock, since it's not recursive.

2012-02-22 16:59:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: set right number of channels on caps in get_caps function

2012-02-21 17:16:32 -0800  David Schleef <ds@schleef.org>

	* autogen.sh:
	  autogen: avoid touching .po files during 'make'
	  A simple workaround to deal with GNU gettext automake integration
	  failing to deal with git.  Fixes: #669207

2012-02-22 02:06:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/atoms.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsrc.c:
	* gst/y4m/gsty4mencode.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/ximage/ximageutil.c:
	* tests/check/elements/deinterleave.c:
	* tests/check/elements/interleave.c:
	  update for new memory api

2012-02-21 17:57:44 +0100  Vincent Untz <vuntz@gnome.org>

	* ext/pulse/pulseaudiosink.c:
	  pulse: Fix a build warning when compiling with asserts disabled
	  Return a value even if the code will never be reached, to make compilers
	  happy.
	  https://bugzilla.gnome.org/show_bug.cgi?id=670561

2012-02-21 18:42:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstmpegaudioparse.h:
	  mpegaudioparse: support parsing freeform bitrate stream

2012-02-21 18:39:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* gst/monoscope/gstmonoscope.c:
	* gst/monoscope/gstmonoscope.h:
	  monoscope: port to 0.11

2012-02-21 10:53:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2012-02-20 12:22:12 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Force baseline is profile-level-id is unspecified

2012-02-21 10:40:00 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: Fix merge error

2012-02-20 12:22:12 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Force baseline is profile-level-id is unspecified

2012-02-20 16:35:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  fix compiler warnings

2012-01-26 03:29:28 -0500  Matej Knopp <matej.knopp@gmail.com>

	* gst/udp/gstudpsrc.c:
	  fix compiler warnings

2012-01-26 06:58:46 -0500  Matej Knopp <matej.knopp@gmail.com>

	* gst/dtmf/gstdtmfsrc.c:
	  Fix compiler warnings

2012-02-18 11:38:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/level.c:
	  tests: fix up level test for GstValueList -> GValueArray change
	  https://bugzilla.gnome.org/show_bug.cgi?id=670303

2012-02-16 18:01:29 +0200  Peteris Krisjanis <pecisk@gmail.com>

	* gst/level/gstlevel.c:
	  level: use GValueArray instead of GstValueList in messages
	  Updated GstLevel element to use GValueArray instead of
	  GstValueList for rms/peak/decay keys attached to element
	  message.
	  https://bugzilla.gnome.org/show_bug.cgi?id=670303

2012-02-18 00:00:54 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* win32/common/config.h:
	  win32: back to development

2012-02-17 23:54:29 +0100  Dominique Leuenberger <dominique-gnomezilla at leuenberger.net>

	* docs/plugins/Makefile.am:
	  No longer reference deprecated header files while building docs.

2012-02-17 23:49:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/equalizer/gstiirequalizer.c

2012-02-17 17:21:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: fix switching from passthrough to non-passthrough when parameters change
	  commit b5bf0294 moved the if(need_new_coefficients) set_passthrough(equ)
	  after the if(is_passthrough) return FLOW_OK shortcut, so the passthrough
	  mode would never get updated even if the coefficients change.
	  Fixes equalizer-test doing .. nothing.

2012-02-17 17:57:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	  goom*: fix leaked caps event

2012-02-17 13:26:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: parse either Xing or VBRI data
	  ... and avoid confusing debug message claiming neither present.

2012-02-17 14:38:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matrosk: fix segment update

2012-02-17 11:05:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  back to development

=== release 0.11.1 ===

2012-02-17 11:04:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	* win32/common/config.h:
	* win32/common/gstrtpbin-marshal.c:
	* win32/common/gstrtpbin-marshal.h:
	  RELEASE 0.11.1

2012-02-16 23:33:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: fix buffer leak

2012-02-16 23:40:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/goom2k1/gstgoom.c:
	  goom2k1: use some more boilerplate

2012-02-16 23:33:01 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	  goom2k1: port to 0.11

2012-02-16 15:31:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/shout2/gstshout2.c:
	  shout2: use some more boilerplate

2012-02-16 15:29:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* ext/shout2/gstshout2.c:
	  shout2: port to 0.11

2012-02-14 11:56:00 +0100  Philippe Normand <philn@igalia.com>

	* gst/interleave/Makefile.am:
	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	* gst/interleave/plugin.c:
	* gst/interleave/plugin.h:
	* tests/check/elements/interleave.c:
	  interleave: port to 0.11
	  Port of the interleave element and its unittests.
	  https://bugzilla.gnome.org/show_bug.cgi?id=669643

2012-02-16 14:23:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2012-02-16 17:14:20 +0800  Gary Ching-Pang Lin <chingpang@gmail.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: failure to query some optional controls is not a fatal error
	  Don't post a (fatal) error message on the bus just because we
	  failed to query some control. Fixes issue with built-in
	  Suyin Corp webcam for HP notebook (usbid 064e:e28a) on
	  OpenSuse 12.1, where querying red/blue balance fails.
	  https://bugzilla.gnome.org/show_bug.cgi?id=670197

2012-02-16 12:59:10 +0000  Tuukka Pasanen <tuukka.pasanen@ilmi.fi>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: fix for webcamstudio vloopback
	  Because vlooback emits 25 - ENOTTY and no EINVAL v4l2src thought it
	  can't handle this and does not work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=669455

2012-02-16 11:21:28 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: declare variables at the beginning of the block
	  It's how we roll. Fixes 'ISO C90 forbids mixed declarations and code'
	  compiler warning.

2012-02-15 23:55:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/spectrum/Makefile.am:
	  examples: fix spectrum example build issues
	  Find fft headers in uninstalled setup, fix LIBS order.

2012-02-15 12:41:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: remove some unused declarations

2012-02-15 11:25:45 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/spectrum/Makefile.am:
	* tests/examples/spectrum/demo-audiotest.c:
	  spectrum-demo: show the effect of fast-mode

2012-02-14 12:26:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videocrop/gstaspectratiocrop.c:
	  aspectratiocrop: fix caps refcount

2012-02-14 11:22:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/pipelines/effectv.c:
	  tests: fix test, use videoconvert

2012-02-14 10:51:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  tests/check/elements/flacparse.c

2012-02-09 13:41:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparsers: adjust to modified baseparse API

2012-02-13 17:13:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	* gst/udp/gstmultiudpsink.c:
	  update for memory api change

2012-02-13 12:06:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/flacparse.c:
	  tests: flacparse: check and compare intended data

2012-02-12 17:03:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11
	  Conflicts:
	  ext/taglib/gstapev2mux.cc
	  ext/taglib/gstid3v2mux.cc
	  ext/taglib/gsttaglibmux.c
	  ext/taglib/gsttaglibmux.h

2012-02-12 16:22:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/taglib/Makefile.am:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstapev2mux.h:
	* ext/taglib/gstid3v2mux.cc:
	* ext/taglib/gstid3v2mux.h:
	* ext/taglib/gsttaglibmux.c:
	* ext/taglib/gsttaglibmux.h:
	* ext/taglib/gsttaglibplugin.c:
	  taglib: port to GstTagMux base class

2012-02-12 12:24:50 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/taglib/gsttaglibmux.c:
	  taglib: finish off a few missed variable changes
	  Local variables are now unused, and the values from the segment copy
	  are used instead, so remove the now useless local variables and write
	  to the segment where appropriate.

2012-02-10 16:23:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/flac/gstflacenc.c
	  ext/jack/gstjackaudioclient.c
	  ext/jack/gstjackaudiosink.c
	  ext/jack/gstjackaudiosrc.c
	  ext/pulse/plugin.c
	  ext/shout2/gstshout2.c
	  gst/matroska/matroska-mux.c
	  gst/rtp/gstrtph264pay.c

2012-02-08 23:03:28 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: add stream-format and alignment to h264 sink caps
	  We're happy to accept both byte-stream and avc, advertise
	  that on the sink caps and fix up _get_caps() function to
	  not just return "video/x-h264".
	  https://bugzilla.gnome.org/show_bug.cgi?id=606662

2012-02-08 20:58:04 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: add stream-format and alignment fields to src template caps
	  Because we can. And so we get a warning if we try to output avc with
	  nal alignment or somesuch.
	  https://bugzilla.gnome.org/show_bug.cgi?id=606662

2012-02-10 13:44:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/rtp-payloading.c:
	  tests: clean up rtp-payloading test a little
	  Feed data into the pipeline using appsrc instead of fdsrc and
	  a pipe. Store unsigned byte values in guint8 instead of char.
	  Getting rid of the capsfilter also helps to avoid 'format is
	  not fully specified' warnings when pushing "video/x-h264" data
	  into rtph264pay with fully specified h264 caps in the sink template.

2012-02-10 10:07:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flv: use default pad query
	  We need to chain up unknown queries to the default query handler instead of
	  blindly forwarding them. In this case it caused the caps query to be forwarded
	  to the upstream typefind and return the wrong type for the audio/video pad.

2012-02-09 22:12:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/mpegaudioparse.c:
	  tests: mpegaudioparse: remove stray declaration

2012-02-09 22:07:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: correctly set ADIF src caps

2012-02-09 22:10:07 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: prevent a few direct exits without cleanup

2012-02-09 22:07:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: shift in proper direction for audio sample conversion

2012-02-09 18:09:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/deinterleave.c:
	  tests: fix compilation

2012-02-09 10:11:48 +0100  Marc Leeman <marc.leeman@gmail.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: typo fix (bytes send -> bytes sent)

2012-02-08 16:34:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/effectv/gstquark.c:
	* gst/flv/gstflvdemux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/isomp4/qtdemux.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/udp/gstudpsrc.c:
	* gst/wavenc/gstwavenc.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/ximage/gstximagesrc.c:
	  GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING

2012-02-08 16:37:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING

2012-02-07 14:10:44 -0800  Ralph Giles <giles@mozilla.com>

	* ext/shout2/gstshout2.c:
	  shout2send: send video/webm through libshout.
	  This requires SHOUT_FORMAT_WEBM, added in libshout 2.3.0,
	  so video/webm support is contingent on that symbol being
	  defined.
	  Also an indentation change required by the pre-commit hook.
	  https://bugzilla.gnome.org/show_bug.cgi?id=669590

2012-01-30 16:40:19 +0100  Philippe Normand <philn@igalia.com>

	* configure.ac:
	* gst/interleave/Makefile.am:
	* gst/interleave/deinterleave.c:
	* gst/interleave/deinterleave.h:
	* gst/interleave/plugin.c:
	* gst/interleave/plugin.h:
	* tests/check/elements/deinterleave.c:
	  deinterleave: port to 0.11
	  Port of the deinterleave element and its unittests. The interleave
	  element will be ported as part of another patch, hence disabling it
	  for now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=668847

2012-02-07 23:41:13 +0200  Raimo Järvi <raimo.jarvi@gmail.com>

	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: Fix compiling
	  https://bugzilla.gnome.org/show_bug.cgi?id=669607

2012-02-08 00:08:49 +0200  Raimo Järvi <raimo.jarvi@gmail.com>

	* sys/waveform/gstwaveformsink.c:
	  waveformsink: Port to 0.11
	  https://bugzilla.gnome.org/show_bug.cgi?id=669612

2012-02-07 21:57:47 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: rework transport support
	  Move common code to jackclient. There we can also handle the request state
	  message in a better way, as the element callbacks are only run if the element is
	  active.

2012-02-07 10:47:19 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/apev2mux.c:
	* tests/check/elements/id3v2mux.c:
	  tests: improve tagmux tests

2012-02-07 10:29:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/taglib/gsttaglibmux.c:
	  taglib: fix object registration
	  We can't use G_DEFINE_TYPE because the class is not set in the class_init and we
	  need it to get the srcpad template.
	  Fix a caps leak

2012-02-07 10:16:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/jpegenc.c:
	  tests: fix jpeg test

2012-02-07 10:15:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  soup: fix caps

2012-02-07 09:54:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstdice.c:
	* gst/effectv/gstshagadelic.c:
	  effecttv: fix initialisation

2012-02-07 09:42:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/y4m/gsty4mencode.c:
	  y4m: fix negotiation

2012-02-07 09:41:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/videofilter.c:
	* tests/check/elements/y4menc.c:
	  tests: fix more tests

2012-02-06 22:13:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* ext/dv/Makefile.am:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdec.h:
	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dv: port to 0.11

2012-02-06 18:35:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rglimiter.c:
	* tests/check/elements/rgvolume.c:
	* tests/check/elements/spectrum.c:
	* tests/check/elements/videocrop.c:
	  test: fix more tests

2012-02-06 15:52:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/id3demux.c:
	* tests/check/elements/level.c:
	* tests/check/elements/multifile.c:
	  tests: fix more tests

2012-02-06 15:52:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/flv/Makefile.am:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	  flv: fix caps

2012-02-06 15:20:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	* tests/check/elements/equalizer.c:
	  iirequalizer: fix equalizer and unit test

2012-02-06 13:44:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/audiopanorama.c:
	* tests/check/elements/audiowsincband.c:
	* tests/check/elements/audiowsinclimit.c:
	  tests: fix some more tests

2012-02-06 13:43:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: take the pad from collectpads2 correctly

2012-02-06 13:29:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/audioiirfilter.c:
	* tests/check/elements/audioinvert.c:
	  tests: fix more unit tests

2012-02-06 13:28:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/audiodynamic.c:
	  audiodynamic: fix negotiation

2012-01-28 11:13:16 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: avoid posting invalid duration for each frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=666583

2012-02-06 10:07:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/audioamplify.c:
	* tests/check/elements/audiochebband.c:
	* tests/check/elements/audiocheblimit.c:
	* tests/check/elements/audiodynamic.c:
	* tests/check/elements/audioecho.c:
	  tests: fix more tests

2012-02-06 09:49:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/aspectratiocrop.c:
	* tests/check/elements/rganalysis.c:
	  tests: improve some tests

2012-02-06 09:23:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: fix jitterbuffer test

2012-02-06 09:23:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix caps after pt change

2012-02-06 09:18:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix caps leak

2012-02-03 22:05:59 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/plugin.c:
	  pulseaudiosink: Lower rank to prevent autoplugging
	  pulseaudiosink breaks visualisations in its current form, so let's
	  prevent it from being autoplugged for the time being.
	  The best we can hope to do in the 0.10 series is query the list of
	  available sinks and their formats, and expose these as the bin's sinkpad
	  caps. While this is not a comprehensive solution, it will make sure that
	  we're only trying to support compressed formats if we're certain that
	  one exists.
	  The long-term fix for this will be in the form of proper upstream
	  renegotiation support in the 0.11/1.0 series.
	  https://bugzilla.gnome.org/show_bug.cgi?id=666361

2012-02-03 17:23:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/cmmldec.c:
	  tests: fix more tests

2012-02-03 16:13:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/apev2mux.c:
	* tests/check/elements/audiofirfilter.c:
	* tests/check/elements/audioiirfilter.c:
	* tests/check/elements/cmmldec.c:
	* tests/check/elements/id3v2mux.c:
	* tests/check/elements/interleave.c:
	* tests/check/elements/parser.c:
	* tests/check/pipelines/wavenc.c:
	  tests: fix some more tests

2012-02-03 16:12:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix srcpad caps handling

2012-02-03 16:12:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/annodex/gstcmmlenc.c:
	  cmmlenc: fix caps handling

2012-02-03 14:53:31 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: fix event leak when there is no peer on the src pad

2012-02-02 16:21:29 +0000  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>

	* gst-plugins-good.spec.in:
	  Update spec file

2012-02-02 12:27:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: specify we only accept raw AAC in template caps
	  No header seems to be added, and the codec ID is the same as used
	  for raw by flvdemux, so raw seems the only supported case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665394

2012-02-02 12:25:21 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: specify we only output raw AAC in template caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=665394

2012-02-01 18:01:27 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/taglib/gsttaglibmux.c:
	* ext/taglib/gsttaglibmux.h:
	  taglib: port to 0.11

2012-02-01 16:40:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/annodex/Makefile.am:
	* gst/audiofx/Makefile.am:
	* gst/rtpmanager/Makefile.am:
	* tests/examples/audiofx/Makefile.am:
	* tests/examples/rtp/Makefile.am:
	  build: ignore GValueArray deprecation warnings for the time being
	  until this gets sorted out with the GLib folks and we have a
	  viable alternative.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667228

2012-02-01 16:36:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulseprobe.c:
	* ext/pulse/pulseprobe.h:
	  pulse: disable some unused property probe code
	  which was using GValueArray

2012-02-01 16:20:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/twolame/gsttwolamemp2enc.c:
	  twolame: Use new audio encoder/decoder base class API for srcpad caps

2012-02-01 16:20:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lame: Use new audio encoder/decoder base class API for srcpad caps

2012-02-01 16:11:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	  speex: Use new audio encoder/decoder base class API for srcpad caps

2012-02-01 16:05:51 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	  flac: Use new audio encoder/decoder base class API for srcpad caps

2012-01-31 15:39:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/equalizer.c:
	* tests/check/elements/id3demux.c:
	* tests/check/elements/interleave.c:
	* tests/check/elements/level.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rglimiter.c:
	* tests/check/elements/rgvolume.c:
	* tests/check/elements/rtpbin.c:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/shapewipe.c:
	* tests/check/elements/spectrum.c:
	* tests/check/elements/udpsrc.c:
	* tests/check/elements/y4menc.c:
	* tests/check/pipelines/flacdec.c:
	* tests/check/pipelines/wavenc.c:
	  tests: fix more tests

2012-01-30 14:52:37 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpmp2tpay.c:
	  rtpmp2tpay: do not try to flush a packet when no data is available
	  https://bugzilla.gnome.org/show_bug.cgi?id=668874

2012-01-31 13:41:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/alphacolor.c:
	* tests/check/elements/audiochebband.c:
	* tests/check/elements/audiocheblimit.c:
	* tests/check/elements/audiofirfilter.c:
	* tests/check/elements/audioiirfilter.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/audiowsincband.c:
	* tests/check/elements/audiowsinclimit.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/deinterlace.c:
	* tests/check/elements/deinterleave.c:
	  tests: update some tests for new memory api

2012-01-31 12:22:19 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	  controller: adapt to control-source type changes

2012-01-30 21:39:34 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	  controller: rename control-bindings
	  gst_control_binding_xxx -> gst_xxx_control_binding for consistency.

2012-01-30 17:16:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/annodex/gstcmmlenc.c:
	* ext/flac/gstflacenc.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/speex/gstspeexenc.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* tests/check/elements/cmmldec.c:
	* tests/check/elements/cmmlenc.c:
	  update for HEADER flag

2010-06-11 08:36:33 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Exclude NALu size from payload length on truncated packets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667846

2012-01-28 23:35:50 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: remove obsolete variable, set but not used
	  Reported by andredieb on #gstreamer.

2012-01-28 13:05:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: avoid wrapping opaque to transparent

2012-01-28 12:35:13 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: do not free memory twice
	  A recent change to fix leaking codec ID string accidentally caused
	  one of the very few places that weren't leaking to now free twice.

2012-01-27 16:27:49 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/law/alaw-decode.c:
	  alawdec: Each output sample is 2 bytes

2012-01-27 12:14:49 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Don't leak caps event when not pushing

2012-01-27 12:04:53 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: Forward sticky events

2012-01-27 12:04:05 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: Protect all uses pad list with OBJECT LOCK
	  Actually protect the entire pad list and use it in a thread safe
	  way.

2012-01-27 12:02:25 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Forward sticky events to new pads

2012-01-27 12:01:40 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Add ssrc to forwarded CAPS events
	  Also iterate the list of GstRtpSsrcDemuxPad safely

2012-01-27 11:59:08 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrccdemux: Factor out getting dpad by pad

2012-01-26 18:35:48 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Keep the buffer mapped while it is being modified

2012-01-26 18:35:27 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpsession: Initialise the address pointer to NULL

2012-01-27 12:07:43 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	  dtmf: Use new-style caps

2012-01-27 16:37:19 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  direcsoundsink: Port element to 0.11

2012-01-26 19:48:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: remove pad event function
	  We use the one from collectpads

2012-01-26 18:26:02 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  Revert "qtdemux: fix GstDateTime/GDateTime mixup"
	  This reverts commit 53261261120b4c008de61691c70e94354b28004a.
	  The GstDateTime->GDateTime change in core was apparently accidental,
	  and is now reverted.

2012-01-26 18:25:21 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  Revert "avidemux: fix GstDateTime/GDateTime mixup"
	  This reverts commit acc9f150968b25c5ae5a6940b34ad2d51b174fd2.
	  The GstDateTime->GDateTime change in core was apparently accidental,
	  and is now reverted.

2012-01-26 17:50:30 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix GstDateTime/GDateTime mixup
	  This is a blind fix to match the one I just made to qtdemux,
	  as I do not have an AVI file where the code gets executed.

2012-01-26 17:47:29 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix GstDateTime/GDateTime mixup

2012-01-26 18:51:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: more fixes

2012-01-26 18:43:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer: make videomixer work somewhat

2012-01-26 18:15:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: port to 0.11
	  It builds and gst-inspect-0.11 works.. otherwise untested

2012-01-26 15:48:01 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstdynudpsink.c:
	  dynudpsink: fix get-stats signal registration some more

2012-01-26 15:46:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  Revert "udp: mark action signals as RUN_FIRST"
	  This reverts commit 5c8308599129d9e1606eedb2d3543617658dc306.

2012-01-26 15:39:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udp: mark action signals as RUN_FIRST

2012-01-26 15:37:23 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstdynudpsink.c:
	  udp: mark "get-stats" as action signal

2012-01-26 15:30:41 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstmultiudpsink.c:
	  udp: fix get-stats action signal registration
	  It returns a GstStructure now, not a GValueArray

2012-01-26 16:05:34 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix print format

2012-01-26 11:50:19 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroskamux: Fix size of output buffers

2012-01-26 11:33:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: include right collectpads version

2012-01-26 11:29:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Properly use the alignment parameter of gst_buffer_new_allocate()
	  It's a bitmask for the alignment, not the alignment itself.

2012-01-26 11:18:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroskamux: Properly unmap WRITE maps of the output buffers

2012-01-26 10:44:28 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Update for the new collectpads2 event handling API

2012-01-26 10:40:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Update for the new collectpads2 event handling API

2012-01-26 10:37:52 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Update for the new collectpads2 event handling API

2012-01-26 10:28:51 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Update for new collectpads2 event handling API

2012-01-26 10:27:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: Update for new collectpads2 event handling API

2012-01-25 18:41:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Only forward the event when we didn't handle it ourselves

2012-01-25 18:40:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	* gst/videomixer/videomixer2pad.h:
	  videomixer: some more porting

2012-01-25 18:00:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	  videomixer: port blend function

2012-01-25 16:58:12 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flv: Fix unitialized variables
	  (or rather circumvent issues with naive compilers ...)

2012-01-25 15:21:44 +0000  Jayakrishnan M <jay.krishnanm@gmail.com>

	* ext/cairo/Makefile.am:
	  cairo: fix build, make sure libgstvideo can be found
	  https://bugzilla.gnome.org/show_bug.cgi?id=668648

2012-01-25 14:50:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	  port to new memory API

2012-01-25 13:19:12 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: don't pretend our random hostnames are fully-qualified domain names

2012-01-25 13:47:30 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* common:
	  Automatic update of common submodule
	  From c463bc0 to 7fda524

2012-01-25 12:49:34 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	  Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11

2012-01-25 12:49:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/flac/gstflacdec.c
	  ext/jpeg/gstjpegenc.c
	  ext/pulse/pulsesink.c
	  sys/v4l2/gstv4l2src.c

2012-01-25 12:41:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	  png: port to new memory API

2012-01-25 12:41:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska: port to new memory API

2012-01-24 14:38:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	* ext/pulse/pulsesink.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/spectrum/gstspectrum.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/ximage/gstximagesrc.c:
	* tests/check/elements/parser.c:
	  more memory API porting

2012-01-23 17:25:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/tests.c:
	* gst/equalizer/gstiirequalizer.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/level/gstlevel.c:
	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-read.h:
	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	  update for new memory API

2012-01-25 07:24:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/twolame/gsttwolamemp2enc.c:
	  port to new memory API

2012-01-25 07:24:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  port to new memory API

2012-01-25 11:21:50 +0100  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: port to 0.11

2012-01-25 11:38:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 2a59016 to c463bc0

2012-01-24 18:24:13 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/libpng/gstpngenc.c:
	  pngenc: disably snapshot behaviour by default
	  ... since such behaviour is not consistent, if allowable at all.

2012-01-24 18:23:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngdec.h:
	  pngdec: port to 0.11

2012-01-24 18:21:08 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	  pngenc: port to 0.11

2012-01-24 14:53:38 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix string leak

2012-01-24 14:52:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix use of freed memory

2011-12-01 15:49:40 +0100  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  Don't crash on empty laces
	  https://bugzilla.gnome.org/show_bug.cgi?id=665224

2012-01-23 13:15:46 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: don't reveal the user's username, hostname or real name by default
	  Send a randomly made-up user@hostname as CNAME and don't
	  send a NAME at all by default.
	  https://bugzilla.gnome.org/show_bug.cgi?id=668320

2012-01-21 20:07:56 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	  controller: move from control-binding to control-binding-direct

2012-01-22 23:31:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/glib-compat-private.h:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiochebband.h:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiocheblimit.h:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofirfilter.h:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioiirfilter.h:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsincband.h:
	* gst/audiofx/audiowsinclimit.c:
	* gst/audiofx/audiowsinclimit.h:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstaspectratiocrop.h:
	  Don't use deprecated GLib API

2012-01-22 23:15:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpclientsink.c:
	* gst-libs/gst/glib-compat-private.h:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	* gst/interleave/interleave.c:
	* gst/rtpmanager/gstrtpsession.c:
	* sys/oss4/oss4-mixer.c:
	* tests/check/elements/multifile.c:
	* tests/check/elements/souphttpsrc.c:
	* tests/icles/equalizer-test.c:
	* tests/icles/gdkpixbufsink-test.c:
	* tests/icles/test-oss4.c:
	* tests/icles/v4l2src-test.c:
	* tests/icles/videocrop-test.c:
	  Use new GLib API unconditionally

2012-01-20 17:06:42 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: simplify internal src event debug logging
	  ... which avoids almost superfluous obtaining of rtsp element.

2012-01-20 17:03:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid NULL string comparison

2012-01-20 17:03:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: arrange for initialized variables

2012-01-20 17:02:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	  rtpmp4adepay: prevent out-of-bound array access

2012-01-20 17:01:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/atomsrecovery.c:
	  isomp4: recovery: add sanity check
	  ... on possibly bogus/corrupt input data.

2012-01-20 17:00:51 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: remove dead code

2012-01-20 16:58:28 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: remove redundant variable

2012-01-20 16:57:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix arithmetic for unsigned comparison

2012-01-20 16:55:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: add various missing break

2012-01-20 16:54:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: tweak DEFAULT format duration query response

2012-01-20 16:49:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: remove redundant statement

2012-01-20 16:48:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: improve upstream peer duration querying
	  ... to avoid accepting unhandled duration query result.

2012-01-20 16:47:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: additional error condition checking

2012-01-20 16:46:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: additional error condition checking

2012-01-20 16:44:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: check _alloc_buffer result and perform fallback alloc if needed
	  ... rather than carrying on with NULL buffer.

2012-01-20 14:45:01 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	  controller: adapt to control binding changes

2012-01-20 11:37:38 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	  controller: adapt to controller api changes
	  Don't use the convenience api for control sources.

2012-01-19 14:24:04 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	* configure.ac:
	  Add --disable-fatal-warnings configure option

2012-01-19 12:44:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* gst/udp/gstmultiudpsink.c:
	  update for memory API

2012-01-19 11:33:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflacdec.c:
	* ext/jack/gstjackaudioclient.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpclientsink.h:
	* ext/wavpack/gstwavpackparse.c:
	* gst/avi/gstavidemux.c:
	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer.h:
	* gst/flv/gstflvdemux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/gstqtmoovrecover.h:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	* gst/videomixer/videomixer2.c:
	* gst/wavparse/gstwavparse.c:
	* sys/v4l2/gstv4l2videooverlay.c:
	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	* tests/check/elements/deinterleave.c:
	  port to new gthread API

2012-01-18 16:58:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure.ac: Remove GIO check, this is in gst-glib2.m4 now

2012-01-18 16:46:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 0807187 to 2a59016

2012-01-18 16:15:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure.ac: Require GLib 2.31.10 and improve GIO check

2012-01-17 16:58:07 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Remove unneeded socket.h include

2012-01-17 16:53:31 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* gst/rtp/Makefile.am:
	* gst/rtp/gstasteriskh263.c:
	  configure: Remove socket/winsock specific checks
	  Not necessary anymore.

2012-01-17 16:49:10 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Update for the new GIO versions of the udp elements

2012-01-17 13:08:42 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpmanager: Port to GIO

2012-01-17 11:19:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* gst/udp/Makefile.am:
	  configure: Require GIO 2.31.10

2012-01-17 11:18:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstudp.c:
	* gst/udp/gstudpnetutils.c:
	* gst/udp/gstudpnetutils.h:
	  udp: Remove now unecessary code

2012-01-17 11:18:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsink.h:
	  udpsink/multiudpsink: Port to GIO

2012-01-17 09:38:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstudpsrc.c:
	  dynudpsink: Port to GIO

2012-01-17 09:32:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstdynudpsink.h:
	  dynudpsink: Port to GIO

2012-01-17 09:03:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/udp/Makefile.am:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstudpnetutils.c:
	* gst/udp/gstudpnetutils.h:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: Port to GIO

2012-01-16 17:51:18 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/cutter/gstcutter.c:
	  cutter: fix leak of unused GValue

2012-01-16 16:10:08 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/elements/autodetect.c:
	  tests: fix autodetect test not testing correctly for state change success
	  State change to PAUSED can be done async, so if this happens, we need
	  to wait for the change to be done (or failed).

2012-01-16 15:42:46 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: fix caps leak

2012-01-16 12:13:50 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: make interlacedness test deterministic
	  If the interlaced flag is not present in the caps, we assume the
	  data is not interlaced, instead of leaving the boolean uninitialized.

2012-01-13 18:12:05 -0500  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/multifile/gstmultifilesink.c:
	  matroska: fix printf format compiler warnings
	  https://bugzilla.gnome.org/show_bug.cgi?id=662615

2012-01-13 18:11:36 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: fix wrong error check
	  pa_stream_* functions return negative on error, despite the defines
	  for error codes being positive.
	  I only got to repro the error twice, so I'm not sure 100% sure this
	  fixes the issue (the negative var being uninitialized after returning
	  from pa_stream_get_latency).

2012-01-13 17:43:49 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	  oss4: fix caps leaks

2012-01-13 17:25:59 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: fix caps leak

2012-01-13 15:57:20 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/elements/videocrop.c:
	  tests: fix caps leak in videotestsrc test

2012-01-13 12:50:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: clean up obsolete closing segment handling

2012-01-13 10:32:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: plug pad leak in error code path
	  Based on patch by: Stig Sandnes <stig.sandnes@cisco.com>
	  Don't leak srcpad if there are no caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667820

2011-10-04 10:00:02 +0200  Stig Sandnes <stigsand@cisco.com>

	* sys/osxvideo/cocoawindow.m:
	  osxvideo: Fix leak of NSOpenGLPixelFormat object
	  https://bugzilla.gnome.org/show_bug.cgi?id=667818

2011-09-05 10:43:19 +0200  Havard Graff <havard.graff@tandberg.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Don't assert when the interface is not implemented.
	  Simply return FALSE instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667817

2012-01-12 00:18:39 +0200  Raimo Järvi <raimo.jarvi@gmail.com>

	* sys/waveform/gstwaveformsink.c:
	* sys/waveform/gstwaveformsink.h:
	  waveformsink: Fix mingw warnings
	  https://bugzilla.gnome.org/show_bug.cgi?id=667719

2012-01-12 23:55:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/apetag/gstapedemux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/matroska/matroska-read-common.c:
	  GST_TYPE_DATE -> G_TYPE_DATE

2012-01-12 23:48:50 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  eqMerge remote-tracking branch 'origin/master' into 0.11
	  Conflicts:
	  ext/jack/gstjackaudiosink.c
	  ext/jack/gstjackaudiosrc.c
	  gst/matroska/matroska-mux.c
	  gst/matroska/matroska-read-common.c
	  gst/rtpmanager/gstrtpssrcdemux.c

2012-01-12 18:23:42 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  gstrtpssrcdemux: fix element leak

2012-01-12 14:19:22 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  matroska: do not leak attachment buffers

2012-01-12 13:17:55 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: remove obsolete FIXME comments

2012-01-12 10:30:11 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: do not drop the first data buffer on the floor (and leak it either)

2012-01-12 11:08:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstindex.c:
	* gst/flv/gstmemindex.c:
	  flvdemux: add prefix to local GstIndex related copies
	  ... to avoid duplicate type names with other such local copies in the wild.

2012-01-12 11:07:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: activate pad before setting caps
	  ... rather than the usual 0.10 other way around.
	  Fixes #667558.

2012-01-11 18:45:33 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>

	* Android.mk:
	  Temporarily disabling multifile for the Android build
	  There is a hard dependency on inotify comming from gio. We
	  are not currently bundling inotify with the Android dist so
	  I'm disabling multifile for now until someone gets around
	  to sort this out.
	  This change fixes building on Android

2010-10-20 02:17:43 -0700  Leo Singer <leo.singer@ligo.org>

	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* tests/check/elements/audioiirfilter.c:
	  audiofx: Use most common convention for definitions of IIR filter coefficients.
	  Most signal processing texts, including MATLAB, use the following convention for IIR filter coefficients:
	  a_0 y[n] + a_1 y[n-1] + ... + a_M y[n-M] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N]
	  Usually, a_0 is set to 1 because the coefficients can always be rescaled, giving
	  y[n] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N] - a_1 y[n-1] - ... - a_M y[n-M]
	  The convention that was previously used by audiofxbaseiirfilter and derived class had the a and b coefficients swapped, and did not have the minus signs.
	  This change makes the audiofx plugin use the more common convention described above.

2012-01-11 14:47:36 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: add a transport mode enum
	  Clients can configure the desired behaviour via "transport" property. The
	  default behaviour is ignoring the transport state. Other modes are master and
	  slave.

2012-01-11 14:10:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Fix buffer handling
	  souphttpsrc is now usable again and doesn't crash anymore
	  whenever something is read from a HTTP connection.

2012-01-11 01:45:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/pipelines/wavenc.c:
	  tests: fix wavenc test on big endian
	  wavenc only accepts little-endian PCM, but most of our
	  elements such as audiotestsrc only produce or process
	  audio in native endianness, so we need to plug a
	  converter before wavenc on big endian systems.

2012-01-10 23:02:45 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: deactivate the request_state code
	  When qjackctl is started, transport is stopped by default. This would be a
	  regression for gstreamer apps that before just started to play right away.

2012-01-10 22:27:11 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudioclient.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: add transport control handling
	  This feature allows to start and stop playback from other jack applications (e.g. qjackctl).

2012-01-10 18:50:27 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix codec_priv leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=667419

2012-01-10 15:17:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/a52dec/gsta52dec.c
	  ext/a52dec/gsta52dec.h
	  ext/lame/gstlame.c
	  ext/lame/gstlame.h
	  ext/lame/gstlamemp3enc.c
	  ext/mad/gstmad.c
	  ext/mad/gstmad.h
	  gst/mpegaudioparse/gstmpegaudioparse.c
	  gst/mpegstream/gstdvddemux.c
	  gst/realmedia/rdtdepay.c
	  po/es.po
	  po/lv.po
	  po/sr.po

2012-01-10 15:06:39 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/jack/gstjackaudioclient.c:
	  jack: use jack type for the callback
	  Jack headers have a typedef for the shutdown callback as well.

2012-01-10 14:32:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/cairo/gsttextoverlay.c
	  ext/pulse/pulseaudiosink.c
	  gst/audioparsers/gstaacparse.c
	  gst/avi/gstavimux.c
	  gst/flv/gstflvmux.c
	  gst/interleave/interleave.c
	  gst/isomp4/gstqtmux.c
	  gst/matroska/matroska-demux.c
	  gst/matroska/matroska-mux.c
	  gst/matroska/matroska-mux.h
	  gst/matroska/matroska-read-common.c
	  gst/multifile/gstmultifilesink.c
	  gst/multipart/multipartmux.c
	  gst/shapewipe/gstshapewipe.c
	  gst/smpte/gstsmpte.c
	  gst/udp/gstmultiudpsink.c
	  gst/videobox/gstvideobox.c
	  gst/videocrop/gstaspectratiocrop.c
	  gst/videomixer/videomixer.c
	  gst/videomixer/videomixer2.c
	  gst/wavparse/gstwavparse.c
	  po/ja.po
	  po/lv.po
	  po/sr.po
	  tests/check/Makefile.am
	  tests/check/elements/qtmux.c
	  tests/check/elements/rgvolume.c

2012-01-09 22:58:32 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* docs/plugins/Makefile.am:
	  docs: Remove old videomixer headers
	  These got removed in the transition to videomixer2.

2012-01-09 17:28:17 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix codec string leaks

2012-01-09 14:51:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	* gst/videomixer/videomixerpad.h:
	  videomixer: Remove videomixer and register videomixer2 as videomixer

2012-01-09 11:36:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: initialize variable to avoid undefined use

2012-01-06 09:40:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flac: Port to the new raw audio caps

2012-01-05 19:25:33 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  isomp4: fix caps leak

2012-01-05 19:08:03 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  isomp4: remove dead assignment

2012-01-05 14:18:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/auparse/gstauparse.c:
	* gst/wavenc/gstwavenc.c:
	  fix pad templates

2012-01-04 15:44:37 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/twolame/gsttwolamemp2enc.c:
	  twolamemp2enc: Update for the new raw audio caps

2012-01-04 15:45:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: Update for the new raw audio caps

2012-01-04 15:05:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	  speex: Update for the new raw audio caps

2012-01-04 14:54:10 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: Add the new layout field to the raw audio caps

2012-01-04 14:52:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackutil.c:
	* ext/jack/gstjackutil.h:
	  jackaudiosrc: Port to the new multichannel audio caps

2012-01-04 14:13:54 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Add FLAC and interleave to the non-ported plugins list
	  Both need to be updated to the audio/x-raw caps and were only
	  half-ported before.

2012-01-04 13:48:36 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpchannels.c:
	* gst/rtp/gstrtpchannels.h:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: Update for the new audio caps

2012-01-04 12:06:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Update for libgstriff API changes
	  Still needs to handle raw audio channel reordering

2012-01-04 12:05:16 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Update for the new raw audio caps

2012-01-04 12:03:50 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	  spectrum: Update for the new raw audio caps layout field

2012-01-04 11:57:20 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	  replaygain: Update for the new audio caps

2012-01-04 11:52:29 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: Update for the new raw audio interleaved caps field
	  Still needs to be fixed to handle the multichannel channel-mask
	  and reordering.

2012-01-04 11:31:07 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/level/gstlevel.c:
	  level: Update for the new raw audio layout field

2012-01-04 11:29:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/qtdemux.c:
	  isomp4: Port to the new audio caps
	  Still needs to handle the channel positions/masks and
	  channel reordering.

2012-01-04 11:11:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/cutter/gstcutter.c:
	  cutter: Update for the new raw audio layout field

2012-01-04 11:09:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: Port to the new multichannel caps and update for the new raw audio layout field

2012-01-04 11:08:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Update for the new raw audio layout field

2012-01-04 11:07:29 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Update for the libgstriff API changes
	  Still needs to do reordering of channels for raw audio.

2012-01-04 11:06:28 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/auparse/gstauparse.c:
	  auparse: Port to the new multichannel caps and the new raw audio layout field

2012-01-04 11:02:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	  audiofx: Port to the new multichannel caps and the new raw audio layout field

2012-01-04 10:54:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	  oss: Port to the new multichannel caps and the raw audio caps interleaved field

2012-01-04 10:27:09 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	  pulse: Port to the new multichannel caps

2012-01-04 19:51:46 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 762b692 to 0807187

2012-01-04 17:05:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/Makefile.am:
	  lame: fix LIBADD order in Makefile.am

2012-01-04 17:59:55 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: fix some leaks and remove files when done in qtmux test

2011-12-14 10:14:20 +0100  Peter Seiderer <ps.report@gmx.net>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: post better error message when we run out of disk space
	  Map write errno ENOSPC to GST_RESOURCE_ERROR_NO_SPACE_LEFT.

2012-01-04 13:26:45 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	* tests/check/elements/alphacolor.c:
	  alphacolor: More fixes/cleanup

2012-01-04 13:25:40 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Refactor param/process functions
	  When ::set_info() is called, the input/output VideoInfo aren't set
	  yet on the videofilter.

2012-01-04 10:01:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	* ext/dv/gstdvdemux.c:
	* ext/libpng/gstpngdec.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/wavpack/gstwavpackparse.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/interleave/interleave.c:
	* gst/videomixer/videomixer2.c:
	  GST_FLOW_UNEXPECTED -> GST_FLOW_EOS

2011-12-31 23:33:33 -0500  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: use right variable
	  Fixes use of unitialized variable.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667085

2012-01-03 15:26:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/soup/gstsouphttpsrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/ebml-read.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/wavparse/gstwavparse.c:
	  GST_FLOW_UNEXPECTED -> GST_FLOW_EOS

2012-01-03 14:42:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  tests: rewrite test a little
	  Rewrite the tag check so that we don't need to deal with tag lists.

2012-01-03 14:16:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/jpegenc.c:
	* tests/check/elements/multifile.c:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtpbin.c:
	* tests/check/elements/rtpbin_buffer_list.c:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/shapewipe.c:
	* tests/check/elements/souphttpsrc.c:
	* tests/check/elements/udpsink.c:
	* tests/check/elements/videocrop.c:
	* tests/check/elements/videofilter.c:
	* tests/check/elements/y4menc.c:
	* tests/check/pipelines/flacdec.c:
	* tests/check/pipelines/tagschecking.c:
	  tests: make more tests compile

2012-01-03 11:56:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/equalizer.c:
	* tests/check/elements/flacparse.c:
	* tests/check/elements/flvdemux.c:
	* tests/check/elements/flvmux.c:
	* tests/check/elements/icydemux.c:
	* tests/check/elements/imagefreeze.c:
	* tests/check/elements/interleave.c:
	* tests/check/elements/level.c:
	* tests/check/elements/multifile.c:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rglimiter.c:
	* tests/check/elements/rgvolume.c:
	  test: make more unit tests compile

2012-01-03 10:26:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/audiofirfilter.c:
	* tests/check/elements/audioiirfilter.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/audiowsincband.c:
	* tests/check/elements/audiowsinclimit.c:
	* tests/check/elements/autodetect.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/avisubtitle.c:
	* tests/check/elements/capssetter.c:
	* tests/check/elements/deinterlace.c:
	* tests/check/elements/deinterleave.c:
	* tests/check/generic/index.c:
	* tests/check/generic/states.c:
	  tests: fix some unit tests
	  Remove unit test for GstIndex.
	  Make some other unit tests compile

2012-01-02 14:32:40 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/rtsp/gstrtspext.c:
	  autodetect, rtsp: gst_registry_get_default() -> gst_registry_get()

2011-12-31 10:00:41 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/v4l2/camctrl.c:
	  controller: port to API changes

2011-12-30 17:41:46 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: update for GstIndex removal

2011-12-30 17:23:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: update for GstIndex removal

2011-12-30 17:20:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/Makefile.am:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstindex.c:
	* gst/flv/gstindex.h:
	* gst/flv/gstmemindex.c:
	  flvdemux: update for GstIndex removal
	  Add private GstMemIndex for now.

2011-12-30 17:12:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: update for GstIndex removal

2011-12-27 22:59:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/waveform/gstwaveformsink.c:
	  waveformsink: fix compiler warnings with MingW
	  https://bugzilla.gnome.org/show_bug.cgi?id=666485

2011-12-27 22:54:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  lame: fix printf format in debug statements
	  https://bugzilla.gnome.org/show_bug.cgi?id=666926

2011-12-27 12:06:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/.gitignore:
	  tests: make git ignore new unit test binary

2011-12-27 11:50:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix valgrind warning
	  https://bugzilla.gnome.org/show_bug.cgi?id=666644

2011-12-27 11:49:10 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/udpsrc.c:
	  udpsrc: add unit test that sends 0-size packet
	  https://bugzilla.gnome.org/show_bug.cgi?id=666644

2011-12-21 13:22:03 +0100  John Ogness <john.ogness@linutronix.de>

	* gst/udp/gstudpsrc.c:
	  udpsrc: drop dataless UDP packets
	  It is allowed to send/receive UDP packets with no data. When such
	  a packet is available, select() will return with success but
	  ioctl(FIONREAD) will return 0. But a read() must still occur in
	  order to clear off the UDP packet from the queue.
	  This patch will read the dataless packet from the socket. If
	  select() was woken for other reasons (and FIONREAD returns 0),
	  this may result in a UDP packet getting accidentally dropped.
	  But since UDP is not reliable, this is acceptable.
	  NOTE: This patch fixes a nasty bug where sending a dataless
	  UDP packet to a udpsrc instance will cause an infinite
	  loop.
	  https://bugzilla.gnome.org/show_bug.cgi?id=666644
	  Signed-off-by: John Ogness <john.ogness@linutronix.de>

2011-12-26 22:22:59 +0000  Yaakov Selkowitz <yselkowitz@users.sourceforge.net>

	* configure.ac:
	* sys/Makefile.am:
	* sys/waveform/Makefile.am:
	  waveform: add autotools bits for waveform plugin
	  https://bugzilla.gnome.org/show_bug.cgi?id=666485

2011-12-21 20:50:21 +0100  Nicola Murino <nicola.murino@gmail.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix peer_caps leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=666688

2011-12-26 18:24:32 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  lame: ensure parsed output
	  ... by doing some basic parsing of encoded lame data.

2011-12-26 16:34:01 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/gstlame.h:
	  lame: cleanup unused instance struct fields

2011-12-26 18:23:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/Makefile.am:
	* ext/lame/gstlamemp3enc.c:
	* ext/lame/gstlamemp3enc.h:
	  lamemp3enc: ensure parsed output
	  ... by doing some basic parsing of encoded lame data.
	  Fixes #652150.

2011-12-26 18:15:41 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: do not leak merged tags

2011-12-25 23:52:46 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: remove unnecessary check for gdp library

2011-12-25 22:17:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* ext/pulse/Makefile.am:
	* ext/pulse/plugin.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulse: remove pulseaudiosink helper bin
	  This is causing us lots of headaches in 0.10 and needs to be done
	  differently and properly in 0.11. playbin or decodebin should
	  reconfigure themselves based on reconfigure events, for example.

2011-12-25 21:45:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	  pulse: update for ring buffer audio format type enum rename

2011-12-25 20:34:52 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/v4l2/camctrl.c:
	  controller: port to new control source api

2011-12-25 14:23:29 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: don't try to push already-freed buffers
	  Fixes unit test.

2011-12-24 10:57:42 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Use scale_ceil() functions from core instead of custom ones

2011-12-21 23:51:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavigationtest.h:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstaging.h:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstdice.h:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstedge.h:
	* gst/effectv/gstop.c:
	* gst/effectv/gstop.h:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstquark.h:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstradioac.h:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstrev.h:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstripple.h:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstshagadelic.h:
	* gst/effectv/gststreak.c:
	* gst/effectv/gststreak.h:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstvertigo.h:
	* gst/effectv/gstwarp.c:
	* gst/effectv/gstwarp.h:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstgamma.h:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideobalance.h:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	  update for videofilter changes.

2011-12-21 17:43:10 +0100  Branko Subasic <branko@axis.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: do not consider duration of non-finalized file
	  ... to avoid it clamping requested seek position.
	  Non-finalized file case, determined by whether
	  _parse_blockgroup_or_simpleblock ever updates the segment duration.
	  Fixes #652195.

2011-12-21 15:06:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: improve decision to fall back to scanning when seeking
	  ... which is basically iff not streaming and no entry found in index

2011-12-21 09:09:27 +0100  Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

	* gst/audioparsers/gstaacparse.c:
	  ac3parse: remove unused variable
	  remove unused variable to fix compile error:
	  make -C audioparsers
	  make[3]: Betrete Verzeichnis '/home/lex/tmp/gst-plugins-good/gst/audioparsers'
	  CC     libgstaudioparsers_la-gstaacparse.lo
	  gstaacparse.c: In function 'gst_aac_parse_read_loas_audio_specific_config':
	  gstaacparse.c:446:12: error: variable 'sbr' set but not used [-Werror=unused-but-set-variable]
	  cc1: all warnings being treated as errors
	  Signed-off-by: Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>

2011-12-21 11:59:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsemixer.c:
	* ext/pulse/pulseprobe.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* tests/examples/pulse/pulse.c:
	* tests/examples/v4l2/Makefile.am:
	* tests/examples/v4l2/probe.c:
	  update for removed property probe

2011-09-09 11:42:09 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: let bsid 9 and 10 through
	  Files with 9 and 10 happen, and seem to comply with the <= 8
	  format, so let them through.
	  The spec says nothing about 9 and 10.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658546

2011-12-19 23:50:19 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/v4l2/camctrl.c:
	  controller: port to new interpolation-mode api

2011-12-19 22:53:57 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/v4l2/camctrl.c:
	  controller: port to new controller api

2011-12-19 19:03:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: update for new interlaced caps

2011-12-16 19:15:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: properly determine final duration
	  ... which can be authoratively obtained from our own written timestamps.

2011-12-19 13:56:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: only write full metadata at start
	  ... rather than having (potentially) unnecessary duplicates written all over,
	  or even contradictory varying filesize info, or duration info that will not
	  be rewritten upon header rewrite.

2011-12-16 19:15:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: use GstCollectPads2 buffer callback and running time clipper
	  ... since the default collection heuristics suffice.

2011-12-16 18:03:01 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: use GstCollectPads2 buffer callback and running time clipper
	  ... since default collection heuristics suffice.

2011-12-16 17:20:51 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: bring a few debug statements up to specs
	  ... and minor spelling fix.

2011-12-16 16:56:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: additional subtitle support

2011-12-15 21:50:42 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: additional buffer handling cleanup

2011-12-15 21:45:17 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: use GstCollectPads2 buffer callback and running time clipper

2011-12-07 13:24:55 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	  aacparse: parse LOAS variant
	  The LOAS variant seems to have three different subvariants itself,
	  only one of them is implemented as my two samples happen to be
	  using that one.
	  The sample rate is not always reported correctly, as the "main"
	  sample rate is apparently sometimes half what it should be (both
	  of my samples report 24000 Hz there), and there are two other
	  parts of the subvariant with different sampling rates. One of them
	  is parsed, but not the other, as it's located after some other
	  large amount of variable data that needs parsing first, and there
	  seems to be a LOT of it, which is useless for our needs here.
	  This ends up being rather inconsequential, as ffdec_aac_latm,
	  which is the only decoder that can decode such streams, does not
	  need the sample rate on the caps anyway.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665394

2011-12-19 10:48:54 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: don't remove srcpad
	  Don't remove the always srcpad in ready and make the element reusable.

2011-12-15 16:40:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: use GstCollectPads2 event callback
	  ... in stead of local HACK.

2011-12-15 16:30:17 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: use GstCollectPads2 event callback
	  ... in stead of local HACK.

2011-12-15 16:16:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avimux: use GstCollectPads2 event callback
	  ... in stead of local HACK.

2011-12-15 16:15:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: use GstCollectPads2 event callback
	  ... in stead of local HACK.

2011-12-14 19:13:21 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmpte.h:
	  smpte: port to GstCollectPads2

2011-12-14 19:10:53 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/multipart/multipartmux.c:
	* gst/multipart/multipartmux.h:
	  multipartmux: port to GstCollectPads2

2011-12-14 19:07:23 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: port to GstCollectPads2

2011-12-14 19:02:23 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: port to GstCollectPads2

2011-12-14 18:55:36 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	  interleave: port to GstCollectPads2

2011-12-14 18:52:37 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flxmux: port to GstCollectPads2

2011-12-14 18:38:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avimux: port to GstCollectPads2

2011-12-14 18:34:25 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttextoverlay.h:
	  cairotextoverlay: port to GstCollectPads2

2011-12-13 18:18:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: filter bogus index entries with missing block number
	  ... to avoid contradictory information resulting in seeks sending more
	  downstream than needed for the corresponding segment.

2011-12-13 18:15:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: cater for safer arithmetic with global start time

2011-12-13 17:02:01 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: tweak final closing segment sending
	  ... to avoid it interfering with (sparse) stream syncing.

2011-12-12 11:51:06 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: make debug message more useful
	  Add information about the taglist and which pad received the
	  tag event on the debug logging.

2011-12-13 11:46:43 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: avoid using floating point unnecessarily
	  https://bugzilla.gnome.org/show_bug.cgi?id=665911

2011-12-13 11:42:40 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: fix format specifier signedness
	  Use unsigned specifiers for all unsigned values.
	  A lot of the values used here are unsigned, and some can take
	  high enough values that their signed counterpart will be negative.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665911

2011-12-12 16:49:19 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: add a ignore-length property
	  This allows playing broken streams which write an incorrect
	  length in their data chunks (such as, at least, one streaming
	  camera).
	  https://bugzilla.gnome.org/show_bug.cgi?id=665911

2011-12-12 11:54:56 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/glib-compat-private.h:
	  glib-compat: Add license boilerplate for LGPL

2011-12-12 15:15:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: mind (un)signed in some timestamp arithmetic
	  ... to avoid ending up with invalid (negative) duration.

2011-02-09 15:31:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: increase parse tolerance for fuzzy file cases

2011-12-12 10:38:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	  build: dist glib-compat-private.h properly
	  Add missing slash.

2011-12-12 10:18:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/souphttpsrc.c:
	  tests: use atexit, g_atexit has been deprecated in glib master

2011-12-12 02:52:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflacdec.c:
	* ext/wavpack/gstwavpackparse.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/videomixer/videomixer2.c:
	* gst/wavparse/gstwavparse.c:
	  Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly
	  GStaticRecMutex is part of our API/ABI, not much we can do here
	  in 0.10 for most of these.

2011-12-12 02:41:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/souphttpsrc.c:
	* tests/icles/equalizer-test.c:
	* tests/icles/gdkpixbufsink-test.c:
	* tests/icles/test-oss4.c:
	* tests/icles/videocrop-test.c:
	  tests: g_thread_init() is deprecated in glib master
	  It's not needed any longer.

2011-12-12 02:38:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpclientsink.c:
	* gst/rtpmanager/gstrtpsession.c:
	* sys/oss4/oss4-mixer.c:
	* tests/icles/v4l2src-test.c:
	  Use g_thread_try_new() instead of g_thread_crate() with newer glib versions

2011-12-12 02:31:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: use new glib API for static mutex if available

2011-12-12 02:30:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	* ext/jack/gstjackaudioclient.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/soup/gstsouphttpclientsink.c:
	* gst-libs/gst/glib-compat-private.h:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/equalizer/gstiirequalizer.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer2.c:
	* sys/oss4/oss4-mixer.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/ximage/gstximagesrc.c:
	  Work around deprecated thread API in glib master
	  Add private replacements for deprecated functions such as
	  g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
	  to avoid the deprecation warnings. We'll change these
	  over to the new API once we depend on glib >= 2.32.

2011-12-12 10:24:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Require GLib >= 2.24
	  All other modules require this already and nobody is testing with
	  older versions anyway.

2011-12-11 18:40:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	  gdkpixbufsink: fix inverted pixel-aspect-ratio
	  Spotted by Mike Morrison.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665882

2011-12-11 17:55:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: don't leak pad template

2011-12-10 14:48:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpclientsink.c:
	  soup: fix start/stop race in souphttpclientsink
	  Fix crash or hang in generic/states unit test when doing stop()
	  right after start(). Create main loop in the start function already
	  and not just in the thread function, so that stop() always has a
	  valid main loop to quit on. Also, calling g_main_loop_quit() before
	  g_main_loop_run() won't work and result in the stop function waiting
	  for the thread to join forever. Therefore, wait for the thread to
	  be ready and get the main loop running in the start() function, to
	  be sure stop() always works.

2011-12-10 13:35:08 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/files/Makefile.am:
	  tests: dist test file used in matroskaparse unit test

2011-12-10 12:32:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/rgvolume.c:
	  tests: fix up rgvolume test for basetransform event caching
	  Some tests assumed that tag events would always pushed through
	  immediately, which isn't the case any longer, so push a newsegment
	  event and an empty buffer first.

2011-12-10 11:12:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  ssrcdemux: fix iterator and caps

2011-12-10 11:11:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: forward the caps event

2011-12-10 11:09:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: simply forward the caps event
	  forward the caps event we get as input instead of making a new event etc..

2011-12-09 20:10:19 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: forward caps

2011-12-09 19:46:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtp: pass parent to setcaps methods

2011-12-10 02:21:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/LINGUAS:
	* po/eo.po:
	* po/ja.po:
	* po/lv.po:
	* po/sr.po:
	  po: update translations

2011-12-09 16:04:56 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulse: rename "client" properties to "client-name"
	  Better name, but also matches the property on the jack
	  elements (where "client" is used for something else).

2011-12-09 15:50:28 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: don't leak client name when freeing the element
	  And add gtk-doc chunks for the new property.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665872

2011-12-09 15:45:03 +0000  Nicolas Baron <hoggins@radiom.fr>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: add "client-name" property to jackaudiosink and jackaudiosrc
	  https://bugzilla.gnome.org/show_bug.cgi?id=665872

2011-12-09 12:19:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/law/Makefile.am:
	  law: fix CFLAGS and LIBS order in Makefile.am

2011-12-09 12:15:30 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11

2011-12-09 10:51:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtp: fix marshallers
	  Remove custom marshallers for minobject.
	  Init RTCP buffer correctly.
	  Handle results from setcaps
	  Remove asserts.

2011-12-09 10:50:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/law/Makefile.am:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/alaw.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	  law: fix negotiation

2011-12-08 11:00:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: stream-format=raw goes with aac caps, not mp3 caps

2011-12-08 01:28:26 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11
	  Conflicts:
	  sys/v4l2/gstv4l2object.c

2011-12-02 12:07:24 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: do not ignore the highest frame interval
	  https://bugzilla.gnome.org/show_bug.cgi?id=665387

2011-12-02 11:59:03 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: do not ignore the largest resolution
	  The 'max' value isn't an STL style "one after the end" bound,
	  but the largest allowed value.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665387

2011-12-06 16:47:25 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/multifile/gstmultifilesink.h:
	  docs: add add the two enum values that were just added too

2011-12-06 16:14:54 +0100  Stefan Sauer <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: expose the enum property docs for splitting mode.
	  Fixes #665666.

2011-12-06 14:23:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263pay.c:
	  h263pay: fix invalid return value

2011-12-06 13:59:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove unused flush param

2011-12-05 18:40:26 +0100  Edward Hervey <edward@collabora.com>

	* gst/isomp4/gstrtpxqtdepay.c:
	  rtpxqtdepay: Initialize GstRTPBuffer before usage

2011-12-05 18:40:12 +0100  Edward Hervey <edward@collabora.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: Initialize GstRTPBuffer before usage

2011-12-05 18:39:59 +0100  Edward Hervey <edward@collabora.com>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: Initialize GstRTPBuffer before usage

2011-12-05 12:15:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: replace deprecated GST_CLASS_LOCK

2011-11-24 13:58:01 +0100  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Ceil jpeg dimensions, instead of floor
	  A JPEG image inside an RTP stream has a preceeding RFC2435 header that
	  conveys width/height. The dimensions in this header are limited to be
	  multiples of 8. Since JPEG uses an MCU of 8x8 pixels any image must
	  already indirectly have image data dimensions that are rounded up in
	  order to contain enough data to render the image. Therefore this fix
	  safely rounds the image dimensions in the RFC2435 header up to the
	  closest multiple of 8.

2011-12-04 12:50:57 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: ensure we only check for sample/block mixup at start
	  Otherwise we might trigger at some point within the file, but the
	  check is only making sense for the second block.

2011-12-03 18:14:59 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: warn if accumulating headers after they were pushed
	  https://bugzilla.gnome.org/show_bug.cgi?id=665412

2011-10-25 12:54:43 -0700  David Schleef <ds@schleef.org>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: fix parsing
	  Mark more parts as belonging to streamheaders.

2011-12-03 17:30:10 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix discontinuity threshold check when timestamps go backwards
	  Since unsigned types are used, a negative value would show as very, very
	  positive.
	  Fixes A/V sync on some... less than well made files where timestamps go
	  backwards.

2011-12-02 22:25:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/soup/gstsouphttpclientsink.c:
	* gst/debugutils/testplugin.c:
	* gst/multifile/gstmultifilesink.c:
	  update for basesink event handler changes

2011-12-02 12:01:22 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: add a comment about a "hidden" assumption on rank values
	  https://bugzilla.gnome.org/show_bug.cgi?id=665387

2011-12-02 01:58:30 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11
	  Conflicts:
	  docs/plugins/inspect/plugin-esdsink.xml
	  docs/plugins/inspect/plugin-gconfelements.xml
	  ext/pulse/pulseaudiosink.c
	  gst/matroska/matroska-demux.c
	  gst/matroska/matroska-mux.c
	  gst/multifile/gstmultifilesink.c

2011-12-01 18:55:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-read-common.c:
	* tests/check/elements/id3demux.c:
	  update for tag API changes

2011-12-01 15:29:15 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: placate gcc since -Werror is used
	  Initialize values that GCC cannot prove are not used without
	  being initialized, and assert that I did not mess up my proof.

2011-12-01 14:13:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	  tests: fix up LIBS order som more`

2011-12-01 13:22:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: fix name of new property and the unit test
	  https://bugzilla.gnome.org/show_bug.cgi?id=654379

2011-09-25 14:57:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: add basic buffer list handling
	  We assume for now that all buffers in a buffer list
	  should end up in the same file (so we can group GOPs
	  in buffer lists, for example). Could optimise this
	  a bit to avoid the memcpy.

2011-09-23 18:43:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: write stream-headers when switching to the next file in max-size mode

2011-09-23 18:31:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: add new 'max-size' mode for switching to the next file

2011-09-23 17:49:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: add "max-file-size" property for new next-file mode

2011-12-01 13:38:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't forget SSA subtitles in last commit

2011-12-01 13:34:52 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Only check for markup and escape if necessary for plaintext subtitles
	  Otherwise we break USF and ASS/SSA subtitles.

2011-12-01 13:23:33 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/Makefile.am:
	  multifile: fix build in uninstalled setup
	  Add -base libs includes to CFLAGS, fix order of LIBS <cit>.

2011-12-01 13:08:01 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* tests/check/elements/multifile.c:
	  tests: fix g_mkdtemp presence check in multifile tests
	  g_mkdtemp was added in glib 2.30 even though the doc claims it was added in
	  2.26.

2011-07-17 23:56:04 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/Makefile.am:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	* tests/check/Makefile.am:
	* tests/check/elements/multifile.c:
	  multifilesink: add flag to cut after a force key unit event

2011-12-01 12:47:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Copy all buffer flags when creating a subtitle buffer copy after postprocessing
	  This also copies the caps. Otherwise we could end up pusing
	  the first buffer without any caps, which causes downstream
	  to not get notified about the caps.
	  Fixes bug #664892.

2011-10-11 02:07:13 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: make default framerate optional per stream
	  there is at least two use cases where default frame rate
	  should or may be disabled:
	  - vp8 stream with altref frame enabled. If default frame rate
	  is enabled, some players will missinterprete it (critical!)
	  - for webm container, to reduce micro overhead
	  - for stream with variable frame rate.
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>

2011-11-30 22:13:11 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstripple.c:
	  rippletv: fix CLAMP end-values

2011-11-30 19:25:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update docs

2011-11-30 19:00:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/Makefile.am:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/patternspec.c:
	* gst/multifile/patternspec.h:
	  splitfilesrc: specify filenames via normal wildcards instead of regular expressions
	  Less cracktastic in the end.

2011-10-10 18:28:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: check bytes actually read, just in case
	  Handle corner case where we try to read beyond the end of the
	  last file part, in which case we want to return a short read.
	  If we get fewer bytes than expected for any other file part,
	  we should just error out, since something fishy's going on
	  then.

2011-10-06 08:33:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: set offsets on buffers
	  Looks like some parsers (in some versions at least) expect the
	  offsets to be set, and behave weird if that's not the case
	  (e.g. off-by-one in h264parse).

2011-07-28 20:19:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* gst/multifile/Makefile.am:
	* gst/multifile/gstmultifile.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/gstsplitfilesrc.h:
	  multifile: add splitfilesrc element
	  Add new splitfilesrc element that presents multiple files
	  (selectable via a location regex) as one single contiguous
	  file.

2011-11-30 07:57:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsemixerctrl.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  update for moved audio interfaces

2011-11-29 17:34:10 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/pulse/pulseaudiosink.c:
	  Revert "pulseaudiosink: fix caps leak"
	  This reverts commit d6a9de9e2aedc8b66ab3219902b5a37e8d65ada2.
	  setcaps functions aren't supposed to take ownership of the caps passed

2011-11-29 19:10:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstvideobalance.c:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2videooverlay.h:
	* sys/v4l2/gstv4l2vidorient.h:
	* tests/icles/Makefile.am:
	* tests/icles/v4l2src-test.c:
	  fix for moved interfaces

2011-11-28 23:20:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge commit '7521b597f4dc49d8d168f368f0e7ebaf98a72156' into 0.11

2011-11-28 21:31:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11

2011-11-28 21:31:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11

2011-11-28 21:27:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11

2011-11-28 21:27:40 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge commit 'a2337b8af45cb5e8c091ff0e1c3ef4b6cc7b20a3' into 0.11

2011-11-28 18:25:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	  Update for indexable change

2011-11-28 17:52:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtsp/gstrtpdec.c:
	  update for clock provider API change

2011-11-28 16:57:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/rtsp/gstrtspsrc.c:
	  fix for element flag updates

2011-11-28 12:58:44 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairorender.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttimeoverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/esd/esdmon.c:
	* ext/esd/esdsink.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/gconf/gstswitchsink.c:
	* ext/gconf/gstswitchsrc.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/hal/gsthalaudiosink.c:
	* ext/hal/gsthalaudiosrc.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libmng/gstmngdec.c:
	* ext/libmng/gstmngenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/mikmod/gstmikmod.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/taglib/gsttaglibmux.c:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackparse.c:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/efence.c:
	* gst/debugutils/gstcapsdebug.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/interleave/deinterleave.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	* gst/median/gstmedian.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer2.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* gst/y4m/gsty4mencode.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	* tests/check/elements/qtmux.c:
	  various: fix pad template leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=662664

2011-11-28 13:10:01 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  various: fix pad template ref leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=662664

2011-11-28 13:10:01 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/twolame/gsttwolame.c:
	  various: fix pad template ref leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=662664

2011-11-28 13:08:27 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  various: fix pad template ref leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=662664

2011-11-28 11:47:11 +0100  Chad <channa@caltech.edu>

	* gst/debugutils/gsttaginject.c:
	  taginject: set gap-aware
	  The element does not modify the data anyway.

2011-11-27 23:32:18 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update po files

2011-11-27 23:31:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11
	  Conflicts:
	  gst/equalizer/gstiirequalizer.c

2011-11-26 21:39:33 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: also sync the parameters for the filter bands

2011-11-26 16:06:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-ids.c:
	  matroskademux: initialise seen_markup_tag field on subtitle stream context

2011-11-26 10:01:07 +0100  René Stadler <rene.stadler@collabora.co.uk>

	* configure.ac:
	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-read.h:
	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	* gst/matroska/webm-mux.c:
	* tests/check/elements/matroskamux.c:
	  matroska: port to 0.11
	  Support for TAG_IMAGE and TAG_ATTACHMENT is commented out; this requires caps
	  on buffers which is gone from 0.11.
	  Segment handling in the demuxer is a bit complex; I added some FIXME comments
	  in places where I'm not yet sure if I ported correctly.

2011-11-26 13:54:22 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* ext/pulse/plugin.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulseaudio: require pulseaudio >= 1.0

2011-11-26 13:34:10 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11
	  Conflicts:
	  ext/pulse/pulseaudiosink.c
	  ext/pulse/pulsesrc.c
	  gst/audioparsers/gstaacparse.c
	  gst/audioparsers/gstamrparse.c
	  gst/audioparsers/gstdcaparse.c
	  gst/audioparsers/gstflacparse.c
	  gst/effectv/gstradioac.c
	  gst/effectv/gstradioac.h
	  gst/effectv/gstripple.c
	  Some possible FIXMEs remaining in the audio parser getcaps functions.

2011-11-25 19:28:55 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/gstqtmuxmap.c:
	  ismlmux: Use iso-fragmented as variant type
	  Using 'iso' conflicts with mp4mux variant type, ismlmux now
	  uses iso-fragmented
	  Fixes #656823

2011-11-24 12:05:33 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: Implement GstStreamVolume interface
	  PulseAudio 1.0 supports per-source-output volumes, and this exposes the
	  functionality via the GstStreamVolume interface.
	  When compiled against pre-1.0 PulseAudio, the interface is not
	  implemented, and the "volume" or "mute" properties are not available.
	  This bit of ugliness will go away when we can depend on PulseAudio 1.0
	  or greater.
	  https://bugzilla.gnome.org/show_bug.cgi?id=595055

2011-09-10 21:21:38 -0700  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Trivial comment copy-paste-o fix

2011-11-14 12:43:27 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: Remove redundant code

2011-11-14 12:41:41 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: Clean up refcounting in event probe
	  Makes sure we don't leak a refcount if the object is disposed before a
	  NEWSEGMENT turns up.

2011-11-24 16:31:38 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix seeking
	  Which I accidentally broke when fixing flv videos breaking on
	  spurious timestamp discontinuities in broken files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=631430

2011-11-25 13:13:47 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstradioac.c:
	* gst/effectv/gstradioac.h:
	  effectv: repair color modes in radioactv by taking rgb,bgr into account

2011-11-25 11:44:49 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstradioac.c:
	  radioactv: add one more set of caps
	  It also work in this format. Avoids the need for conversion.

2011-11-25 11:44:18 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstradioac.c:
	* gst/effectv/gstshagadelic.c:
	  effecttv: fix reverse negotiation
	  The plugins were using _fixed_caps_ and thus not adjusting to new upstream
	  sizes. Spotted by Tim Müller.

2011-11-25 11:43:16 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstwarp.c:
	  warptv: remove not needed ifdef

2011-11-25 10:15:35 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstripple.c:
	  rippletv: clean up the rendering code a bit
	  This is corrrupts the memoy when resizing. Add a FIXME to make it resizeable
	  once that is solved.

2011-11-24 21:41:03 +0100  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/alphacolor.c:
	* tests/check/elements/audioamplify.c:
	* tests/check/elements/audiochebband.c:
	* tests/check/elements/audiocheblimit.c:
	* tests/check/elements/audiodynamic.c:
	* tests/check/elements/audioecho.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/audiopanorama.c:
	* tests/check/elements/audiowsincband.c:
	* tests/check/elements/audiowsinclimit.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/avisubtitle.c:
	* tests/check/elements/capssetter.c:
	* tests/check/elements/cmmldec.c:
	* tests/check/elements/cmmlenc.c:
	* tests/check/elements/equalizer.c:
	* tests/check/elements/icydemux.c:
	* tests/check/elements/jpegenc.c:
	* tests/check/elements/level.c:
	* tests/check/elements/parser.c:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rglimiter.c:
	* tests/check/elements/rgvolume.c:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/check/elements/spectrum.c:
	* tests/check/elements/videofilter.c:
	* tests/check/elements/y4menc.c:
	  tests: update for gstcheck API change

2011-11-24 20:42:49 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstquark.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	  effecttv: fix reverse negotiation
	  The plugins were using _fixed_caps_ and thus not adjusting to new upstream
	  sizes. Spotted by Tim Müller.

2011-11-24 14:14:53 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: Fix leak of filename strings
	  Do not forget to free the filename strings when deleting
	  the list of files.

2011-11-24 14:11:33 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* tests/check/elements/multifile.c:
	  multifile: fix build of tests
	  Tests fail to build because g_mkdtemp is available from glib since
	  2.26.
	  This patch adds a condition around the redefinition of
	  g_mkdtemp on the tests to only build it if glib is older than
	  2.26.

2011-09-27 16:49:45 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: skip id32 tags
	  This allows decoding at least one sample where something has
	  stuffed some ID3 tag before the (supposedly initial) FMT\ .
	  https://bugzilla.gnome.org/show_bug.cgi?id=660249

2011-10-31 17:06:18 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/effectv/gstedge.c:
	  edgetv: trivial comment fix for clarity
	  https://bugzilla.gnome.org/show_bug.cgi?id=661841

2011-10-31 17:04:23 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/effectv/gstedge.c:
	  edgetv: don't leave bits of the output buffer uninitialized
	  Let's initialize them to zero. It looks alright, but then it
	  also looks alright with v3, or with the corresponding pixels
	  from the source. I don't know what the original intent would
	  be, and the original effectv source also has this bug/feature.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661841

2011-11-24 10:25:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparse: Use the sinkpad template caps as fallback, not the srcpad ones

2011-11-24 09:59:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:57:57 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:55:47 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:53:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	  amrparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:49:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	  amrparse: Mark some more functions as static

2011-11-24 09:48:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:44:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Mark some functions as static and remove unused function declarations

2011-11-24 09:43:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 01:48:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/souphttpsrc.c:
	  tests: update soup test for removed iradio-mode property

2011-11-24 01:45:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: get rid of iradio-* properties, post tags instead

2011-11-24 01:40:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: always send icecast request header, drop iradio-mode property
	  Server should ignore unknown/unhandled headers..

2011-11-24 01:19:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: make connection-speed property a guint64

2011-11-24 00:52:40 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpmanager.c:
	* tests/check/elements/rtpbin.c:
	* tests/examples/rtp/client-PCMA.c:
	* tests/examples/rtp/client-PCMA.py:
	* tests/examples/rtp/server-alsasrc-PCMA.c:
	* tests/examples/rtp/server-alsasrc-PCMA.py:
	  rtpmanager: rename gstrtp* -> rtp*
	  This was done in 0.10 to avoid conflict with the rtp elements in
	  farsight, but the gst-prefixing is no longer needed in 0.11

2011-11-23 23:29:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/twolame/gsttwolamemp2enc.c:
	  ext: fix more printf format warnings in debug messages

2011-11-23 23:29:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  ext: fix more printf format warnings in debug messages

2011-11-23 10:23:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-11-23 09:26:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: avoid endless caps loop
	  Check if the caps are the same before adding a new probe. Because of reconfigure
	  events, upstreams sends multiple caps events.

2011-11-23 00:57:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/matroskaparse.c:
	* tests/files/pinknoise-vorbis.mkv:
	  tests: add basic unit test for matroskaparse

2011-11-23 00:56:26 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: don't leak stream headers
	  https://bugzilla.gnome.org/show_bug.cgi?id=664548

2011-11-22 01:40:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/ximage/gstximagesrc.c:
	  More printf format warning fixes

2011-11-21 20:31:31 +0100  Matej Knopp <matej.knopp@gmail.com>

	* configure.ac:
	* gst/alpha/gstalpha.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/auparse/gstauparse.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/flv/gstflvdemux.c:
	* gst/goom/gstgoom.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/spectrum/gstspectrum.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* sys/ximage/gstximagesrc.c:
	  Fix printf format compiler warnings on OS X / 64bit
	  https://bugzilla.gnome.org/show_bug.cgi?id=662615

2011-11-21 13:37:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/wavparse/gstwavparse.c:
	  update for activation changes

2011-11-18 17:59:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/wavparse/gstwavparse.c:
	  update for new scheduling query

2011-11-18 13:57:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/wavparse/gstwavparse.c:
	  add parent to activate functions

2011-11-17 17:36:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: activate pad before setting caps
	  Seting caps on an inactive flushing pad does nothing.

2011-11-17 17:17:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/speex/gstspeexenc.c
	  gst/rtpmanager/rtpsession.c

2011-11-17 15:02:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/flac/gstflactag.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/pulse/pulseaudiosink.c:
	* gst/auparse/gstauparse.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/goom/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* gst/y4m/gsty4mencode.c:
	  add parent to pad functions

2011-11-17 08:24:58 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/cairo/gsttextoverlay.c:
	* gst/avi/gstavimux.c:
	* gst/flv/gstflvmux.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstqtmux.c:
	* gst/matroska/matroska-mux.c:
	* gst/multipart/multipartmux.c:
	* gst/smpte/gstsmpte.c:
	* gst/videomixer/videomixer.c:
	  collectpads: port API changes

2011-11-16 19:08:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: ensure to free allocated padded data

2011-11-16 18:57:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: reset tag setter interface when appropriate

2011-11-16 18:57:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: reset tag setter interface when appropriate

2011-11-16 17:54:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	  add parent to internal links

2011-11-16 17:27:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/pulse/pulseaudiosink.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/auparse/gstauparse.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/flv/gstflvdemux.c:
	* gst/goom/gstgoom.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/wavparse/gstwavparse.c:
	  add parent to query function

2011-11-16 12:40:08 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: update for renamed flags
	  Use the _check_reconfigure method instead of checking flags.
	  Don't need to ref the parent anymore, core does that.

2011-11-15 18:01:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/auparse/gstauparse.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/progressreport.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/wavparse/gstwavparse.c:
	  _query_peer_*() -> _peer_query_*()

2011-11-15 17:45:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  _accept_caps() -> _query_accept_caps()

2011-11-15 17:29:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesrc.c:
	* gst/goom/gstgoom.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/shapewipe/gstshapewipe.c:
	* sys/v4l2/gstv4l2src.c:
	  _peer_get_caps() -> _peer_query_caps()

2011-11-15 16:55:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* sys/v4l2/gstv4l2src.c:
	* tests/icles/gdkpixbufsink-test.c:
	  update for _get_caps() -> _query_caps()

2011-11-15 16:31:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/videocrop/gstaspectratiocrop.c:
	  change getcaps to query
	  Chain up event function in payloaders.

2011-11-15 13:23:56 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix spurious timestamp discontinuity
	  We need to tell the base class that we're dropping buffers,
	  so it drops the input timestamps corresponding to these.
	  Otherwise, the first actual audio buffers we output will be
	  stamped with those - GST_CLOCK_TIMESTAMP_NONE. That mismatch
	  between input buffer count and output buffer count will stay
	  while playing. With enough headers and long enough buffer
	  durations, the sink will have played enough before receiving
	  the first valid timestamp (usually 0), and will trigger an
	  audible discontinuity.

2011-11-14 15:34:57 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: detect when a file lies about fixed block size
	  If the sample/block number happens to be the same as the block
	  size, we assume variable block size, and thus counters in samples
	  in the headers. This can only get us a false positive for a block
	  size of 1, which is invalid. We can get false negatives more
	  often though (eg, if not starting at the start of the stream),
	  but then that's already GIGO.

2011-09-02 19:20:07 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: Add special mode to use FIR as repair as Google does
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-09-01 17:47:38 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: Send FIR requests in response to key unit requests with all-headers=TRUE
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-09-01 16:25:21 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: Put the PLI requests in each RTPSource
	  Also refactor a bit and put all the keyframe request code in one
	  place inside rtpsession.c
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-08-31 14:35:33 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Hack to FIR because Google doesn't set the sender ssrc correctly
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-08-30 19:06:13 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Process received Full Intra Requests
	  Process FIR requests according to RFC 5104
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-11-07 18:43:26 +0000  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Set pixel-aspect-ratio to 1/1
	  We don't currently support setting the pixel-aspect-ratio from V4L2. So
	  simply set it to be 1/1 in the caps to prevent negotiation failures when
	  fixating to weird values (e.g. when the downstream caps has
	  pixel-aspect-ratio = [ MIN, MAX ] )
	  https://bugzilla.gnome.org/show_bug.cgi?id=663580

2011-11-14 09:39:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/id3demux.c:
	  tests: make id3demux test compile
	  Still fails though.

2011-11-12 15:42:27 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	  controller: no need to explicitely add controlled properties anymore

2011-11-13 23:42:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2src.c:
	  Update for GstURIHandler get_protocols() changes

2011-11-13 18:50:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2src.c:
	  soup, pushfile, rtsp, udp, v4l2: update for GstURIHandler API changes

2011-11-11 19:24:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/pulse/pulseaudiosink.c

2011-11-11 19:21:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtp: fix for rtp header changes

2011-11-11 10:06:25 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: fix caps leak

2011-11-11 14:55:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: do not leak clientname when setting up property

2011-11-11 18:05:35 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulse: Chain up dispose() in pulseaudiosink

2011-11-11 12:32:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/gstrtpxqtdepay.h:
	* gst/rtp/fnv1hash.h:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpac3depay.h:
	* gst/rtp/gstrtpac3pay.h:
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpbvdepay.h:
	* gst/rtp/gstrtpbvpay.h:
	* gst/rtp/gstrtpceltdepay.h:
	* gst/rtp/gstrtpceltpay.h:
	* gst/rtp/gstrtpdvdepay.h:
	* gst/rtp/gstrtpdvpay.h:
	* gst/rtp/gstrtpg722depay.h:
	* gst/rtp/gstrtpg722pay.h:
	* gst/rtp/gstrtpg723depay.h:
	* gst/rtp/gstrtpg723pay.h:
	* gst/rtp/gstrtpg726depay.h:
	* gst/rtp/gstrtpg726pay.h:
	* gst/rtp/gstrtpg729depay.h:
	* gst/rtp/gstrtpg729pay.h:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtpgstdepay.h:
	* gst/rtp/gstrtpgstpay.h:
	* gst/rtp/gstrtph263depay.h:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.h:
	* gst/rtp/gstrtpilbcdepay.h:
	* gst/rtp/gstrtpilbcpay.h:
	* gst/rtp/gstrtpj2kdepay.h:
	* gst/rtp/gstrtpj2kpay.h:
	* gst/rtp/gstrtpjpegdepay.h:
	* gst/rtp/gstrtpjpegpay.h:
	* gst/rtp/gstrtpmp1sdepay.h:
	* gst/rtp/gstrtpmp2tdepay.h:
	* gst/rtp/gstrtpmp2tpay.h:
	* gst/rtp/gstrtpmp4adepay.h:
	* gst/rtp/gstrtpmp4apay.h:
	* gst/rtp/gstrtpmp4gdepay.h:
	* gst/rtp/gstrtpmp4gpay.h:
	* gst/rtp/gstrtpmp4vdepay.h:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadepay.h:
	* gst/rtp/gstrtpmpapay.h:
	* gst/rtp/gstrtpmparobustdepay.h:
	* gst/rtp/gstrtpmpvdepay.h:
	* gst/rtp/gstrtpmpvpay.h:
	* gst/rtp/gstrtppcmadepay.h:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmudepay.h:
	* gst/rtp/gstrtppcmupay.h:
	* gst/rtp/gstrtpqcelpdepay.h:
	* gst/rtp/gstrtpqdmdepay.h:
	* gst/rtp/gstrtpsirendepay.h:
	* gst/rtp/gstrtpsirenpay.h:
	* gst/rtp/gstrtpspeexdepay.h:
	* gst/rtp/gstrtpspeexpay.h:
	* gst/rtp/gstrtpsv3vdepay.h:
	* gst/rtp/gstrtptheoradepay.h:
	* gst/rtp/gstrtptheorapay.h:
	* gst/rtp/gstrtpvorbisdepay.h:
	* gst/rtp/gstrtpvorbispay.h:
	* gst/rtp/gstrtpvrawdepay.h:
	* gst/rtp/gstrtpvrawpay.h:
	  update for base class rename

2011-11-11 12:25:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/gstrtpxqtdepay.h:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3depay.h:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpac3pay.h:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvdepay.h:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpbvpay.h:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltdepay.h:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpceltpay.h:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvdepay.h:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpdvpay.h:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722depay.h:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg722pay.h:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723depay.h:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg723pay.h:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726depay.h:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg726pay.h:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729depay.h:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpg729pay.h:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstdepay.h:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263depay.h:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcdepay.h:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpilbcpay.h:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kdepay.h:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpj2kpay.h:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegdepay.h:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpjpegpay.h:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp1sdepay.h:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tdepay.h:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp2tpay.h:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4adepay.h:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4apay.h:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gdepay.h:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4gpay.h:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vdepay.h:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpadepay.h:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpapay.h:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmparobustdepay.h:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvdepay.h:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtpmpvpay.h:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmadepay.h:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmudepay.h:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtppcmupay.h:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqcelpdepay.h:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpqdmdepay.h:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirendepay.h:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpsirenpay.h:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexdepay.h:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpspeexpay.h:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtpsv3vdepay.h:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheoradepay.h:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtptheorapay.h:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbisdepay.h:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvorbispay.h:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawdepay.h:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtp/gstrtpvrawpay.h:
	  update for base class rename

2011-11-11 12:01:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/pulse/pulsesink.c:
	  update for audiobase* rename

2011-11-11 11:53:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	  audio: update for base class rename

2011-11-11 11:33:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseutil.h:
	* gst/equalizer/gstiirequalizer.h:
	  fix for ringbuffer rename

2011-11-11 11:24:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackringbuffer.h:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  update for ringbuffer change

2011-11-11 01:27:47 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: cosmetic error message change
	  LET'S TRY TO KEEP CAPITALS TO A MINIMUM.

2011-11-11 00:58:24 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/twolame/Makefile.am:
	* ext/twolame/gsttwolamemp2enc.c:
	* ext/twolame/gsttwolamemp2enc.h:
	  twolame: rename to twolamemp2enc

2011-11-11 00:51:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/twolame/gsttwolame.c:
	  twolame: port to 0.11

2011-11-10 23:15:30 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	  controller: port api changes

2011-11-10 23:09:23 +0200  Stefan Sauer <ensonic@users.sf.net>

	* ext/annodex/gstannodex.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audiopanorama.c:
	* gst/equalizer/gstiirequalizer.c:
	  various: add missing includes

2011-11-10 21:35:24 +0100  René Stadler <rene.stadler@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix compilation with pulseaudio 0.9

2011-11-10 18:32:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/flac/gstflactag.c:
	* gst/auparse/gstauparse.c:
	* gst/avi/gstavidemux.c:
	* gst/goom/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/multipart/multipartdemux.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/wavparse/gstwavparse.c:
	  update for adapter api changes

2011-11-10 17:23:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  update for changed base classes

2011-11-10 13:50:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  fix for audio clock change

2011-11-10 11:03:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/pulse/pulsesrc.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/ximage/gstximagesrc.c:
	  update for removed fixate function

2011-11-09 17:40:10 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-11-09 17:38:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	  updates for new acceptcaps query

2011-11-08 15:35:26 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix wrong stride when inverting uncompressed video
	  Such frames have a stride multiple of 4, see
	  http://lscube.org/pipermail/ffmpeg-issues/2010-April/010247.html.
	  This showed up on a sample using a odd width of 24 bit video.
	  https://bugzilla.gnome.org/show_bug.cgi?id=652288

2011-11-09 12:25:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  h263ppay: report to 0.11

2011-11-09 12:18:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/flac/gstflacdec.c
	  gst/audioparsers/gstflacparse.c
	  gst/isomp4/qtdemux.c

2011-11-09 11:56:07 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: fix compiler warning for uninitialized values

2011-11-09 11:53:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/wavparse/gstwavparse.c:
	  remove query types

2011-11-09 10:32:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: minimal sanity check on creation datetime

2011-11-04 17:54:04 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  dtmfsrc: Reject start/stop requests that come out of order

2011-10-29 18:24:26 +0200  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: Post messages when starting to send/receive DTMF
	  This way, the UI can display the DTMF events as they as being sent.

2011-11-02 12:58:12 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Return the sink pad template as sink caps, not the src's
	  https://bugzilla.gnome.org/show_bug.cgi?id=577784

2009-03-15 19:26:48 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Also implement size/framerate restrictions in getcaps
	  https://bugzilla.gnome.org/show_bug.cgi?id=577784

2009-03-04 20:50:19 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Implement getcaps following RFC 4629, picks the right annexes
	  https://bugzilla.gnome.org/show_bug.cgi?id=577784

2011-11-08 14:31:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: also set segment stop at startup rather than only post seek
	  ... so as to ensure consistent playback with or without seek, especially
	  in presence of some bogus edit list entries.

2011-11-08 11:18:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	* gst/rtsp/gstrtspsrc.c:
	  update for probe api changes

2011-11-08 08:50:19 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/goom/gstgoom.c:
	  goom: code cleanups
	  Move variables to the scope where they are needed. Use our macros and functions
	  more.

2011-11-08 08:49:05 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/goom/gstgoom.c:
	  goom: add a sink_query to eat allocation queries
	  We should not forward allocation queries for audio to the video sink.

2011-11-02 17:02:54 +0000  Raul Gutierrez Segales <rgs@collabora.co.uk>

	* gst/flv/Makefile.am:
	  gst/flv/: add amfdefs.h to noinst_HEADERS
	  https://bugzilla.gnome.org/show_bug.cgi?id=663334

2011-11-07 17:14:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	* gst/rtsp/gstrtspsrc.c:
	  fix for probe updates

2011-10-03 17:50:43 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: detect large pts gaps and resync
	  Should work on multiple gaps, but tested on only one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=631430

2011-08-22 10:40:45 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix off by one between granpos and last_stop

2011-10-07 19:41:35 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix last frame timestamp in fixed block size mode
	  The last block may have a different block size, so we should not
	  use it to scale or we'll end up with a wrong timestamp.
	  See comment and quote from the FLAC format documentation in the code.
	  Fixes looped playback of FLAC files (via about-to-finish).
	  https://bugzilla.gnome.org/show_bug.cgi?id=661215

2011-10-27 15:52:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttextoverlay.h:
	  cairotextoverlay: add a 'silent' property to skip rendering
	  https://bugzilla.gnome.org/show_bug.cgi?id=662856

2011-11-07 12:00:12 +0100  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroskamux: fix regression causing malformed files
	  This was caused by me in 1b213d. It seems I was too focused on 0.11 when I did
	  this and tested the wrong branch.
	  The problem was reported by Alexey Fisher.

2011-11-04 18:41:36 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/annodex/gstcmmldec.h:
	* gst/alpha/Makefile.am:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	* gst/audiofx/Makefile.am:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofx.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/effectv/Makefile.am:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstvertigo.c:
	* gst/equalizer/Makefile.am:
	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer.h:
	* gst/shapewipe/Makefile.am:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/Makefile.am:
	* gst/smpte/gstsmptealpha.c:
	* gst/videobox/Makefile.am:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/plugin.c:
	* gst/videomixer/Makefile.am:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer2.c:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* tests/examples/shapewipe/shapewipe-example.c:
	* tests/examples/v4l2/camctrl.c:
	  controller: port to new controller location and api

2011-11-04 18:52:35 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/gststereo.c:
	  controller: port to new controller location and api

2011-11-04 17:39:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  more template fixes

2011-11-04 16:21:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: more 0.11 fixing
	  Make sure the caps event gets to the sink.

2011-11-04 15:35:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: port some more
	  Rename decodebin2 -> decodebin some more
	  Cleanup up sinkpad event handling

2011-11-04 13:56:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: port some more to 0.11
	  We must not forward the caps event. instead we will decide what to do when the
	  pad block is taken.
	  Use decodebin instead of decodebin2

2011-11-04 13:12:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/interleave/deinterleave.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	  more template fixes

2011-11-04 11:58:22 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstqtmux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/webm-mux.c:
	* gst/multipart/multipartmux.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/videomixer/videomixer.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/interleave.c:
	* tests/check/elements/matroskamux.c:
	* tests/check/elements/qtmux.c:
	* tests/check/elements/rtpbin.c:
	  make %u in all request pad templates

2011-11-04 11:01:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/rtp/gstrtpvrawdepay.c

2011-11-04 10:32:46 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* configure.ac:
	* gst/apetag/gstapedemux.c:
	  Port apedemux

2011-11-03 23:28:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtp: use GLib's G_BIG_ENDIAN define instead of BIG_ENDIAN
	  Fixes compiler warning on mingw32

2011-11-03 16:43:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* common:
	* configure.ac:
	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	* gst/udp/Makefile.am:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstudp.c:
	* gst/udp/gstudpsrc.c:
	  update for new net library

2011-11-02 12:09:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* ext/flac/gstflactag.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/gsttaginject.c:
	* gst/flv/gstflvdemux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/wavparse/gstwavparse.c:
	  tags: update for tag API removal

2011-11-02 10:40:12 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-10-31 02:40:08 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstudpsrc.c:
	  update for netbuffer api change

2011-10-31 02:35:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstudp.c:
	* gst/udp/gstudpsrc.c:
	  update for netaddress change

2011-10-31 02:24:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawdepay.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  update for meta api change

2011-10-29 09:29:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/gstqtmoovrecover.c:
	* gst/rtsp/gstrtspsrc.c:
	  update for new task api

2011-10-29 09:09:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtsp/gstrtspsrc.c:
	* sys/v4l2/gstv4l2object.c:
	  structure: fix for api update

2011-10-29 08:25:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  bufferlist: update for new API

2011-11-01 00:40:40 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	* gst/rtsp/gstrtspsrc.c:
	  Update for pad API changes
	  GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*

2011-10-31 18:38:55 +0100  René Stadler <rene.stadler@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: fix obvious crash

2011-10-31 16:18:32 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: avoid shortcut evaluation when adding paired mp4 tag
	  Fixes (part of) #638711.

2011-10-31 15:43:25 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: do not use unoffical V_MJPEG codec id
	  ... but as not spec'ed especially, consider it a VfW compatibility case.
	  Fixes #659837.

2011-10-30 19:30:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.h:
	  flacenc: remove dead code from header
	  We require a new-enough libflac that this condition will never apply.

2011-10-30 19:09:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: parse stream headers from caps in set_format function
	  Not that this seems to be actually needed, libflac happily decodes
	  stuff even if we just drop all headers and never feed it to the
	  library.

2011-10-30 18:49:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: don't extract metadata, leave that to the parser or container

2011-10-30 18:45:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: we expect framed input now, remove some more code

2011-10-09 16:18:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: naive port to GstAudioDecoder
	  This would probably have been too invasive to do in the 0.10
	  branch, with all the pull-mode and parser handling code in
	  there.

2011-10-30 12:29:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/Makefile.am:
	* ext/lame/README:
	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	* ext/lame/plugin.c:
	* ext/lame/test-lame.c:
	* tests/check/pipelines/lame.c:
	  lame: remove lame element, it's been superseded by lamemp3enc

2011-10-30 11:51:58 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  ext, gst: update for taglist API changes

2011-10-30 11:44:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/debugutils/gsttaginject.c:
	* gst/flv/gstflvdemux.c:
	* gst/icydemux/gsticydemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/multipart/multipartdemux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/wavparse/gstwavparse.c:
	  ext, gst: update for taglist API changes

2011-10-30 11:41:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	  tests: fix compilation of audio tests in uninstalled setup

2011-10-28 21:26:33 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: simplify get_unit_size

2011-10-28 21:19:42 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/audioecho.c:
	  tests: audioecho: port to 0.11

2011-10-28 21:18:33 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/audiofx/audioecho.c:
	  audioecho: fix internal buffer size calculation

2011-10-28 14:05:48 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/audiochebband.c:
	  tests: audiochebband: port to 0.11

2011-10-28 16:52:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-10-28 15:08:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: fix porting errors
	  The probes were ported wrongly and caused deadlocks.

2011-10-28 09:57:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg
	  jpegdec (using libjpeg 6.2/8) can't decode some lossless types of JPEG.
	  https://bugzilla.gnome.org/show_bug.cgi?id=556648

2011-10-28 13:06:20 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/audiocheblimit.c:
	  tests: audiocheblimit: port to 0.11

2011-10-28 13:02:56 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/audiofx/audiofxbaseiirfilter.c:
	  audiofx: fix crash in process()

2011-10-28 11:48:31 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/audioamplify.c:
	  tests: audioamplify: port to 0.11

2011-10-28 12:51:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulse: fix check for empty caps

2011-10-28 12:30:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: elaborate some debug statements

2011-10-11 20:56:51 +0400  Stas Sergeev <stsp@users.sourceforge.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: be careful with negative cts
	  Fixes #661477.

2011-10-06 13:04:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: tune non-update seek handling cases
	  Fixes #661049.

2011-10-28 11:46:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/videomixer/gstcollectpads2.c

2011-10-28 11:16:38 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/audiofx/audiodynamic.c:
	  audiodynamic: don't set process function too early
	  GstAudioInfo and GstAudioFilter have been changed so that this code doesn't
	  crash anymore when a property is set in NULL state.

2011-10-28 10:42:04 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/audiodynamic.c:
	  tests: audiodynamic: port to 0.11

2011-10-28 00:24:14 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/spectrum.c:
	  tests: spectrum: port to 0.11

2011-10-27 23:57:17 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* tests/check/elements/audiopanorama.c:
	  tests: audiopanorama: port to 0.11

2011-10-27 23:56:12 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: fix get_unit_size

2011-10-28 10:40:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Use the clip function instead of the prepare_buffer function

2011-10-28 09:05:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* sys/v4l2/gstv4l2object.c:
	  rtpmanager, v4l2: fix compiler warnings after gst_caps_new_simple() change

2011-10-28 09:01:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix compiler warnings after gst_caps_new_simple() change

2011-10-28 09:36:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/gstcollectpads2.c:
	* gst/videomixer/gstcollectpads2.h:
	* gst/videomixer/videomixer2.h:
	* gst/videomixer/videomixer2pad.h:
	  videomixer2: Use collectpads2 from core

2011-10-27 19:39:20 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/wavenc/Makefile.am:
	* gst/wavenc/gstwavenc.c:
	  wavenc: port to 0.11 raw audio caps

2011-10-27 19:06:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/flv/gstflvmux.c

2011-10-27 19:00:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/icydemux/gsticydemux.c:
	* gst/rtp/README:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	  make some more things compile again

2011-10-27 16:08:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/pulse/pulseaudiosink.c
	  ext/pulse/pulsesink.c

2011-10-27 16:03:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* sys/v4l2/gstv4l2object.c:
	  fix compilation

2011-10-28 00:41:45 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Don't pointlessly hold object lock over caps operations
	  Avoids a deadlock when getcaps is recursive due to the getcaps being
	  reflected upstream/downstream. The lock isn't actually protecting
	  anything here.

2011-10-27 00:37:03 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/flv/amfdefs.h:
	* gst/flv/gstflvmux.c:
	  flvmux: add some comments and defines to clarify code.

2011-10-10 15:36:14 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroska: refactor ebml-write to be more 0.11 friendly
	  Switching to a more 0.11-friendly pattern, where getting the buffer's data
	  pointer and setting the size many times is less natural. This is of course in
	  preparation to the upcoming port of the plugin.

2011-10-11 21:45:46 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroska: remove stale floatcast include
	  GDOUBLE_TO_BE was moved to core a long time ago.

2011-10-11 22:10:27 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix possible crash with malformed dirac codec_data
	  Since size is unsigned, we need to safeguard against wrapping below zero.

2011-10-21 22:33:34 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: remove avoidable call to gst_object_set_name

2011-10-21 22:32:38 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: remove avoidable call to gst_object_set_name

2011-10-21 14:51:23 +0200  Stefan Sauer <ensonic@users.sf.net>

	* ext/pulse/pulsemixerctrl.h:
	* gst/videofilter/gstvideobalance.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstossmixer.h:
	* sys/oss4/oss4-mixer.c:
	* sys/oss4/oss4-source.c:
	* sys/osxaudio/gstosxaudioelement.c:
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/gstv4l2videooverlay.c:
	* sys/v4l2/gstv4l2videooverlay.h:
	* sys/v4l2/gstv4l2vidorient.c:
	* sys/v4l2/gstv4l2vidorient.h:
	  interfaces: clean up the use of iface and class/klass

2011-10-21 11:37:05 +0100  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>

	* gst-plugins-good.spec.in:
	  Update spec file so its paralel-installable and only tries to package ported plugins

2011-10-16 20:30:25 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/gstpngenc.c:
	  pngenc: increase arbitrary resolution limits
	  Apparently libpng can technically do up to 2^31-1 rows and columns. However it
	  imposes an (arbitrary) default limit of 1 million (that could theoretically be
	  lifted by using some additional API).
	  Moved array allocation to the heap now.

2011-10-16 20:25:41 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/gstpngenc.c:
	  pngenc: don't unconditionally allocate 4096 pointers on the stack
	  Instead allocate as many as needed (on the stack still).

2011-10-16 20:05:28 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/gstpngenc.c:
	  pngenc: ensure setcaps was called before chain function
	  This is needed to properly error out for e.g. "fakesrc ! pngenc ! fakesink".

2011-10-16 19:44:27 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/gstpngenc.c:
	  pngenc: validate input buffer size
	  Just for safety; of course such mismatch represents a bug in another element.

2011-10-16 19:41:28 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/Makefile.am:
	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	  pngenc: make setcaps more robust, use gstvideo functions
	  A setcaps function needs to actually verify the caps carefully. In this case,
	  it was possible to e.g. link a video decoder with YUV+RGB template caps to
	  pngenc.  That would cause a crash when the decoder pushes a YUV buffer. Same
	  thing when pushing a valid buffer that exceeds the resolution limits.
	  Also, missing framerate caps field would cause a glib critical warning due to
	  invalid GValue. This fails hard now.

2011-10-21 10:01:43 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  ebml: small correction to previous commit
	  Signal a short read with UNEXPECTED, exactly like the peek_bytes function.

2011-10-19 13:09:51 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  ebml: Fix push-based behaviour
	  The 'peek' method was completely wrong (!?)

2011-10-18 18:31:17 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulse: Get caps correctly on pad block
	  Instead of always going upstream, we should first see if already got
	  caps from a setcaps() call.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-18 12:25:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/wavpack/gstwavpackenc.c:
	  wavpackenc: don't unref buffer with gst_object_unref()

2011-10-18 12:05:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: only use is_pcm for 1.0 of pulseaudio

2011-10-18 11:58:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: only disable trickmodes for !pcm
	  Only disable trickmodes when we are not dealing with raw PCM samples.

2011-10-16 15:32:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videocrop/gstvideocrop.c:
	  videocrop: fix compilation

2011-10-16 15:26:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/rtp/gstrtpvrawdepay.c

2011-10-14 10:56:16 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Fix a leak
	  Buffers weren't being unref'ed in one case inside, causing memory usage
	  to blow up.

2011-10-14 09:10:01 +0200  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  set colour masks for video/x-raw-rgb in rtpvrawdepay

2011-10-13 01:05:13 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* configure.ac:
	  configure: re-enable videocrop plugin
	  Already ported to 0.11

2011-10-13 01:05:04 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstaspectratiocrop.h:
	  aspectratiocrop: Port to 0.11

2011-10-13 00:39:28 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/videocrop/Makefile.am:
	* gst/videocrop/gstvideocrop.c:
	* gst/videocrop/gstvideocrop.h:
	  videocrop: Port to 0.11

2011-10-12 17:43:47 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* tests/check/elements/aspectratiocrop.c:
	  tests: aspectratiocrop: Port to 0.11

2011-10-12 08:24:28 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* tests/check/elements/alphacolor.c:
	  tests: alphacolor: Port to 0.11

2011-10-13 17:12:23 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: Properly register type
	  It's a subclass of GstAudioEncoder and not of GstElement

2011-10-13 16:59:50 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Fix incorrect gst_buffer_replace() call
	  This got exposed when gst_buffer_replace() was changed from a macro to a
	  function.

2011-10-13 09:34:04 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Fix wrong usage of gst_iterator_filter
	  It takes a GValue* as the user_data.
	  And don't forget to unref the demuxer before returning.

2011-10-13 09:02:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  fix compile

2011-10-13 08:58:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/jpeg/gstjpegdec.c
	  gst/rtp/gstrtpvrawpay.c

2011-10-12 08:09:20 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* tests/check/elements/cmmlenc.c:
	  tests: cmmlenc: Port to 0.11

2011-10-12 08:02:08 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* tests/check/elements/cmmldec.c:
	  tests: cmmldec: Port to 0.11

2011-10-12 07:29:30 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: Use new GstIterator API correctly
	  GstIterator now uses GValue, use it correctly.

2011-10-12 11:26:50 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: Only use 24 LSB for depth=24 RGB caps
	  ... and indent the masks for clarity

2011-10-11 14:58:43 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix segment handling, so we actually use running time
	  gst_matroska_mux_best_pad adjusts the buffer timestamp to running time using
	  the segment stored in the pad's collect data. However, the event handler didn't
	  pass the newsegment event on to collectpads' handler, so this segment was never
	  updated at all.
	  Re-fixes bug #432612.

2011-10-10 19:01:23 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/rtp/gstrtpg722pay.c:
	  gstrtpg722pay: Compensate for clockrate vs. samplerate difference
	  The RTP clock-rate used for G722 is 8000, even though the samplerate is
	  16000. Compensate for this by pretending G722 has 8 bits per sample
	  instead of the 4 bits as if it were a codec that ran at half the speed,
	  but with twice the number of bits. Fixes #661376

2011-09-27 19:25:53 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Implement upstream negotiation
	  Add upstream negotiation for jpegdec. Fixes #660275

2011-10-10 19:02:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: don't leak audio codec_data buffer

2011-10-10 17:41:10 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	  alpha: Don't use start() vmethod
	  The only thing we're doing is initializing parameters ...
	  * which won't work because we don't have upstream/downstream caps
	  * which will be initialized when ::set_caps() is called

2011-10-10 14:08:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-10-10 13:22:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/id3demux/gstid3demux.c:
	  id3demux: port to 0.11

2011-10-10 13:20:04 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/cairo/Makefile.am:
	  tests: add missing PLUGIN_ASE_LIBS to LDADD

2011-10-10 12:54:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/icydemux/gsticydemux.c:
	  icydemux: port to 0.11

2011-10-10 12:27:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	  annodex: port to 0.11

2011-10-10 11:48:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/speex/gstspeexenc.c

2011-10-10 00:18:56 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulse: port pulseutil to 0.11

2011-10-09 21:17:24 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: port to 0.11

2011-10-09 18:58:29 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Fixing getcaps function
	  Update getcaps function to 0.11 API

2011-10-09 21:31:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	* ext/speex/gstspeexenc.h:
	  speexenc: only push header buffers following initial events

2011-10-09 16:29:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11

2011-10-09 16:24:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux: update for __gst_debug_min name change

2011-10-09 11:18:18 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/atomsrecovery.c:
	  qtmux: Fix memory leak on atoms recovery function
	  Remember to free the ftyp data after writing it to a file.
	  Fixes #660969

2011-10-06 12:26:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: report new bits

2011-10-06 12:23:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/speex/gstspeexdec.c
	  ext/speex/gstspeexenc.c
	  gst/isomp4/atoms.c
	  gst/isomp4/gstqtmux.c

2011-09-21 18:45:42 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: improve segment handling with non-zero starting timestamp
	  ... as well as related items, such as seeking and position reporting.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659808

2011-09-29 18:41:53 +0400  Stas Sergeev <stsp@users.sourceforge.net>

	* sys/v4l2/gstv4l2object.c:
	* sys/ximage/gstximagesrc.c:
	  v4l2, ximagesrc: fix some printf format compiler warnings
	  https://bugzilla.gnome.org/show_bug.cgi?id=660150

2011-09-30 12:42:22 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: Refactor bitrate check test
	  Refactor bitrate check test to accomodate multiple tests
	  for bitrate

2011-09-30 13:02:31 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/atoms.c:
	  qtmux: update esds atom under wave atom for aac bitrates
	  AAC in mov format puts an ESDS atom inside of a WAVE atom in
	  STSD atom, we need to update the bitrate on this ESDS. This patch
	  fixes it.

2011-09-30 12:41:52 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/atoms.c:
	* gst/isomp4/fourcc.h:
	  qtmux: Also update btrt atom
	  When rewriting bitrates, also update the btrt atom under stsd

2011-09-30 10:55:53 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: add tests for bitrate average calculation
	  Adds tests to make sure qtmux/mp4mux sets average bitrate
	  correctly

2011-09-28 11:41:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Calculate average bitrate for streams
	  Calculate and use average bitrate for streams when no
	  bitrate tag was received

2011-09-28 10:41:14 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Avoid a buffer metadata copy if possible
	  If first_ts is 0 there is no need to subtract, so we might
	  skip some copying to make the buffer metadata writable.

2011-09-29 23:21:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: initialise variable before adding to it

2011-09-29 17:21:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexdec.h:
	  speexdec: port to audiodecoder

2011-09-29 16:33:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.h:
	  speexenc: clean up some unused remnants

2011-09-29 17:32:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/Makefile.am:
	* ext/speex/gstspeexenc.c:
	* ext/speex/gstspeexenc.h:
	  speexenc: port to audioencoder

2011-09-28 19:10:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: get rid of granulepos handling
	  Leave that to the parser or demuxer. There's still some
	  code for operating in DEFAULT (samples) format, but that
	  will be removed later.

2011-09-28 18:32:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: get rid of pull-mode support and focus on being a decoder
	  Leave all the other stuff to flacparse.

2011-09-28 17:29:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflactag.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	  flac, jpeg: fix compiler warning

2011-09-28 17:40:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflactag.c:
	  flac: port to 0.11

2011-09-28 17:39:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/flac/gstflacenc.c

2011-09-28 16:18:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-09-28 16:09:58 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/Makefile.am:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flacenc: port to audioencoder

2011-09-27 15:59:24 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-parse.c:
	  matroskademux: ensure minimal alignment for audio/x-raw-* buffers
	  Since matroskademux will attempt to push unaligned buffers,
	  downstream might have trouble with those, especially if downstream
	  uses ORC, such as audioconvert.
	  Ensure we push buffers aligned to the basic type at least for
	  those raw buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659798

2011-09-28 12:44:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  common
	  ext/pulse/pulsesink.c
	  ext/soup/gstsouphttpclientsink.c
	  gst/audioparsers/gstaacparse.c
	  gst/audioparsers/gstac3parse.c
	  gst/rtp/gstrtph264depay.c
	  gst/rtpmanager/gstrtpjitterbuffer.c
	  gst/rtpmanager/rtpjitterbuffer.c
	  gst/rtsp/gstrtspsrc.c
	  sys/ximage/gstximagesrc.c

2011-09-28 00:10:09 +0300  Raimo Järvi <raimo.jarvi@gmail.com>

	* gst/goom2k1/goom_core.c:
	  goom2k1: Fix compiler warnings on 64 bit mingw-w64
	  Fixes bug #660294.

2011-09-27 18:19:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  lame: fix raw audio caps too

2011-09-27 18:15:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  lame: port to 0.11

2011-09-26 16:29:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/twolame/gsttwolame.c:
	  twolame: Simple fix for GstAudioEncoder API change

2011-09-26 16:28:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/twolame/gsttwolame.c:
	  twolame: Fix variable 'gstelement_class' set but not used compiler warning

2011-09-26 16:08:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  lame: Don't get the parent class again, GST_BOILERPLATE does this already

2011-09-26 16:07:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  lame: Fix variable 'gstelement_class' set but not used compiler warning

2011-09-26 12:07:15 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/twolame/gsttwolame.c:
	  twolame: improve output framing and timestamping
	  ... which simply comes down to requesting one frame of input data at a time,
	  since the encoder nicely turns this into 1 encoded frame.

2011-09-26 11:56:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/twolame/Makefile.am:
	* ext/twolame/gsttwolame.c:
	* ext/twolame/gsttwolame.h:
	  twolame: port to audioencoder

2011-09-23 15:32:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/gstlame.c:
	  lame: use some more boilerplate

2011-09-23 15:26:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  lame: port to audioencoder

2011-09-23 14:33:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: use some more boilerplate

2011-09-26 14:44:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: really report bitrate rather kbitrate

2011-09-26 14:44:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/lame/Makefile.am:
	* ext/lame/gstlamemp3enc.c:
	* ext/lame/gstlamemp3enc.h:
	  lamemp3enc: port to audioencoder

2011-09-25 15:13:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/Makefile.am:
	* ext/soup/gstsoup.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpclientsink.h:
	  soup: rename souphttpsink to souphttpclientsink
	  To avoid confusion, and because we might want a server
	  sink at some point too.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659947

2011-09-23 16:39:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsink.c:
	* ext/soup/gstsouphttpsink.h:
	  souphttpsink: don't create unused second sink pad object
	  The base class will create the sink pad.

2011-09-23 15:36:36 +0200  Julien Isorce <julien.isorce@gmail.com>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: correctly check for ac3/e-ac3 switch
	  https://bugzilla.gnome.org/show_bug.cgi?id=659943

2011-09-21 14:01:20 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Update common to 0.11 branch

2011-09-20 13:38:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: improve downstream flow return feedback to upstream
	  ... although basertpdepay does not really make it easy/possible to do so
	  all the way.

2011-09-20 12:11:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	  ximagesrc: add xid and xname properties to allow capturing a particular window
	  A particular window may be selected using the new xid (X-Window
	  XID, eg a pointer) and xname (window title) properties. If both
	  are specified, the XID is used in preference, falling back to
	  xname if not found.
	  Default (if none of xid and xname are specified, or if no such
	  window is found) is to capture the root window.
	  https://bugzilla.gnome.org/show_bug.cgi?id=546932

2011-08-02 17:39:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: add unit test to make sure encodebin picks mp4mux for variant=iso
	  https://bugzilla.gnome.org/show_bug.cgi?id=651496

2011-09-19 12:15:11 +0200  Ha Nguyen <hanguytv@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Fix a leaked clock for each buffering message
	  Fixes bug #659237.

2011-09-19 12:11:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: parse embedded ID32 tags

2011-09-02 13:41:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpsession: avoid source premature timing out
	  Use slightly adjusted sender interval to determine sender timeout rather than
	  our own sender side interval (which may have been forced small).

2011-08-25 12:40:52 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: avoid timing out source too quickly
	  ... following a PAUSE/PLAY cycle, particularly applicable when operating
	  with a short RTCP interval (possibly forced so server-side).

2011-08-24 14:37:52 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer/rtpbin: relax dropping rtcp packets
	  ... to at least having it trigger a/v synchronization, possibly without
	  using provided values which are still not considered sane
	  (as previously dropped).

2011-08-24 14:34:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: some more reset when clearing pt map
	  ... which in particular caters for some more reset following a possible
	  rtsp PLAY.

2011-08-21 21:58:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: do not set elements to PLAYING when doing seek in PAUSED

2011-09-01 14:47:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: only reset skew on gap if input ts available

2011-08-18 14:12:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: check some more for possible rtp timestamp discontinuity
	  ... when operating in non slave mode, and reset if detected.
	  This should avoid some (large) bogus outgoing timestamp due to jumps
	  in rtp time, as result of PAUSE/PLAY or seek or ...

2011-08-08 12:48:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: switch to rtp time based syncing when guessed appropriate

2011-08-08 12:15:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: alternative inter-stream syncing methods
	  ... at least if not syncing to NPT time:
	  * either sync using RTCP SR data (as currently)
	  * only perform the above once using initial RTCP SR packets
	  * discard RTCP and sync by equating provided stream's clock-base rtptime,
	  as provided by jitterbuffer (typically obtained from RTP-Info in RTSP).

2011-08-08 12:11:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: also provide clock-base to sync signal

2011-08-08 12:09:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: allow configurable rtcp stream syncing interval
	  ... rather than necessarily syncing at each RTCP SR.

2011-08-01 08:35:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: trigger reconsideration if rtcp interval set

2011-08-01 08:32:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: configure rtcp interval if provided
	  ... in PLAY response.

2011-09-16 16:53:22 +0300  Lasse Laukkanen <lasse.laukkanen@digia.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: Fix allowing zero duration tracks
	  https://bugzilla.gnome.org/show_bug.cgi?id=637486

2011-09-05 10:11:18 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/udp/gstudpnetutils.c:
	  udpsrc: error out when no protocol is specified in the uri
	  It is certainly better than to crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658178

2011-09-19 09:37:58 +0200  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: do not use invalid buffer timestamps

2011-03-29 12:09:18 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/Makefile.am:
	* ext/pulse/plugin.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.h:
	  pulse: New pulseaudiosink element to handle format changes
	  This introduces a new bin which wraps around pulsesink and depending on
	  the formats supported by the sink, plugs in/out a decodebin2 as
	  required. This allows users to switch sinks on the stream and adapts
	  accordingly (for example, you could watch a movie in passthrough mode on
	  your receiver which supports AC3 decode, then plug out and switch to a
	  non-digital profile to continue uninterrupted on analog output).
	  The bin is required because doing the same with playbin2/playsink will
	  require API changes that cannot be made in 0.10. With 0.11/1.0, we
	  should be able to ask for upstream caps renegotiation to deal with all
	  this.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657179

2011-09-16 15:03:23 +0200  Branko Subasic <branko@axis.com>

	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-read.h:
	* gst/matroska/matroska-read-common.c:
	  matroskademux: Avoid sending EOS when in paused state
	  Changed the ebml reader's gst_ebml_peek_id_length() function so
	  that it returns the actual reason for why the peek failed, instead
	  of (almost) always returning GST_FLOW_UNEXPECTED. This prevents
	  the pulling task from sending EOS when doing a flushing seek.

2011-09-15 15:53:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix stuttering A/V
	  Someone got had by implicit promotion to unsigned in ops with
	  a signed and an unsigned value.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659153

2011-09-14 16:37:12 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/debugutils/gstnavseek.c:
	  navseek: toggle pause/play on space bar
	  A useful thing to have.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659065

2011-09-14 14:46:00 +0200  David Svensson Fors <davidsf@axis.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: configurable timestamp gap handling
	  matroskademux performs segment tricks to skip gaps in streams,
	  notably at start for non 0 based files.  There may however be
	  cases when full presentation (including intermediate gaps) is
	  desired, so a property allows to configure as of which gap
	  to act (or not at all).
	  API: GstMatroskaDemux::max-gap-time
	  Fixes #659009.

2011-09-12 09:21:47 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/flvmux.c:
	  tests: flvmux: Fix flvmux's tests after fix for request pads handling
	  Now that flvmux doesn't release its request pads on PAUSED->READY the
	  test doesn't need to re-request them for every reuse test start.

2011-09-09 09:12:56 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix ctts generation for streams that don't start at 0 timestamps
	  Subtract the first timestamp of a stream from all input buffers to
	  get 0-based timestamps for creating a sane ctts table. Without this
	  patch the ctts could have larger values than needed, causing the
	  playback to have a delay at startup.
	  As the first timestamp is only found after a few buffers are queued
	  (due to possible reordered buffers), once we find the first timestamp
	  we subtract it from all buffers on the queue, from that point on,
	  all buffers have their timestamps subtract when they are collected.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658659

2011-09-12 07:55:19 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/flv/gstflvmux.c:
	  flvmux: don't release request pads going PAUSED->READY
	  Don't release request pads but just reset them. This makes pipelines using
	  flvmux reusable.

2011-09-09 12:35:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: use bsid 9 and 10 to control sample rate
	  See http://matroska.org/technical/specs/codecid/index.html
	  The spec is silent about this though...
	  https://bugzilla.gnome.org/show_bug.cgi?id=658546

2011-09-07 14:13:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: ensure some initial state variable setup
	  ... which might otherwise be skipped if the PLAY command is issued before
	  the OPEN command had a chance to actually be acted upon.
	  Fixes #657376.

2011-09-08 15:02:05 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: tweak gap handling
	  ... so as to avoid buffers before and after gap to have identical running time.

2011-09-08 13:28:24 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: use GST_RESOURCE_ERROR_BUSY if v4l2_ioctl fails with EBUSY
	  https://bugzilla.gnome.org/show_bug.cgi?id=658543

2011-09-07 08:54:17 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: remove one G_UNLIKELY for user property
	  Using G_UNLIKELY on user properties isn't nice, specially when
	  that is the default option.

2011-03-15 11:03:53 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: handle GstForceKeyUnit event
	  ... by starting a new cluster after forwarding event.
	  Fixes #644154.

2011-09-07 14:27:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/cmmldec.c:
	* tests/check/elements/cmmlenc.c:
	  cmml: Use complete cmml caps in the unit test

2011-09-07 14:26:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  qtmux: Use complete MPEG caps in the unit test

2011-09-07 14:18:58 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	  docs: cleanup makefiles
	  Remove commented out parts that we don't need. Remove "the wingo addition" - no
	  so useful after all. Narrow down file-globs for plugin docs.

2011-08-29 14:12:22 +0200  Konstantin Miller <konstantin.miller@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't handle HTTP response 407 as error if proxy authentication data is available
	  Fixes bug #657422.

2011-09-07 12:11:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: Add Converter to the classification because it can convert between different alignments
	  This allows decodebin2 to let it negotiate properly.

2011-09-07 12:10:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparsers: Improve src template caps
	  Remove the parsed/framed fields and add all fields to the template
	  caps that always exist.

2011-09-06 15:59:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	  aacparse: parse codec_data to determine number of samples per frame
	  Fixes #656734.

2011-09-06 21:24:46 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From a39eb83 to 11f0cd5

2011-09-06 16:57:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  configure: try to disable deinterlace..

2011-09-06 15:40:32 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 605cd9a to a39eb83

2011-09-06 16:37:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  common

2011-09-06 16:06:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/audioparsers/gstamrparse.c
	  gst/isomp4/qtdemux.c

2011-09-06 15:40:32 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 605cd9a to a39eb83

2011-09-06 15:05:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: make default duration check less sensitive
	  Frame duration might vary for 1 usecond, in this case matroskamux
	  decides to create BLOCKGROUP instead of SIMPLEBLOCK.
	  Convert duration to timecodescale which is (typically) less precise, and
	  then also allow the difference of 1/-1 to arrange for less sensitive check.
	  Based on patch by Alexey Fisher <bug-track@fisher-privat.net>
	  Fixes #653080.

2011-09-06 13:18:40 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: improve bogus interleaved index compensating
	  Patch by <gudake@gmail.com>
	  Fixes #654585.

2011-09-06 13:16:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjack.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiopanorama.h:
	* gst/auparse/gstauparse.c:
	* gst/avi/gstavimux.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw.c:
	* gst/spectrum/gstspectrum.c:
	* gst/wavparse/gstwavparse.c:
	  -good: port to new audio caps

2011-09-06 10:33:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Allow positive, non-1.0 segment rates
	  Only negative rates are not supported. Fixes bug #658305.

2011-09-05 15:50:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/parser.c:
	  tests: parsers: provide more real data when testing draining of garbage

2011-09-05 15:50:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	  amrparse: fix and streamline valid frame checking
	  ... to handle various combinations of sync or not, and sufficient data
	  or not as might be expected.
	  Fixes #650714.

2011-09-05 14:49:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fragmented support; avoid adjustment for keyframe seek
	  ... since all index data may not yet be available at that time.

2011-09-05 14:48:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fragmented support; mark all audio track samples as keyframe

2011-09-05 14:46:29 +0200  Brian Li <brian7003@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fragmented support; properly init return variable value
	  Fixes #655918.

2011-09-05 13:31:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add gtk-doc for new short-header property

2011-09-05 13:18:39 +0200  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: allow sending short RTSP requests to a server
	  Some encoders (Arecont) do not like the long OPTIONS sent at startup as sent by
	  GStreamer, but do accept the short header as sent by Live555.
	  This patch makes the extending the request optional by adding a property
	  (short-header).
	  Fixes #655805.
	  API: GstRTSPSrc:short-header

2009-03-04 14:51:09 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Set H263-2000 if thats what the other side wants
	  The static caps states this element supports H263-2000, but setcaps never
	  sets it, so it was lie.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=577784

2011-08-30 19:02:51 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Initialise the last_keyframe_request variable

2011-08-31 16:04:24 +0200  Peter Korsgaard <jacmet@sunsite.dk>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: make add/remove/clear/get-stats action signals
	  http://bugzilla.gnome.org/show_bug.cgi?id=657830
	  Signed-off-by: Peter Korsgaard <jacmet@sunsite.dk>

2011-08-31 18:45:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	  mp2t: fix encoding name according to RFC3551

2011-08-30 13:33:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: push mode; perform some extra checks prior to upstream seeking

2011-08-30 13:28:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: push mode; fix buffered streaming
	  That is, in case where no seek is peformed to moov, but preceding
	  limited mdat is buffered.

2011-08-30 14:06:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	  shapewipe: port to 0.11

2011-08-30 12:49:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  law is ported now

2011-08-30 12:25:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/law/alaw.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/law/mulaw.c:
	  law: port to 0.11

2011-08-29 19:11:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	  alaw: port to 0.11

2011-08-29 19:10:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: fix comment

2011-08-29 18:02:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* ext/soup/gstsouphttpsink.c:
	* ext/soup/gstsouphttpsrc.c:
	  soup: port soup elements to 0.11

2011-08-29 15:13:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid overflow wraparound in timestamp when adding durations
	  Do some type juggling to avoid overflow, while still allowing for 'negative'
	  durations (which would need a wraparound effect).

2011-08-29 13:43:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  sys/v4l2/v4l2src_calls.c

2011-08-26 14:20:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  allocation: fix for vmethod changes

2011-08-25 23:37:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: make this work more than once in a row
	  We used to skip frame rate setup if the camera was already setup
	  with the requested frame rate. This breaks some cameras though,
	  causing them to not output data (several models of Thinkpad cameras
	  have this problem at least).
	  So, don't skip.
	  https://bugzilla.gnome.org/show_bug.cgi?id=638300

2011-08-25 16:41:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/y4m/gsty4mencode.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	  port to new video flags

2011-08-24 18:40:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseutil.c:
	  pulse: add some more channels

2011-07-12 21:48:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: Add more debug

2011-07-12 19:09:02 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfcommon.h:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: Max event type is 15

2011-04-14 15:46:08 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	  dtmfsrc: Align DTMF sound buffers with last-stop from event
	  Also make sure the timestamps never go backwards

2011-07-11 21:31:07 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: Correctly recognize the end of a buffer

2011-07-11 20:47:23 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: Make sure rtpdtmfsrc timestamps don't overlap

2011-07-11 20:46:20 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: Put the inter digit interval at the end, not at the start
	  The reason is to let rtpdtmfmux drop buffers during the inter digit interval,
	  this way, there will be more silence around the DTMF tones so IVFs will have
	  a better chance recognizing them.

2011-04-14 17:08:57 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  rtpdtmfsrc: Start at the last_stop from the start event if there was one
	  The goal is to try to not have a GAP between the audio and the DTMF

2011-04-14 16:49:39 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  rtpdtmfsrc: Respect ptime from the caps
	  Respect the ptime from the caps for the DTMF packets

2011-07-11 21:30:28 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: Just error out if there is no clock

2011-08-24 14:16:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-08-23 12:12:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: only require two frames in a row when we do not have sync
	  This avoids a single bit error dropping two frames unnecessarily.
	  The two consecutive frames check is still required when we don't
	  have sync.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657080

2011-08-23 21:41:15 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Trivial indentation fix

2011-08-23 19:09:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/rtp/gstrtpvrawpay.c:
	  video: port to new colorimetry info

2011-07-21 17:23:28 -0400  Monty Montgomery <cmontgom@redhat.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Correct sample number rounding resulting in timestamp jitter
	  flacdec converts the src timestamp to a sample number, uses that internally, then reconverts the sample number to a timestamp for the output buffer.  Unfortunately, sample numbers can't be represented in an integer number of nanoseconds, and the conversion process was truncating rather than rounding, resulting in sample numbers and output timestamps that were often off by a full sample.
	  This corrects the time->sample convesion

2011-08-22 13:10:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-08-22 12:24:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/rtp/gstrtpj2kdepay.c:
	  fourcc: remove fourcc from caps

2011-08-20 14:48:20 -0700  David Schleef <ds@schleef.org>

	* gst/debugutils/breakmydata.c:
	  breakmydata: element is not passthrough

2011-07-13 11:20:34 -0700  David Schleef <ds@schleef.org>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: quiet debugging

2011-07-10 21:40:20 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: change field handling through methods
	  This likely breaks stuff.  The good: all of the methods now create
	  field images aligned with input frames, without timestamp mangling.
	  The bad: this touches a lot of code, much of which is hairy and in
	  need of cleanup.  However, at this point we can reasonably create a
	  PSNR-based test.

2011-08-21 14:41:14 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: reset ->streamheaders to NULL on _stop
	  Fixes invalid memory access reusing multifilesink

2011-08-20 10:46:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/cutter/gstcutter.c:
	* gst/cutter/gstcutter.h:
	  cutter: bring cutter somewhat into this millennium

2011-08-19 16:27:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/replaygain/gstrganalysis.c:
	  rg: fix caps

2011-08-19 16:13:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: port after merge

2011-08-19 16:12:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-08-19 16:09:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/auparse/Makefile.am:
	* gst/equalizer/gstiirequalizer.c:
	* gst/goom/gstgoom.c:
	* gst/level/Makefile.am:
	* gst/replaygain/Makefile.am:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/spectrum/gstspectrum.c:
	  port to more audio api changes

2011-08-19 14:01:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* gst/auparse/gstauparse.c:
	* gst/auparse/gstauparse.h:
	* gst/cutter/gstcutter.c:
	* gst/equalizer/gstiirequalizer.c:
	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/spectrum/gstspectrum.c:
	* sys/oss/gstosshelper.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/level.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/rtpjitterbuffer.c:
	* tests/examples/level/level-example.c:
	* tests/examples/spectrum/spectrum-example.c:
	  port more elements to new audio caps and API

2011-08-19 11:49:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofirfilter.h:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioiirfilter.h:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiokaraoke.h:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsincband.h:
	* gst/audiofx/audiowsinclimit.c:
	  port to new audio API and caps

2011-08-18 13:37:39 +0200  David Henningsson <david.henningsson@canonical.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Allow writes in bigger chunks
	  There's no use in splitting the incoming data down to the segsize
	  limit - by writing as much as possible in one chunk, we increase
	  performance and avoid PulseAudio unnecessary rewinds.
	  Signed-off-by: David Henningsson <david.henningsson@canonical.com>

2011-08-18 19:37:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-08-18 19:21:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	  port to new audio caps.

2011-08-08 22:14:28 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: ensure no-more-pads is always emitted
	  In particular, do so even if failing to read while prerolling,
	  such as when reading from a partial file (eg, while it is being
	  downloaded).
	  This fixes a wedge in playbin2.
	  https://bugzilla.gnome.org/show_bug.cgi?id=651965

2011-08-17 17:57:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: improve fixate function
	  Use new core function to fixate a field.
	  Chain up to parent fixate function.

2011-08-17 15:52:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/flac/gstflacdec.c

2011-08-17 15:39:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* ext/jpeg/Makefile.am:
	* ext/jpeg/gstjpeg.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpeg: port to 0.11
	  Also disable smoke for now.

2011-08-16 17:27:13 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: avoid timestamp/offset tracking going out of sync
	  The libFLAC API is callback based, and we must only call it to
	  output data when we know we have enough input data. For this
	  reason, a single processing step is done when receiving a buffer.
	  However, if there were metadata buffers still pending, a step
	  intended for the first audio frame might end up writing that
	  leftover metadata. Since a single step is done per buffer, this
	  will cause every buffer to be written one step late.
	  This would add some latency (a bufferfull's worth), possibly
	  lose a buffer when seeking or the like, and also cause timestamp
	  and offset to be applied to the wrong buffer, as updates to
	  the "current" segment last_stop (from incoming buffer timestamp)
	  will be applied to an output buffer originating from the previous
	  incoming buffer.
	  This fixes the issue by ensuring that, upon receiving the first
	  audio frame, processing is done till all metadata is processed,
	  so the next "single step" done will be for the audio frame. After
	  this, we should keep to 1 input buffer -> 1 output buffer and so
	  avoid getting out of sync.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650960

2011-08-17 11:17:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-08-16 15:32:07 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: bail on reserved value
	  Now that we look at the right bits, we can test against the reserved
	  value as we do for other fields.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650960

2011-08-16 15:27:43 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix bit twiddling
	  Right shifting a 8 bit value by 8 bits is twice too much
	  to get the high 4 bits.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650960

2011-08-16 15:22:46 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: warn if we see a variable block size where unsupported
	  https://bugzilla.gnome.org/show_bug.cgi?id=650960

2011-08-16 18:25:29 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	  spectrum: avoid crashing by resetting the correct number of channels
	  https://bugzilla.gnome.org/show_bug.cgi?id=656606

2011-08-16 18:35:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  sys/v4l2/v4l2src_calls.c

2011-08-16 13:16:22 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix off by one in frame size check
	  Yes, I was tracking another bug and the small test file I generated
	  to test with improbably just happened to trigger this, with a second
	  and last frame of 1615 bytes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656649

2011-08-15 12:19:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/parser.c:
	  tests: update for _negotiated_caps() change

2011-08-14 20:46:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3v2.3.0.html:
	* gst/id3demux/id3v2.4.0-frames.txt:
	* gst/id3demux/id3v2.4.0-structure.txt:
	  id3demux: remove specs from git as well now that parsing code is in -base

2011-07-14 15:42:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* gst/id3demux/Makefile.am:
	* gst/id3demux/gstid3demux.c:
	* gst/id3demux/id3tags.c:
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c:
	  id3demux: use -base provided id3 tag parsing
	  https://bugzilla.gnome.org/show_bug.cgi?id=654388

2011-08-13 16:51:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jack/gstjackaudiosrc.c:
	  jackaudiosrc: fix error message code
	  And also post 'not found' error if jackd is not even installed.

2011-08-12 16:32:58 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: initialize bitrate variable and reset for each loop
	  Don't check eventually unset variable and don't accidentially use values from last
	  cycle.

2011-08-10 11:28:26 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	  aasink: Remove unused variables

2011-08-09 11:28:17 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Properly error out if SDP contains no streams
	  Also fixes unitialized variable error on macosx.

2011-08-09 09:05:31 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: clear flags on buffer reuse
	  This will ensure a logically new buffer does not keep flags from
	  a previous use of that buffer (eg, DISCONT would be set on the first
	  buffer, and mistakenly kept when reused).
	  https://bugzilla.gnome.org/show_bug.cgi?id=653709

2011-08-08 10:54:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: take care not to change the current format where appropriate
	  Some drivers are buggy are will change the current format when
	  processing VIDIOC_TRY_FMT. Save and restore the current format
	  to ensure the format is kept unchanged.
	  https://bugzilla.gnome.org/show_bug.cgi?id=649067

2011-08-08 15:27:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update translations

2011-08-08 15:26:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/aalib/Makefile.am:
	  aalib: make sure -DGST_USE_UNSTABLE_API is defined
	  So we don't get warnings.

2011-08-08 15:25:31 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2videooverlay.c:
	* sys/v4l2/gstv4l2videooverlay.h:
	  v4l2: update for GstXOverlay => GstVideoOverlay rename

2011-08-07 12:23:26 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: Use fraction compare util function.
	  Use the fraction compare utility to compare function, not the
	  handcrafted one. The handcrafted one is buggy as it doesn't take into
	  account rounding error. For example comparing a framerate of 20/1 on a
	  camera configured as 30/1 fps would yield true: 1 == (1 * 20)/30 and not
	  re-configure the camera. Fixes #656104

2011-08-07 11:14:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: avoid race in starting
	  Sine the base class now does the negotiation from the streaming thread we have
	  to be careful and check if the stream is ready before changing its corked state.

2011-08-05 12:27:18 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* tests/check/Makefile.am:
	  check: Use GST_CFLAGS when building tests
	  Ensures we have the proper define for using unstable API

2011-08-05 08:59:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/qtdemux.c:
	  isomp4: fixup after small api changes
	  Port to recently changed api so that it compiles again.

2011-08-05 11:32:45 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/y4m/Makefile.am:
	  y4menc: Now depends on libgstvideo

2011-08-04 18:41:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulse: more cleanups

2011-08-04 18:15:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: small cleanups

2011-08-04 16:35:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: call set_caps method of baseclass
	  Call the baseclass set_caps function to make it send the caps event and
	  properly trigger the negotiation functions.

2011-08-04 16:25:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: small cleanups

2011-08-04 15:25:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/goom/gstgoom.c:
	  goom: port to new caps

2011-08-04 13:52:18 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Size variable should be a guint and not a gsize

2011-08-04 12:50:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: activate the pool in fallback
	  When nobody is using our pool, activate it ourselves.
	  Avoid leaking the buffer array.
	  Set default pool configuration with caps.
	  Don't keep current_caps, core does that for us now.

2011-08-03 22:57:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* tests/icles/videocrop-test.c:
	  fix compilation
	  hal elements were removed, remove them from docs too
	  change example for pad-block API (actually remove the pad block, an application
	  should not be bothered with working around bugs in elements)

2011-08-03 18:37:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/rtp/gstrtph264depay.c:
	  port to new API

2011-08-03 18:25:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/pulse/pulsesink.c
	  ext/pulse/pulsesrc.c
	  gst/audioparsers/gstac3parse.c
	  gst/rtp/gstrtph264depay.c
	  gst/rtp/gstrtph264pay.c
	  gst/rtpmanager/gstrtpssrcdemux.c

2011-08-03 22:50:05 +1000  Jan Schmidt <thaytan@noraisin.net>

	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	* gst/matroska/matroska.c:
	  matroska: Register new debug category
	  Register the matroskareadcommon debug category when the
	  plugin is loaded to avoid assertion output when debug is turned on.

2011-08-03 13:38:01 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* tests/icles/gdkpixbufsink-test.c:
	  test/ickles: Port gdkpixbufsink test

2011-08-03 13:33:59 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/autodetect.c:
	  Revert "tests/check/Makefile.am: Disable autodetect test temporarily, so that the build bots update -bad and the ranks of unr..."
	  This reverts commit 475aed8af6d2a57c1d21490c824e754a6b2367a9.
	  It won't consider elements from anywhere else anymore

2011-08-03 13:10:46 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/parser.c:
	  check: Update parser mini-lib to 0.11 API

2011-08-03 13:09:07 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* po/POTFILES.in:
	  po: update for modified source file location

2011-08-03 13:08:43 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* configure.ac:
	  configure.ac: cairo_gobject isn't ported either

2011-08-03 10:59:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/Makefile.am:
	* ext/hal/Makefile.am:
	* ext/hal/gsthalaudiosink.c:
	* ext/hal/gsthalaudiosink.h:
	* ext/hal/gsthalaudiosrc.c:
	* ext/hal/gsthalaudiosrc.h:
	* ext/hal/gsthalelements.c:
	* ext/hal/gsthalelements.h:
	* ext/hal/hal.c:
	* ext/hal/hal.h:
	  hal: Remove hal plugin
	  hal is not developed anymore and nobody is using the plugin nowadays.

2011-07-29 13:03:55 +0200  Philippe Normand <pnormand@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: soften assertion check on stream size
	  https://bugzilla.gnome.org/show_bug.cgi?id=655570

2011-08-03 10:09:42 +0200  Robert Krakora <rob.krakora@messagenetsystems.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Add support for H.264 payload in MJPEG container
	  See http://www.quickcamteam.net/uvc-h264/USB_Video_Payload_H.264_0.87.pdf
	  Fixes bug #655530.

2011-08-02 22:05:08 -0400  Tristan Matthews <tristan@sat.qc.ca>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	  jackaudiosink: Don't call g_alloca() in process_cb
	  g_alloca() is not RT-safe, so instead we should allocate the
	  memory needed in advance. Fixes #655866

2011-08-03 08:58:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Add hal to the list of non-ported plugins

2011-08-03 08:53:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Add monoscope to the list of non-ported plugins

2011-08-03 08:51:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstquark.c:
	* gst/effectv/gstwarp.c:
	  effectv: Fix unused but set variable compiler warnings

2011-08-02 23:42:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	* sys/v4l2/gstv4l2object.c:
	  docs: fix two more Since: tags

2011-07-31 04:19:00 +0300  Mart Raudsepp <leio@gentoo.org>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix Since tags for fieldanalysis related new properties
	  commit c1b100cf9c is after 0.10.29 and 0.10.30 was a branched release.
	  So fix Since tags from 0.10.29 to 0.10.31 for the new properties.

2011-08-02 11:51:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: fix porting error

2011-08-02 11:29:40 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* configure.ac:
	  configure.ac: Define list of non-ported plugins

2011-08-02 11:29:25 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* common:
	  Update common submodule

2011-08-02 11:17:38 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* configure.ac:
	  configure.ac: Sort AG_GST_CHECK_PLUGIN alphabetically

2011-07-29 17:27:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawdepay.h:
	  -good: fix for bufferpool API change

2011-07-29 17:21:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l: change for new API

2011-07-29 13:05:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix variable-set-but-not-used compiler warning with older pulse versions

2011-07-29 12:07:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: properly init rtcp_min_interval

2011-03-09 11:04:36 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.c:
	  pulsesink: Add support for compressed formats
	  This adds support for various compressed formats (AC3, E-AC3, DTS and
	  MP3) payloaded in IEC 61937 format (used for transmission over S/PDIF,
	  HDMI and Bluetooth).
	  The acceptcaps() function allows bins to probe for what formats the sink
	  being connected to support. This only works after the element is set to
	  at least READY.
	  If the underlying sink changes and the format we are streaming is not
	  available, we emit a message that will allow upstream elements/bins to
	  block and renegotiate a new format.

2011-03-01 15:34:46 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulsesink: Use the extended stream API if available
	  This uses the new extended API for creating streams. This will allow us
	  to support compressed formats natively in pulsesink as well.

2011-07-29 00:07:52 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: Add a source-output-index property
	  This exposes the source output index of the record stream that we open
	  so that clients can use this with the introspection if they want (to
	  move the stream, for example).

2011-07-28 14:44:57 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: keep a ref on the src pad while using it
	  Prevent a possible race if clear_ssrc() is called between getting the pad and
	  doing the push.
	  Based on patch by <olivier.crete@collabora.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=650916

2011-05-24 11:29:57 +0300  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	  rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit
	  We need to keep the lock held because we don't want a push before the "new-ssrc-pad"
	  handler has completed. But we may want to push an event from inside that handler, hence
	  the recursive mutex.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650916

2011-05-24 11:17:25 +0300  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Use PADs lock
	  https://bugzilla.gnome.org/show_bug.cgi?id=650916

2011-07-28 11:09:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	  speex: update for position/query/convert API changes

2011-07-28 10:54:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/auparse/gstauparse.c:
	* gst/avi/gstavidemux.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/progressreport.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/isomp4/qtdemux.c:
	* gst/wavparse/gstwavparse.c:
	  gst: udpate for position/duration/convert query API changes

2011-07-28 00:37:13 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix compiler warning
	  gstavidemux.c: In function 'gst_avi_demux_parse_stream':
	  gstavidemux.c:1261:24: error: 'data' may be used uninitialized in this function [-Werror=uninitialized]
	  gstavidemux.c:1204:11: note: 'data' was declared here

2011-07-27 18:15:20 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: Cope with FU-A E bit not being set
	  Some h264 payloaders are unfortunately buggy and don't correctly set the
	  E bit in FU-A NAL when they have ended. Work around this by assuming
	  such a fragmentation unit has ended when there was no packet loss and a
	  new NAL is started

2011-04-12 17:01:47 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	  ac3parse: Support switching alignment on-the-fly
	  This allows switching of alignment for E-AC3 streams at run-time. This
	  is requested by downstream elements via a custom event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650313

2011-07-27 16:46:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: remove unused variables
	  Use the more specialized type for the bufferpool.
	  Use the size from the driver as the size of the image to read.
	  Don't configure the pool when created. This will be done in the setup_allocation
	  method later or by upstream for sinks.
	  Remove unused properties and variables. Bufferpool sizes are now configured in
	  the bufferpool by the elements in the pipeline. We might want to influence the
	  pool size later somehow.

2011-07-27 13:46:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2bufferpool: remove unused variable

2011-07-27 13:43:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: add metadata

2011-07-27 13:41:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  bufferpool: check for metadata
	  Only add video metadata when it was configured in the pool. Fail if there was no
	  video metadata configured and the strides are not the default ones.

2011-07-27 12:42:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	* gst/effectv/gstwarp.h:
	  warp: add stride support

2011-07-27 12:41:33 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: add colorspace to debug

2011-07-26 17:45:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtp: fix compilation

2011-07-26 16:15:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: rename a variable
	  Rename the size variable to sizeimage and fill it with the size that has been
	  given to use by the v4l2 driver instead of making something up..

2011-07-26 13:18:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2: use new setup_allocation vmethod

2011-07-26 10:56:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: implement more bits of RW I/O mode
	  Implement the relaese of RW buffers in the pool.
	  Warn for unsupported write() mode for sinks.

2011-07-26 10:54:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: improve IO mode error handling
	  Error out when an unsupported IO mode was selected

2011-04-09 12:26:56 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	* tests/check/elements/ac3parse.c:
	  ac3parse: Add support for IEC 61937 alignment
	  When pushing out buffers over S/PDIF or HDMI, IEC 61937 payloading
	  requires each buffer to contain 6 blocks from each substream. This adds
	  code to collect all the frames needed to meet this requirement before
	  pushing out a buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650313

2011-06-08 15:57:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Always send application requested feedback in immediate mode
	  Send as many application requested feedback messages in immediate mode, even if they
	  have already been sent.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654583

2011-06-08 14:48:01 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Don't let the computed RTP bandwidth fall too low
	  If it falls too low, the computed RTCP bandwidth will be near zero and
	  the RTCP thread will be stopped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654583

2011-04-25 16:13:38 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Wait longer to timeout SSRC collision
	  Using the current RTCP interval to timeout SSRC collision can lead to
	  collisions being timed out immediately if a BYE packet is sent because
	  it is sent immediately, so the interval is 0. This is not what we
	  want. So just set a static 10 times the default RTCP interval, it
	  should be enough
	  https://bugzilla.gnome.org/show_bug.cgi?id=648642

2011-07-25 15:51:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: remove unused method

2011-07-25 15:38:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix flushing start and stop
	  Move the flushing calls to the right place in the bufferpool.
	  Fix the min and max buffer sizes.

2011-07-25 14:47:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: dequeue buffers when all are queued
	  Prefer to always use the default bufferpool queue for the _acquire function
	  because it properly supports unblocking when setting inactive etc. As a result,
	  we need to dequeue buffers and put them back in the bufferpool queue when we
	  have queued all buffers in the sink.
	  Rename some variables to more meaningfull names to avoid a problem with
	  freeing the wrong amount of buffers.

2011-07-19 13:38:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: set SOURCE flag at init time
	  Fixes #654816.

2011-07-25 10:10:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstvertigo.c:
	  vertigotv: add stride support

2011-07-19 18:25:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: only to STREAMOFF when streaming
	  Only call STREAMOFF when we previously called STREAMON

2011-07-22 21:26:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/replaygain/gstrganalysis.c:
	  replay: fix for event handler

2011-07-22 21:19:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/progressreport.c:
	  fixes for event handler changes

2011-07-18 16:46:27 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Complete merged AU on marker bit
	  The marker bit on a RTP packet means the AU has been completed, so push it out
	  immediately to reduce the latency.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654850

2011-07-18 20:27:38 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Only set the marker bit on the last NALU of a multi-NALU access unit
	  An access unit could contain multiple NAL units, in that case, only the last
	  RTP packet of the last NALU should have its marker bit set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654850

2011-07-20 08:52:58 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multipart/multipartmux.c:
	  multipart: fix compiler warning

2011-07-19 18:20:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2: handle unsupported formats

2011-07-19 16:59:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2: Fix sink bufferpool handling
	  Remove old method, use neww _process method for the sink.
	  Inform the parent bufferpool class about the settings too. This is needed to let
	  it know about the max-buffers.
	  Allocate the negotiated max-buffers and initially mmap min-buffers. The idea is
	  that the bufferpool will allocate more when needed.
	  Improve debugging.
	  Only poll in capture mode, it does not seem to work in playback mode on this
	  beagleboard.

2011-07-19 12:05:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/auparse/gstauparse.c:
	  auparse: avoid hanging on invalid short input
	  ... as in such case there is no srcpad yet on which to forward EOS.

2011-07-18 15:13:33 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Fix default value leaking
	  Remember to free the default value of client name, avoiding a
	  leak

2011-07-18 18:54:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2: More work on bufferpools
	  Add different transport methods to the bufferpool (MMAP and READ/WRITE)
	  Do more parsing of the bufferpool config.
	  Start and stop streaming based on the bufferpool state.
	  Make separate methods for getting a buffer from the pool and filling it with
	  data. This allows us to fill buffers from other pools too. Either use copy or
	  read to fill up the target buffers.
	  Add property to force a transfer mode in v4l2src.
	  Increase default number of buffers to 4.
	  Negotiate bufferpool and its properties in v4l2src.

2011-07-18 14:24:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: reset upon FLUSH_STOP
	  ... which is particularly needed when merging NAL units, where not resetting
	  would lead to output of an older (pre-flush) AU (with unintended timestamp).

2011-07-18 14:30:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: do not use g_slist_free_full
	  ... as that is only in GLib 2.28, which is not yet required at this time.

2011-07-18 10:52:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: add IO method enum

2011-07-18 10:51:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  bufferpool: improve _new function

2011-07-18 09:38:26 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	* tests/check/elements/multifile.c:
	  multifilesink: add max-files property
	  Add max-files property to limit the number of files saved on disk.
	  API: multifilesink::max-files

2011-07-17 23:36:55 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: refactor file opening and closing code

2011-07-16 19:38:51 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix pixel-aspect-ratio if header has only one display variable
	  Current matroska demux calculates the pixel aspect ratio only if both
	  DisplayHeight and DisplayWidth are set, but it is legal to use only
	  one variable if the other is equal to PixelWidth or PixelHeight, at
	  least the mkclean utility is doing that. So this makse mkcleaned
	  files play correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654744

2011-07-16 23:47:50 +0100  Antoine Jacoutot <ajacoutot@openbsd.org>

	* gst/goom/plugin_info.c:
	  goom: fix build on PPC on openbsd
	  A missing sys/param.h include results in:
	  /usr/include/sys/proc.h:64: error: 'MAXLOGNAME' undeclared here (not in a
	  function)
	  /usr/include/sys/proc.h:285: error: 'MAXCOMLEN' undeclared here (not in a
	  function)
	  when compiling goom on openbsd/ppc. We can just remove the two sys/ includes
	  here, they are not needed for anything.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654749

2011-07-15 17:06:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-07-15 16:55:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: implement setup_allocation
	  Implement the setup_allocation vmethod, we'll hopefully do something clever in
	  there later.

2011-07-15 16:26:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: improve bufferpool config setting
	  Pass the caps and the default video size to the bufferpool config.
	  Don't activate the bufferpool, this will be done by the object that decides to
	  use the bufferpool.
	  Improve debugging and error reporting.

2011-07-15 13:52:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: handle dequeueing correcly
	  First clean up the buffers in the queue, then the remaining ones in the
	  device.

2011-07-15 13:29:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: unref copied buffer
	  After we copy the incomming buffer to one of our bufferpool buffers, unref the
	  target buffer after rendering so that it is put back in the pool.

2011-07-15 13:07:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: dequeue buffers for the sink
	  When we have all buffers queued for playback and we need a new empty buffer,
	  dequeue one and return it.
	  Set the right size for sink buffers.
	  Improve counting of queued buffers.

2011-07-15 12:35:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: use the parent queue for the sink
	  We want to maintain a queue of free buffers for the sink, use the parent methods
	  to do that.

2011-07-15 12:00:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix error messages

2011-07-15 11:30:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2: add ALLOCATION query to the sink

2011-07-15 11:27:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: convert to GstBufferPool
	  Extend from GstBufferPool.
	  Handle the lifetime of the pool buffers correctly with the start/stop vmethods.
	  Map acquire and release directly to QBUF and DQBUF. We still expose an explicit
	  qbuf for the v4l2sink for now.

2011-07-15 11:18:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: remove experimental markers

2011-07-14 20:10:02 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	  rtppcmApay/depay: Static clock rates on static payloads, dynamic on dynamic
	  Partially reverts 397dc60b

2011-07-14 16:21:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: merge code

2011-07-14 16:12:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	  v4l2: Move output details to device object
	  Move the details of how a buffer is rendered to the device object.

2011-03-04 15:41:22 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Implement getcaps
	  Convert profile-level-id from RTP caps into video/x-h264 style caps (with profile and level)

2011-07-13 18:32:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2vidorient.c:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  v4l2: move capture code to device object
	  Move the details of how to capture to the device object. Remove the
	  v4l2src_calls.[ch] files because they are empty now.
	  Provide two simple methods to get and return a buffer to the device.
	  Also do a slow copy when the buffer is not from our pool.

2011-07-13 16:58:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: add some more debug

2011-07-13 16:56:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2: stop streaming in READY and NULL

2011-07-13 16:40:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: start streaming for the output as well

2011-07-13 16:33:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  v4l2: Let the device object manage the pool
	  Rename start and stop methods to open and close because that is what they do.
	  After setting the format on the device object, setup the bufferpools. Move this
	  code from the v4l2src_calls.c file, it is shared between source and sink.
	  Make new device start and stop method that merges various bits of common code
	  spread over several files.

2011-07-13 13:52:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: don't store stride in the videoinfo
	  We want to keep the default strides in the videoinfo. Keep the stride of the
	  video frames separate so that we can use both to copy a video frame and do
	  correct stride conversion.

2011-07-13 13:38:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2: Use video frame copy for raw video
	  Use the video frame copy API for raw video frames so that we copy with the right
	  strides.

2011-07-13 13:37:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: add video metadata to raw video buffers

2011-07-13 13:15:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: small cleanups

2011-07-13 13:00:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: improve caps parsing
	  Use GstVideoInfo to store the parsed caps.
	  Remove outsize from the caps parsing code, it's wrong because it does not use
	  the stride given by the driver.

2011-07-13 11:40:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: use errno

2011-07-13 11:36:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: handle EINVAL without posting a warning
	  EINVAL means that a call is not supported, we only want to post a WARNING when
	  something is really wrong.

2011-07-13 11:29:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: only set framerate for capture for now

2011-07-13 11:19:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2_calls.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  v4l2: Move configuration of framerate to _set_format
	  Move the configuration of the framerate to where we set the other format
	  parameters.
	  Remove hack to check if the device is active.
	  Store streamparm in the device info.
	  Use some macros to access the current device configuration.
	  Remove some duplicate fields in src and sink and use the device configuration
	  instead.

2011-07-12 19:13:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix return value...

2011-07-12 19:03:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  v4l2: simplify setting the capture format
	  Pass the caps to the set_format function and make _set_format parse the caps.
	  Also keep the parsed values in the v4l2object so that we can refer to them when
	  we want.

2011-07-12 18:41:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  v4l2: remove more unused parameters

2011-07-12 18:29:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l: handle object out of the normal flow

2011-07-12 18:13:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2: Let the bufferpool own the V4l2Object
	  Keep track of the currently configured format and setting in the
	  v4l2object.
	  Pass the v4l2object to the bufferpool constructor so that the bufferpool can
	  know everything about the currently configured settings. This also allows us
	  to remove some awkward code.

2011-07-12 17:06:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l: remove caps argument, it's not needed
	  Remove the caps parameter, we don't need it anymore because we don't set
	  caps on buffers anymore.

2011-07-12 16:46:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l: pass the bytesperline around
	  When setting a format, return the bytesperline to the caller so that it can be
	  used to allocate buffers.

2011-07-12 16:43:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  pool: make buffer writable
	  We need writable buffers when we need to do a slow memcpy.

2011-07-12 15:04:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix seeking regression
	  ... introduced when shuffling around code for the async implementation
	  by setting state of source (and udp sources) in _play before downstream
	  flushing is undone.

2011-07-11 15:23:41 +0300  René Stadler <rene.stadler@nokia.com>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	  ac3parse: fix buffer duration on blocks-per-frame change
	  The gst_base_parse_set_frame_rate call was predicated on a change to
	  sample rate, duration or profile. However, the block count per frame can
	  also change between packets, which would result in incorrect buffer
	  durations.

2011-07-11 13:51:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: handle pools
	  Create a new pool in setcaps and stop/destroy the old one.
	  Remove buffer_alloc functions.
	  Check that we have v4l2 metadata in show_frame and fall back to memcpy into a
	  buffer from our pool if we don't receive one of our own buffers.

2011-07-11 12:04:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2: various cleanups
	  Various cleanups, avoids useless casts, move error handling outside of the main
	  code flow.
	  Negotiate to a resonable resolution instead of the max resolution.

2011-07-10 21:50:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawdepay.h:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtp/gstrtpvrawpay.h:
	  rtp: port remaining to 0.11

2011-07-10 14:56:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/ximageutil.c:
	  ximage: port to 0.11

2011-07-10 13:44:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  y4m: port some more
	  Use video helpers.

2011-07-10 13:28:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/y4m/gsty4mencode.c:
	  y4m: port to 0.11

2011-07-10 12:46:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/multipart/multipartmux.h:
	  multipart: port to 0.11

2011-07-10 11:42:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-07-10 11:40:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/debugutils/Makefile.am:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/efence.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstdebug.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavigationtest.h:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/tests.c:
	  debug: port to 0.11, disable others
	  Diasable the efence and capsdebug elements, port them later.

2011-07-09 19:23:41 -0700  David Schleef <ds@schleef.org>

	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstmultifilesrc.h:
	  multifilesrc: Improve looping
	  Add start-index and stop-index properties.

2011-06-16 13:57:03 +0100  Jonny Lamb <jonnylamb@jonnylamb.com>

	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstmultifilesrc.h:
	  multifile: add loop property to multifilesrc
	  Fixes: #652727
	  Signed-off-by: Jonny Lamb <jonnylamb@jonnylamb.com>
	  Signed-off-by: David Schleef <ds@schleef.org>

2009-11-20 10:07:43 +0100  Philip Jägenstedt <philipj@opera.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: 16-bit audio is signed, 8-bit is unsigned.
	  Pretending to handle 8-bit signed causes distorted audio when
	  actually given such audio, which you will get if passing 8-bit
	  unsigned through audioconvert ! audioresample, as audioresample
	  only handles 8-bit signed.  Fixes #605834.
	  Signed-off-by: David Schleef <ds@schleef.org>

2011-07-08 16:37:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2: fix gray format, use filter in getcaps

2011-07-08 16:10:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	  v4l2: port and enable v4l2sink

2011-07-08 14:34:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2src: port to new video formats

2011-07-08 12:51:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-07-08 12:49:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2colorbalance.c:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2: port to 0.11

2011-07-07 18:27:36 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: handle blocks with duration=0
	  Some video frames, for example alt-ref frame in VP8, will be
	  never displayed. This is why it has duration=0.
	  This patch allow to use this duration.
	  Bug: 654175
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>

2011-07-06 17:18:05 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Add direct dirac mapping

2011-07-07 17:59:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstripple.c:
	* gst/effectv/gstripple.h:
	  effectv: port last effectv element to 0.11

2011-07-07 17:49:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstradioac.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gststreak.h:
	  effectv: port streaktv to 0.11

2011-07-07 17:40:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstradioac.c:
	* gst/effectv/gstradioac.h:
	  effectv: port radioactv to 0.11

2011-07-07 17:29:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	  effectv: fix docs

2011-07-07 17:29:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstop.c:
	* gst/effectv/gstop.h:
	  effectv: port op to 0.11

2011-07-07 17:18:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstquark.c:
	* gst/effectv/gstquark.h:
	* gst/effectv/gstrev.c:
	  effectv: port quark tv

2011-07-07 16:57:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstrev.c:
	* gst/effectv/gstrev.h:
	  effectv: port revtv to 0.11

2011-07-07 16:46:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstvertigo.h:
	  effectv: port vertigotv to 0.11

2011-07-07 16:38:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstshagadelic.h:
	  effectv: port shagadelictv to 0.11

2011-07-07 11:22:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/auparse/gstauparse.c:
	  auparse: use ALWAYS src pad rather than SOMETIMES

2011-07-07 11:14:16 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/auparse/gstauparse.c:
	  auparse: port to 0.11

2011-07-06 19:03:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: beginnings of porting

2011-07-06 18:50:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	* gst/effectv/gstwarp.h:
	  warptv: port to 0.11

2011-07-06 18:50:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstdice.c:
	  dice: keep track of info

2011-07-06 18:32:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstdice.c:
	* gst/effectv/gstdice.h:
	  effectv: port dice

2011-07-06 18:09:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstaging.h:
	  effectv: port agingtv

2011-07-06 17:50:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/Makefile.am:
	* ext/aalib/gstaasink.c:
	* ext/aalib/gstaasink.h:
	  aasink: port to new video API

2011-07-06 17:40:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/libcaca/Makefile.am:
	* ext/libcaca/gstcacasink.c:
	* ext/libcaca/gstcacasink.h:
	  cacasink: port to 0.11

2011-07-06 16:50:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpeg: beginnings of porting to 0.11

2011-07-06 16:31:18 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: use ALWAYS source pad rather than SOMETIMES

2011-07-06 16:10:34 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: port to 0.11

2011-07-06 16:10:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	  wavenc: port to 0.11

2011-07-06 12:22:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: adjust to unsigned segment fields

2011-07-06 15:57:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	  speex: port speex elements

2011-07-06 12:05:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-07-06 10:11:52 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: port to 0.11
	  * use G_DEFINE_TYPE
	  * adjust to new GstBuffer and corresponding rtp and rtcp buffer interfaces
	  * misc caps and segment handling changes
	  FIXME: also relies on being able to pass caps along with a buffer,
	  which has no evident equivalent yet, so that either needs one,
	  or still needs quite some code path modification to drag along caps.

2011-06-29 20:59:26 +0300  René Stadler <rene.stadler@nokia.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: prevent race condition causing ref leak
	  Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the
	  deferred call to be run before returning. This causes a race when
	  READY->NULL is executed shortly after, which stops the mainloop. This
	  leaks the element reference which is passed as userdata for the callback
	  (introduced in commit 7cf996, bug #614765).
	  The correct fix is to wait in READY->NULL for all outstanding calls to
	  be fired (since libpulse doesn't provide a DestroyNotify for the
	  userdata). We get rid of the reference passing from 7cf996 altogether,
	  since finalization from the callback would anyways lead to a deadlock.
	  Re-fixes bug #614765.

2011-07-04 08:58:14 +0300  René Stadler <rene.stadler@nokia.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: small cleanup of copy-paste code

2011-06-29 19:50:42 +0300  René Stadler <rene.stadler@nokia.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: remove unused member variable and misleading log message
	  Wim changed it in commit 8bfd80 so that pa_defer_ran is not read
	  anywhere.
	  The log message used to annotate a mainloop_wait call which is gone.

2011-07-05 15:37:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: fix caps

2011-07-05 11:40:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/effectv/gstedge.c:
	* gst/effectv/gstedge.h:
	  effectv: port edgetv

2011-07-05 10:12:25 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  Add -DGST_USE_UNSTABLE_API to the compiler flags to avoid warnings

2011-07-04 12:58:38 -0700  David Schleef <ds@schleef.org>

	* gst/goom/gstgoom.c:
	  goom: Don't answer lantency queries before negotiation

2011-07-04 18:15:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	  udp: port to new API

2011-07-04 18:12:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsemixer.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulse: remove implementsinterface

2011-07-04 18:10:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: fix caps

2011-07-04 18:06:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  alpha: port to new video API

2011-07-04 17:00:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: more porting

2011-07-04 16:09:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  port to new video api

2011-06-28 14:03:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstgamma.h:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideobalance.h:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	  video: port to new video apis

2011-07-04 14:30:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: avoid crashing on invalid input without components

2011-07-04 11:09:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvmux.c:
	  flv: port to 0.11
	  * use G_DEFINE_TYPE
	  * adjust to new GstBuffer
	  * misc segment and caps changes

2011-07-04 11:48:13 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/pulse/pulsesink.c

2011-07-04 11:25:28 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: pass along segment info to collectpads
	  ... so it can track this and be subsequently used to determine running time etc.

2011-07-04 11:24:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: indicate raw format in aac caps

2011-07-04 11:07:13 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: mind requested name for request pad

2011-07-04 11:06:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: free scheduling query

2011-07-03 19:51:32 -0700  David Schleef <ds@schleef.org>

	* ext/pulse/plugin.c:
	  pulse: Increase ranks to PRIMARY + 10
	  So that pulsesrc/pulsesink get chosen over other possible PRIMARY
	  src/sinks by autoaudiosink.  Presumably, if pulse is available, it
	  is always preferred over another src/sink.
	  Fixes: #647540.

2011-06-30 18:47:48 -0700  David Schleef <ds@schleef.org>

	* gst/multipart/multipartmux.c:
	  multipartmux: Add \r\n to tail of pushed buffers
	  Clients such as Firefox require the \r\n after the payload.

2011-06-16 14:52:51 +0200  Branko Subasic <branko@axis.com>

	* gst/matroska/ebml-read.c:
	* gst/matroska/matroska-demux.c:
	  matroskademux: avoid looping when searching for clusters
	  Fixes some bugs that results in the demuxer looping when seaching
	  for clusters in non-finalized files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=652195

2011-06-30 12:30:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	  multifile: port to 0.10
	  * use G_DEFINE_TYPE
	  * adjust to new GstBuffer
	  * misc caps handling

2011-06-30 11:35:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/cutter/gstcutter.c:
	  cutter: port to 0.11
	  * use G_DEFINE_TYPE
	  * adjust to new GstBuffer
	  * minor misc

2011-06-30 11:17:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	  replaygain: port to 0.11
	  * use G_DEFINE_TYPE
	  * adjust to new GstBuffer

2011-06-30 10:53:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	  spectrum: remove deprecated property

2011-06-30 10:51:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	  spectrum: port to 0.11
	  * use G_DEFINE_TYPE
	  * adjust to new GstBuffer

2011-06-30 10:38:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/level/gstlevel.c:
	  level: port to 0.11
	  * use G_DEFINE_TYPE
	  * adjust to new GstBuffer

2011-06-30 10:30:16 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/equalizer/gstiirequalizernbands.c:
	  equalizer: port to 0.11

2011-06-10 18:54:48 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: fix reference counting of parse->streamheader
	  https://bugzilla.gnome.org/show_bug.cgi?id=652286
	  Signed-off-by: David Schleef <ds@schleef.org>

2011-06-29 14:39:52 -0700  David Schleef <ds@schleef.org>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Don't round up size of encoded buffers
	  For some reason, in code dating to 2001, encoded jpeg buffers were
	  rounded up to multiples of 4 bytes.  With the added bonus that the
	  extra bytes are unwritten, causing valgrind issues.  Oops.  I can't
	  think of any reason why JPEG buffers need to be multiples of 4 bytes,
	  so I removed the padding.  There might be some code somewhere that
	  depends on this behavior, so if this needs to be reverted, please fix
	  the valgrind issues.

2011-06-29 12:46:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/Makefile.am:
	* gst/isomp4/atoms.c:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  isomp4: port to 0.11

2011-06-28 12:55:45 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: tweak some ported segment handling
	  ... to avoid losing duration during push mode seeking, and to properly
	  accumulate running time when segment seeking.

2011-06-29 12:05:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: free date tag

2011-06-28 12:26:37 +0200  Jonas Larsson <jonas.larsson@hiq.se>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: not so greedy minimum frame size
	  Fixes #653559.

2011-06-25 11:39:23 -0700  David Schleef <ds@schleef.org>

	* configure.ac:
	  configure: remove non-pkg-config check for shout
	  Fixes: 653327

2011-06-20 18:49:57 +0200  Andoni Morales Alastruey <amorales@flumotion.com>

	* ext/raw1394/gst1394clock.c:
	  dv1394src: make the internal clock thread safe
	  Fixes: #653091.

2011-06-24 11:54:29 +0200  Miguel Angel Cabrera Moya <madmac2501@gmail.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: return correct type when assertion fails

2011-06-23 11:28:27 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From 69b981f to 605cd9a

2011-06-22 16:41:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: fix for uri changes

2011-02-02 16:18:54 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulse: Drop support for PA versions before 0.9.16
	  This drops support fof PulseAudio versions prior to 0.9.16, which was
	  released about 1.5 years ago. Testing with very old versions is not
	  feasible and we don't want to maintain 2 independent code-paths.

2011-06-21 18:24:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  configure.ac
	  docs/plugins/inspect/plugin-esdsink.xml
	  docs/plugins/inspect/plugin-gconfelements.xml

2011-06-21 18:19:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix for header cleanups

2011-06-21 15:15:06 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	  rtpmp4adepay: fix output buffer timestamps in case of multiple frames

2011-06-20 16:47:36 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: The signal has 5 arguments, not 4

2011-06-20 12:13:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: use string for video format now

2011-06-20 12:04:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/Makefile.am:
	  avi: link against gstvideo now

2011-06-20 12:03:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avi: port to new caps

2011-06-18 13:43:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Bump git version after unplanned 0.10.30 release
	  Merge branch '0.10.30'
	  Conflicts:
	  configure.ac
	  docs/plugins/inspect/plugin-1394.xml
	  docs/plugins/inspect/plugin-aasink.xml
	  docs/plugins/inspect/plugin-alaw.xml
	  docs/plugins/inspect/plugin-alpha.xml
	  docs/plugins/inspect/plugin-alphacolor.xml
	  docs/plugins/inspect/plugin-annodex.xml
	  docs/plugins/inspect/plugin-apetag.xml
	  docs/plugins/inspect/plugin-audiofx.xml
	  docs/plugins/inspect/plugin-audioparsers.xml
	  docs/plugins/inspect/plugin-auparse.xml
	  docs/plugins/inspect/plugin-autodetect.xml
	  docs/plugins/inspect/plugin-avi.xml
	  docs/plugins/inspect/plugin-cacasink.xml
	  docs/plugins/inspect/plugin-cairo.xml
	  docs/plugins/inspect/plugin-cutter.xml
	  docs/plugins/inspect/plugin-debug.xml
	  docs/plugins/inspect/plugin-deinterlace.xml
	  docs/plugins/inspect/plugin-dv.xml
	  docs/plugins/inspect/plugin-efence.xml
	  docs/plugins/inspect/plugin-effectv.xml
	  docs/plugins/inspect/plugin-equalizer.xml
	  docs/plugins/inspect/plugin-esdsink.xml
	  docs/plugins/inspect/plugin-flac.xml
	  docs/plugins/inspect/plugin-flv.xml
	  docs/plugins/inspect/plugin-flxdec.xml
	  docs/plugins/inspect/plugin-gconfelements.xml
	  docs/plugins/inspect/plugin-gdkpixbuf.xml
	  docs/plugins/inspect/plugin-goom.xml
	  docs/plugins/inspect/plugin-goom2k1.xml
	  docs/plugins/inspect/plugin-gstrtpmanager.xml
	  docs/plugins/inspect/plugin-halelements.xml
	  docs/plugins/inspect/plugin-icydemux.xml
	  docs/plugins/inspect/plugin-id3demux.xml
	  docs/plugins/inspect/plugin-imagefreeze.xml
	  docs/plugins/inspect/plugin-interleave.xml
	  docs/plugins/inspect/plugin-isomp4.xml
	  docs/plugins/inspect/plugin-jack.xml
	  docs/plugins/inspect/plugin-jpeg.xml
	  docs/plugins/inspect/plugin-level.xml
	  docs/plugins/inspect/plugin-matroska.xml
	  docs/plugins/inspect/plugin-mulaw.xml
	  docs/plugins/inspect/plugin-multifile.xml
	  docs/plugins/inspect/plugin-multipart.xml
	  docs/plugins/inspect/plugin-navigationtest.xml
	  docs/plugins/inspect/plugin-oss4.xml
	  docs/plugins/inspect/plugin-ossaudio.xml
	  docs/plugins/inspect/plugin-png.xml
	  docs/plugins/inspect/plugin-pulseaudio.xml
	  docs/plugins/inspect/plugin-replaygain.xml
	  docs/plugins/inspect/plugin-rtp.xml
	  docs/plugins/inspect/plugin-rtsp.xml
	  docs/plugins/inspect/plugin-shapewipe.xml
	  docs/plugins/inspect/plugin-shout2send.xml
	  docs/plugins/inspect/plugin-smpte.xml
	  docs/plugins/inspect/plugin-soup.xml
	  docs/plugins/inspect/plugin-spectrum.xml
	  docs/plugins/inspect/plugin-speex.xml
	  docs/plugins/inspect/plugin-taglib.xml
	  docs/plugins/inspect/plugin-udp.xml
	  docs/plugins/inspect/plugin-video4linux2.xml
	  docs/plugins/inspect/plugin-videobox.xml
	  docs/plugins/inspect/plugin-videocrop.xml
	  docs/plugins/inspect/plugin-videofilter.xml
	  docs/plugins/inspect/plugin-videomixer.xml
	  docs/plugins/inspect/plugin-wavenc.xml
	  docs/plugins/inspect/plugin-wavpack.xml
	  docs/plugins/inspect/plugin-wavparse.xml
	  docs/plugins/inspect/plugin-ximagesrc.xml
	  docs/plugins/inspect/plugin-y4menc.xml
	  win32/common/config.h

2011-06-17 10:37:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosink.h:
	  sunaudio: fix typo in comment

2011-06-17 18:12:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-06-17 18:11:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	  autodetect: fix caps

2011-06-16 15:38:10 +0200  Luis de Bethencourt <luis.debethencourt@collabora.com>

	* gst/goom/gstgoom.c:
	  goom: fix unused-but-set-compiler warnings
	  Remove unnecessary res variables, core checks existance
	  and type of these fields for us already via the template
	  caps, and we know that these fields exist because we've
	  fixated them before in _negotiate().

2011-06-17 03:07:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/audiofx/audioecho.c:
	  audioecho: fix param flags
	  If the parameter cannot be changed in paused&playing, it is not controlable. Set
	  the appropriate mutability flag instead.

=== release 0.10.30 ===

2011-06-15 23:57:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.30
	  This is an ad-hoc release that is almost identical to 0.10.29:
	  * work around GLib atomic ops API change
	  * better handling of malformed buffers in RTP depayloders
	  * some minor compilation fixes

2011-06-08 18:33:10 +0300  Raimo Järvi <raimo.jarvi@gmail.com>

	* gst/udp/gstudpnetutils.h:
	  udp: Fix compiler warning on mingw-w64
	  Fixes: #652144.
	  gstudpnetutils.h:32:0: error: "WINVER" redefined
	  /usr/i686-w64-mingw32/sys-root/mingw/include/_mingw.h:231:0: note: this is the
	  location of the previous definition

2011-06-04 13:49:52 -0700  David Schleef <ds@schleef.org>

	* gst/interleave/interleave.c:
	  interleave: Work around changes in g_atomic API
	  See #651514 for details.

2011-05-18 12:36:40 +0200  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	  rtp: Fix segmentation fault processing payload buffers
	  This commit checks if the value returned by
	  gst_rtp_buffer_get_payload_buffer and
	  gst_rtp_buffer_get_payload_subbuffer is NULL before using it.

2011-05-16 09:04:31 +0200  Pino Toscano <toscano.pino@tiscali.it>

	* ext/pulse/pulseutil.c:
	  pulse: Define PATH_MAX if it isn't defined
	  GNU Hurd for example doesn't define it.

2011-04-29 08:55:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Allow setcaps to be called after a format was negotiated if it's compatible
	  Otherwise wavenc will fail if upstream decides to set equivalent caps or caps
	  with additional information later.
	  Thanks to Alexander Schremmer for finding this bug.

2011-06-15 15:06:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* REQUIREMENTS:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* ext/Makefile.am:
	* ext/esd/Makefile.am:
	* ext/esd/esdmon.c:
	* ext/esd/esdmon.h:
	* ext/esd/esdsink.c:
	* ext/esd/esdsink.h:
	* ext/esd/gstesd.c:
	* gst-plugins-good.spec.in:
	* m4/Makefile.am:
	* m4/as-arts.m4:
	* m4/esd.m4:
	* po/POTFILES.in:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Remove esound/esdsink plugin

2011-06-15 14:37:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	* REQUIREMENTS:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* ext/Makefile.am:
	* ext/gconf/Makefile.am:
	* ext/gconf/gstgconf.c:
	* ext/gconf/gstgconf.h:
	* ext/gconf/gstgconfaudiosink.c:
	* ext/gconf/gstgconfaudiosink.h:
	* ext/gconf/gstgconfaudiosrc.c:
	* ext/gconf/gstgconfaudiosrc.h:
	* ext/gconf/gstgconfelements.c:
	* ext/gconf/gstgconfelements.h:
	* ext/gconf/gstgconfvideosink.c:
	* ext/gconf/gstgconfvideosink.h:
	* ext/gconf/gstgconfvideosrc.c:
	* ext/gconf/gstgconfvideosrc.h:
	* ext/gconf/gstswitchsink.c:
	* ext/gconf/gstswitchsink.h:
	* ext/gconf/gstswitchsrc.c:
	* ext/gconf/gstswitchsrc.h:
	* gconf/.gitignore:
	* gconf/Makefile.am:
	* gconf/gstreamer.schemas.in:
	* gst-plugins-good.spec.in:
	* m4/Makefile.am:
	* m4/gconf-2.m4:
	* po/POTFILES.in:
	* tests/check/Makefile.am:
	  Remove gconf elements and plugin
	  GConf was deprecated in favour of GSettings etc.

2011-06-15 15:17:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix unitialized access

2011-06-09 21:06:28 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-read-common.c:
	  matroska: add missing stdio include for sscanf

2011-06-13 19:08:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-06-13 17:51:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/audiopanorama.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  -good: port some more plugins

2011-06-13 17:14:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: fix for flush_stop API change

2011-06-13 17:14:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	  rtp: port some more (de)payloader

2011-06-13 17:05:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparsers: not so greedy minimum frame size
	  ... which will be determined by parsing anyway, and avoids introducing
	  redundant additional latency.

2011-06-13 16:33:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsrc.c:
	  -good: update for buffer API change

2011-06-13 16:33:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	  rtp: port to 0.11

2011-06-13 13:25:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpdepay.h:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvpay.c:
	  rtp: fix for API changes in the base classes

2011-06-13 13:07:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: use caps event for negotiation

2011-06-13 13:07:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix for flush stop event changes

2011-06-08 18:33:10 +0300  Raimo Järvi <raimo.jarvi@gmail.com>

	* gst/udp/gstudpnetutils.h:
	  udp: Fix compiler warning on mingw-w64
	  Fixes: #652144.
	  gstudpnetutils.h:32:0: error: "WINVER" redefined
	  /usr/i686-w64-mingw32/sys-root/mingw/include/_mingw.h:231:0: note: this is the
	  location of the previous definition

2011-06-11 18:58:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: fix for bufferpool update

2011-06-10 18:05:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: update for alignment change

2011-06-09 17:56:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: port some more

2011-06-09 17:52:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtsp: port to 0.11

2011-06-09 17:50:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udp: port to 0.11

2011-06-09 11:37:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	  aasink: register template and klass correctly

2011-06-09 10:50:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom/gstgoom.c:
	* gst/goom/gstgoom.h:
	  goom: port goom

2011-06-08 18:06:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-06-08 18:05:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	  assink: port aasink to 0.11

2011-06-07 12:06:08 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/debugutils/breakmydata.c:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/gstcapsdebug.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	  debugutils: Switch from GST_BOILERPLATE to G_DEFINE_TYPE

2011-06-07 11:25:18 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videofilter: Use new GstBaseTransform::transform_caps API

2011-06-07 11:23:55 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/auparse/gstauparse.c:
	  auparse: Don't use GST_BOILERPLATE

2011-06-07 11:22:35 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Buffers no longer have caps

2011-06-07 11:20:00 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	  alpha: Use new transform_caps vmethod (with filter)

2011-06-06 20:43:31 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparsers: fix some more parsers

2011-06-06 18:21:04 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_parse_chapters
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-06-06 14:47:27 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_parse_attachments
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-06-06 12:43:14 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_parse_attached_file
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-06-05 22:45:55 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_parse_info
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-06-05 10:15:23 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_parse_metadata
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-06-05 09:54:42 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_parse_metadata_id_tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-06-05 02:24:41 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_parse_metadata_id_simple_tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-06-06 12:42:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: reset state tracking variable when appropriate
	  ... so we don't end up interrupting an operation that should not be interrupted
	  based on the indication of a previous interruptable operation.

2011-06-04 13:49:52 -0700  David Schleef <ds@schleef.org>

	* gst/interleave/interleave.c:
	  interleave: Work around changes in g_atomic API
	  See #651514 for details.

2011-06-04 13:43:00 -0700  David Schleef <ds@schleef.org>

	* ext/soup/gstsouphttpsink.c:
	* ext/soup/gstsouphttpsink.h:
	  souphttpsink: code cleanup

2011-06-05 02:00:08 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: Use ARTIST tag instead of AUTHOR for GST_TAG_ARTIST
	  AUTHOR only existed in an old version of the spec and ARTIST is
	  the new replacement for this. We are still reading both to still
	  be compatible with old files.
	  Fixes bug #644875.

2011-06-02 18:51:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  sys/ximage/ximageutil.c

2011-06-02 18:47:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	  avi: port AVI elements to new API

2011-06-02 13:38:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	  dvdemux: First query the peer duration in the requested format before converting to BYTES
	  Fixes usage of dvdemux after another demuxer, e.g. mxfdemux.
	  Fixes bug #650503.

2011-06-02 10:41:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsink.c:
	  souphttpsink: Fix refcounting of the "session" property
	  Properties should never take ownership of the values
	  passed to them.

2011-06-01 17:04:27 -0700  David Schleef <ds@schleef.org>

	* gst/matroska/matroska-mux.c:
	  matroskamux: For streaming files, push tags first

2011-05-24 14:52:01 -0700  David Schleef <ds@schleef.org>

	* ext/soup/Makefile.am:
	* ext/soup/gstsoup.c:
	* ext/soup/gstsouphttpsink.c:
	* ext/soup/gstsouphttpsink.h:
	* ext/soup/gstsouphttpsrc.c:
	  soup: Add souphttpsink

2011-06-01 10:19:31 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: allow skip-first-bytes of full buffer size

2011-05-30 18:31:50 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following functions to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_parse_header
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-30 12:09:31 +0200  Antonio Frediani <antonio.frediani@inwind.it>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use GST_TAG_IMAGE for coverart too
	  Fixes bug #638107.

2011-05-30 10:40:08 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following functions to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_get_seek_track
	  - gst_matroska_{demux,parse}_reset_streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-28 22:04:34 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska{demux,parse}_found_global_tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-28 10:59:09 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following functions to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_index_seek_find
	  - gst_matroska{demux,parse}_do_index_seek
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-27 23:15:23 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_tracknumber_unique
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-27 20:28:19 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_decode_data
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-27 19:30:48 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_get_length
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-27 09:17:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: Revert 1a90a6c4 and drop Dirac support again
	  It does not work at all (A/V sync issues), is not very useful,
	  other containers work much better with Dirac and Dirac in AVI
	  is not supported by other software.
	  Fixes bug #541215.

2011-05-26 23:35:52 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following functions to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_encoding_cmp
	  - gst_matroska_{demux,parse}_read_track_encodings
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-23 18:06:44 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following functions to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_peek_id_length_pull
	  - gst_matroska_{demux,parse}_peek_id_length_push
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-23 18:06:44 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_peek_adapter
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-26 12:48:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/ximage/ximageutil.c:
	  xvimagesink: Fallback to non-XShm mode if allocating the XShm image failed
	  Fixes bug #630456.

2011-05-26 12:22:52 +0200  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtp/gstrtpmp4vpay.c:
	  rtpmp4vpay: Deprecated send-config property and replace by config-interval
	  Fixes bug #622412.

2010-06-23 11:12:00 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: UTF-8 subtitles may have markup
	  Fixes #616936.

2011-01-23 15:56:49 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttextoverlay.h:
	  cairotextoverlay: forward new segment events from the sink to the source
	  Not doing so will cause buffers to be received by downstream without
	  a time base set.
	  We use the same method avimux uses to get access to the event when
	  collectpads got the sink event function.
	  https://bugzilla.gnome.org/show_bug.cgi?id=640323

2011-01-24 11:11:48 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	  textoverlay: forward source events to sinks
	  Events are passed to the video sink, and to the text sink if it is
	  linked.
	  This will allow seeking, for instance.
	  https://bugzilla.gnome.org/show_bug.cgi?id=586450

2011-05-25 21:12:12 +0200  David Hoyt <dhoyt@llnl.gov>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: Add property to assume a single stream and emit no-more-pads
	  Fixes bug #616686.

2011-05-25 14:50:26 +0200  Miguel Angel Cabrera Moya <madmac2501@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: uniform unknown message handling
	  Do the same processing in all the cases when an unknown message is received.
	  That is, give a warning.
	  https://bugzilla.gnome.org/show_bug.cgi?id=651059

2011-05-23 18:06:44 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_peek_pull
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-23 18:06:44 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following function to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_peek_bytes
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-23 18:06:44 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following functions to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_{demux,parse}_encoding_order_unique
	  - gst_matroska_{demux,parse}_read_track_encoding
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-24 18:27:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	  autodetect: port to new API

2011-05-24 17:34:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/avi/gstavidemux.c
	  gst/rtp/gstrtpac3depay.c
	  gst/rtp/gstrtpg726depay.c
	  gst/rtp/gstrtpmpvdepay.c
	  gst/videofilter/gstgamma.c

2011-05-24 13:12:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtppcmudepay.c:
	  pcmudepay: allow variable sample rate

2011-05-24 13:11:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtppcmadepay.c:
	  pcmadepay: allow variable sample rate

2010-04-04 06:43:41 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: add norm property
	  Based on a patch by Guennadi Liakhovetski.
	  v2: updates because I forgot to add GstTuner interface to v4l2sink
	  v3: update to add all possible values to norm enum

2011-05-23 20:46:04 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: fixed copyright headers
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-23 18:06:44 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Move the following functions to matroska-read-common.[ch] from
	  matroska-demux.c and matroska-parse.c:
	  - gst_matroska_decode_content_encodings
	  - gst_matroska_decompress_data
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-23 18:48:57 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.h:
	  matroska: move GstMatroska{Demux,Parse}::state to GstMatroskaReadCommon
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-24 09:48:56 +0200  Jonas Larsson <jonas.larsson@hiq.se>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix buffer leak with corrupted files
	  Fixes bug #650912.

2011-05-23 02:46:38 -0700  Miguel Angel Cabrera Moya <madmac2501@gmail.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix parameter type in trace
	  https://bugzilla.gnome.org/show_bug.cgi?id=650937

2011-05-23 18:06:44 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/Makefile.am:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroska: refactor code common to matroskademux and matroskaparse
	  Replace the following functions with their gst_matroska_read_common_*
	  counterparts:
	  - gst_matroska_{demux,parse}_parse_index
	  - gst_matroska_{demux,parse}_parse_skip
	  - gst_matroska_{demux,parse}_stream_from_num
	  Introduce GstMatroskaReadCommon to contain those members of
	  GstMatroskaDemux and GstMatroskaParse that were used by the above
	  functions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650877

2011-05-23 13:50:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: tell baseparse the duration in samples for better accuracy
	  Tell GstBaseParse the duration in samples instead of time, so that
	  a duration query in DEFAULT format will return the correct number
	  of samples without rounding errors. Baseparse will convert this
	  into time itself when needed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650785

2011-05-23 13:25:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: also try upstream first for duration query in DEFAULT format
	  https://bugzilla.gnome.org/show_bug.cgi?id=650785

2011-05-23 13:23:21 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: make conversion from TIME to DEFAULT format (samples) work
	  Fix copy'n'paste error in the previous commit.

2011-05-23 11:36:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Implement conversions between TIME and DEFAULT format
	  Fixes bug #650785.

2011-05-22 18:50:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: don't error out on invalid minimum_blocksize value in streaminfo header
	  We don't use it, so may just as well accept an invalid value
	  of 0 here, which is likely inconsequential anyway.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650691

2011-05-20 10:34:47 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	  rtp: fix static array overruns in a nicer way
	  Use G_N_ELEMENTS instead of hard-coding the array size.

2011-05-20 00:53:44 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	  rtp: fix static array overruns
	  Yes array[10] has elements from 0...9.

2011-05-19 23:31:19 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	  docs: update plugin introspection data
	  Now more files are merged and produced in a canonical fashion, which hopefully
	  creates less or no delta in the future.

2011-05-19 22:57:15 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 9e5bbd5 to 69b981f

2011-05-19 18:21:33 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add missing break

2010-11-08 14:06:15 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Add support for deinterlacing using buffer caps/flags
	  When not using the fieldanalysis element immediately upstream of deinterlace,
	  behaviour should remain unchanged. fieldanalysis will set the caps and flags on
	  the buffers such that they can be interpreted and acted upon to produce
	  progressive output.
	  There are two main modes of operation:
	  - Passive pattern locking
	  Passive pattern locking is a non-blocking, low-latency mode of operation that
	  is suitable for close-to-live usage. Initially a telecine stream will be
	  output as variable framerate with naïve timestamp adjustment. With each
	  incoming buffer, an attempt is made to lock onto a pattern. When a lock is
	  obtained, the src pad and output buffer caps will reflect the pattern and
	  timestamps will be accurately interpolated between pattern repeats. This
	  means that initially and at pattern transitions there will be short periods
	  of inaccurate timestamping.
	  - Active pattern locking
	  Active pattern locking is a blocking, high-latency mode of operation that is
	  targeted at use-cases where timestamp accuracy is paramount. Buffers will be
	  queued until enough are present to make a lock. When locked, timestamps will
	  be accurately interpolated between pattern repeats. Orphan fields can be
	  dropped or deinterlaced. If no lock can be obtained, a single field might be
	  pushed through to be deinterlaced.
	  Locking can also be disabled or 'auto' chooses between passive and active
	  locking modes depending on whether upstream is live.

2011-05-10 16:25:40 -0700  David Schleef <ds@schleef.org>

	* configure.ac:
	  configure: Remove config script check for caca

2011-05-18 12:36:40 +0200  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	  rtp: Fix segmentation fault processing payload buffers
	  This commit checks if the value returned by
	  gst_rtp_buffer_get_payload_buffer and
	  gst_rtp_buffer_get_payload_subbuffer is NULL before using it.

2011-05-18 14:49:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/Makefile.am:
	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: Post CODEC and BITRATE tags
	  Also filter any CODEC/AUDIO_CODEC tags from incoming
	  tag events.
	  Fixes bug #391543.

2011-05-18 16:10:07 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From fd35073 to 9e5bbd5

2011-05-18 12:52:31 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: ensure 0-padding when correcting dubious list size

2011-05-18 12:24:25 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 46dfcea to fd35073

2011-05-18 10:22:27 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: use EINVAL for missing url parameter
	  Fixes gcc warning about using uninitialized variable 'res'.

2011-04-28 15:37:40 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/debugutils/rndbuffersize.c:
	* gst/videofilter/gstgamma.c:
	  various: fix author tag in element details

2011-04-20 15:25:58 -0400  Chris E Jones <chris@chrisejones.com>

	* gst/auparse/gstauparse.c:
	  auparse: implement seeking
	  Implement seeking and seeking query. Fixes #644512

2011-05-17 16:13:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-04-06 16:05:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: also allow PAUSE to be interrupted
	  ... as it is on the way out to NULL.
	  See #632504.

2011-04-06 15:51:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: ensure proper closing and cleanup
	  ... since the TEARDOWN sequence might not have had a chance to even start,
	  but at least connections should be closed (synchronously) and state cleaned up.
	  See #632504.

2011-04-06 15:49:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: fix and improve async handling
	  Simplify the command handling; passing a command to thread means we really
	  want it to get the message, which means to always flush provided the command
	  can handle being interrupted.  Command thread indicates whether command
	  allows interruption and ensure non-flushing connection as it subsequently
	  needs it.
	  In particular, this also makes the TEARDOWN sequence interruptable
	  and also prevents races where _loop_ could miss a command and would
	  continue receiving (or at least trying to).
	  See #632504.

2011-04-06 14:53:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: tweak post-seek loop handling

2011-01-10 12:46:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: open on play and pause when not done yet
	  With the async state changes, it is possible that we need to open the stream
	  before play and pause.
	  Also make sure we remember a previous open failure so that we don't keep trying
	  again.

2011-01-10 11:45:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: improve async handling
	  Simplify the command handling, only continue looping when we have not received
	  another command or when the previous loop was successfull.
	  Avoid looping on a disconnected socket.

2011-01-07 18:02:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: rework reconnect code
	  Use the same async code path to implement reconnects.
	  Make sure we only post progress messages when doing async things.

2011-01-07 17:19:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: small cleanups
	  Make sure we cancel the previous task when queuing a new one.
	  Move the messages to a central place so we can more easily post them.

2011-01-07 15:15:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't post errors when interrupting

2011-01-07 13:43:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: implement more async handling
	  Remove some old locks.
	  Make sure we never go into the loop function when flushing.

2011-01-07 11:40:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: first attempt at async implementation

2011-01-07 11:40:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: small header cleanups

2011-05-17 10:47:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  ssrcdemux: Fix uninitialized variable compiler warning for (pre-) releases too

2011-04-28 15:57:04 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2objects: Only allow mpeg-ts on source objects
	  Ugly fix for #648312

2011-05-17 09:24:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Fix uninitialized variable compiler warning

2011-05-06 19:09:17 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  ssrcdemux: Implement iterate internal links for sink pads
	  https://bugzilla.gnome.org/show_bug.cgi?id=649617

2011-05-06 18:41:01 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: iterate pad function is only valid for src pads
	  The iterate function is only used for src pads, so mark it as such and remove
	  dead code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=649617

2011-05-06 18:12:53 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Release lock before emitting signal
	  If the lock is not released before emitting a signal, it may cause a deadlock
	  if any other function in the element is called.
	  Also removed an unused timestamp parameter
	  https://bugzilla.gnome.org/show_bug.cgi?id=649617

2011-05-15 23:25:15 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: calculate segment duration after parsing all the IDs
	  Since the segment duration is given in terms of the
	  GST_MATROSKA_ID_TIMECODESCALE we should only convert it into
	  nanoseconds when we are sure that any scale specified in the file has
	  been read.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650258

2011-05-16 17:52:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  configure.ac

2011-05-16 17:50:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	  -good: fix for new API

2011-05-04 11:55:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: additional lock safety
	  Fixes #619590.

2011-04-26 16:06:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: also check for bitrate info in caps

2010-05-25 01:04:43 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: guess bitrate if only one stream's bitrate is unknown
	  If the bitrates for all but one audio/video streams are known, and the
	  total stream size and duration can be determined, this calculates the
	  unkown bitrate as (stream size / duration) - (sum of known bitrates).
	  While this is not guaranteed to be very accurate, it should be good
	  enough for most purposes.
	  For example, this is useful for H.263 + AAC streams where no 'btrt' atom
	  is available for the video portion.
	  https://bugzilla.gnome.org/show_bug.cgi?id=619548

2010-05-31 23:59:59 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Export max bitrate for AMR-NB/-WB streams
	  This parses the 'damr' atom if present, and exports the maximum bitrate
	  of the stream using the mode set field to determine the highest bitrate
	  frame type that might be present.
	  https://bugzilla.gnome.org/show_bug.cgi?id=620186

2011-05-16 09:04:31 +0200  Pino Toscano <toscano.pino@tiscali.it>

	* ext/pulse/pulseutil.c:
	  pulse: Define PATH_MAX if it isn't defined
	  GNU Hurd for example doesn't define it.

2011-05-15 23:25:15 +0300  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: calculate segment duration after parsing all the IDs
	  Since the segment duration is given in terms of the
	  GST_MATROSKA_ID_TIMECODESCALE we should only convert it into
	  nanoseconds when we are sure that any scale specified in the file has
	  been read.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650258

2011-05-09 19:00:45 +0200  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Add support for mpegversion 2, which is also AAC

2011-05-11 10:25:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: Send EOS when seeking after the end of file instead of failing
	  Fixes bug #649780.

2011-04-29 08:59:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Set fixedcaps getcaps function on the sinkpad
	  wavenc does not allow to change the caps during playback
	  and always returning the template caps is just wrong.

2011-04-29 08:55:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Allow setcaps to be called after a format was negotiated if it's compatible
	  Otherwise wavenc will fail if upstream decides to set equivalent caps or caps
	  with additional information later.
	  Thanks to Alexander Schremmer for finding this bug.

2011-05-14 10:02:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development

=== release 0.10.29 ===

2011-05-10 10:04:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	* win32/common/config.h:
	  Release 0.10.29
	  Highlights:
	  - amrparse, aacparse, ac3parse, flacparse, mpegaudioparse, dcaparse audio parsers (moved from -bad)
	  - muxers now mux based on running time
	  - ISO MP4 muxers: mp4mux/3gppmux/qtmux/mj2mux (moved from -bad)
	  - new matroskaparse element
	  - new v4l2radio element
	  - rtpsession: support RTCP Early Feedback (the AVPF profile)
	  - orc 0.4.14 or newer recommended
	  - many other fixes and improvements

2011-05-05 13:24:23 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix signed floating point values writing
	  You would end up on some architectures with 0 being written out
	  instead of the proper value.
	  https://bugzilla.gnome.org/show_bug.cgi?id=649449

2011-05-04 12:04:15 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: avoid building index when streamable
	  ... as it will not be written anyway.
	  Fixes #648937 (?).

2011-05-02 12:09:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	  build: add old qtdemux/quicktime directories to CRUFT_DIRS and CRUFT_FILES

2011-05-01 00:04:03 -0400  Tom Janiszewski <tom.janiszewski@alcatel-lucent.com>

	* gst/flv/gstflvmux.c:
	  flvmux: don't overwrite metadata tag with duration in streaming mode
	  A duration tag gets inserted only for streamable=false, so only
	  update/write the duration later if we actually inserted that tag,
	  otherwise we write garbage into other tags.
	  https://bugzilla.gnome.org/show_bug.cgi?id=649060

2011-04-30 18:16:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* po/fr.po:
	* win32/common/config.h:
	  0.10.28.4 pre-release

2011-04-30 17:46:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Android.mk:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* gst-plugins-good.spec.in:
	* gst/isomp4/LEGAL:
	* gst/isomp4/Makefile.am:
	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/atomsrecovery.h:
	* gst/isomp4/descriptors.c:
	* gst/isomp4/descriptors.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/ftypcc.h:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/gstqtmoovrecover.h:
	* gst/isomp4/gstqtmux-doc.c:
	* gst/isomp4/gstqtmux-doc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/gstqtmuxmap.h:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/gstrtpxqtdepay.h:
	* gst/isomp4/isomp4-plugin.c:
	* gst/isomp4/properties.c:
	* gst/isomp4/properties.h:
	* gst/isomp4/qtatomparser.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux.vcproj:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_lang.c:
	* gst/isomp4/qtdemux_lang.h:
	* gst/isomp4/qtdemux_types.c:
	* gst/isomp4/qtdemux_types.h:
	* gst/isomp4/qtpalette.h:
	* po/POTFILES.in:
	  quicktime: rename plugin to isomp4
	  https://bugzilla.gnome.org/show_bug.cgi?id=648004

2011-04-29 17:55:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	  audioparsers: fix some parsers

2011-04-29 17:54:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  fix error caused by merging

2011-04-29 15:49:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  configure.ac
	  gst/rtp/gstrtpgstpay.c

2011-04-29 15:46:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofx: fix pad_alloc

2011-04-27 12:45:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* po/bg.po:
	* po/ja.po:
	* po/nl.po:
	* po/ru.po:
	* win32/common/config.h:
	  0.10.28.3 pre-release

2011-04-26 15:58:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: fix buffer leak

2011-04-26 15:58:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: fix buffer leak

2011-04-26 15:42:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: port jack elements

2011-04-25 10:04:52 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: documentation typo "jpegddec"
	  https://bugzilla.gnome.org/show_bug.cgi?id=648589

2011-04-25 18:14:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pdepay.c:
	  rtp: port some more elements

2011-04-25 17:27:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	  rtp: port more to 0.11

2011-04-25 13:16:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	  rtp: port some more (de)payloaders

2011-04-25 12:49:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	  port some more elements to 0.11

2011-04-25 11:38:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-04-24 16:45:07 -0700  David Schleef <ds@schleef.org>

	* gst/avi/gstavimux.c:
	* gst/matroska/matroska-mux.c:
	  avimux,matroskamux: Add stream-format to h264 caps
	  Fixes #606662.

2011-02-20 12:13:49 -0800  David Schleef <ds@schleef.org>

	* ext/libpng/gstpngdec.c:
	  pngdec: Remove temporary code
	  Now that we depend on (what will be) -base-0.10.33.

2011-04-24 14:03:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: don't pass -Waddress to ObjC compiler on OSX when compiling osxvideosink
	  Temporary workaround until we fix this properly and check for
	  the ObjC warning/error flags instead of just passing CFLAGS to the
	  ObjC compiler.
	  https://bugzilla.gnome.org/show_bug.cgi?id=643939

2011-04-24 13:29:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/inspect/plugin-quicktime.xml:
	* gst-plugins-good.spec.in:
	* gst/quicktime/Makefile.am:
	  quicktime: rename plugin filename from *qtdemux* to *quicktime*
	  https://bugzilla.gnome.org/show_bug.cgi?id=648004

2011-04-24 14:03:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From c3cafe1 to 46dfcea

2011-04-21 23:30:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/quicktime/Makefile.am:
	* gst/quicktime/gstqtmoovrecover.c:
	* gst/quicktime/gstqtmux-doc.c:
	* gst/quicktime/gstqtmux-doc.h:
	  docs: add various qtmux variants to documentation

2011-04-21 22:51:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	* gst/quicktime/gstqtmuxmap.h:
	  quicktime: register 3gppmux element in addition to the misnamed gppmux

2011-04-18 18:08:30 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Remove incomplete support for RTCP FIR
	  Remove bits that were meant to suppport RTCP FIR
	  https://bugzilla.gnome.org/show_bug.cgi?id=648160

2011-04-19 18:55:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	  flac: port to 0.11

2011-04-19 17:35:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	  use G_DEFINE_TYPE some more

2011-04-19 17:20:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	  avi: use G_DEFINE_TYPE

2011-04-19 17:07:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsemixer.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	  use G_DEFINE_TYPE

2011-04-19 16:25:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-04-19 14:33:25 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/generic/.gitignore:
	* tests/check/generic/index.c:
	  tests: add generic set_index test

2011-04-19 14:33:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix deadlock on setting index on flvdemux

2011-04-19 14:16:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/flacparse.c:
	  tests: add index-setting test for baseparse/flacparse
	  https://bugzilla.gnome.org/show_bug.cgi?id=646811

2011-04-18 11:29:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/pipelines/wavpack.c:
	  wavpack: Remove bus GSource to prevent a valgrind warning

2011-04-18 11:14:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/pipelines/wavenc.c:
	  wavenc: Remove bus GSource to prevent a valgrind warning

2011-04-18 11:11:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  tagschecking: Remove bus GSource to prevent a valgrind warning

2011-04-18 11:10:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/imagefreeze.c:
	  imagefreeze: Remove bus GSource to prevent a valgrind warning

2011-04-18 10:54:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiofx/audiopanorama.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	  port more plugins to 0.11

2011-04-18 10:23:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  android/apetag.mk
	  android/avi.mk
	  android/flv.mk
	  android/icydemux.mk
	  android/id3demux.mk
	  android/qtdemux.mk
	  android/rtp.mk
	  android/rtpmanager.mk
	  android/rtsp.mk
	  android/soup.mk
	  android/udp.mk
	  android/wavenc.mk
	  android/wavparse.mk
	  configure.ac

2011-04-17 01:29:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix 'variable may be used uninitialized' warnings caused by -DG_DISABLE_ASSERT

2011-04-16 18:50:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	* win32/common/gstrtpbin-marshal.c:
	* win32/common/gstrtpbin-marshal.h:
	  0.10.28.2 pre-release

2011-04-16 18:49:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	  gst: update disted orc backup code

2011-04-16 18:29:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update for pre-release

2011-04-16 18:27:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/bg.po:
	* po/cs.po:
	* po/de.po:
	* po/es.po:
	* po/id.po:
	* po/sl.po:
	  po: update translations

2011-04-16 18:17:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: refuse incomplete legacy h264 caps
	  Refuse h264 caps without stream-format and codec_data fields for
	  now, to avoid creating broken files. This might cause some pipelines
	  that worked previously to fail. However, the move from -bad to -good
	  is our only chance to fix this up, so make it strict for now. We can
	  always change it back to be less strict in future.
	  https://bugzilla.gnome.org/show_bug.cgi?id=647919

2011-04-16 18:16:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: fix another unused-but-set-variable warning

2011-04-16 18:10:24 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/speex/gstspeexenc.c:
	* gst/rtp/gstrtpgsmpay.c:
	  pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling
	  Don't use g_assert() for error handling, even if they're highly unlikely.
	  Either we *know* that something can't happen, in which case we
	  should just not handle it, or we think something can happen, but it is
	  very very unlikely that it will ever happen, in which case we should
	  handle it like any other error instead of asserting.
	  g_assert() is best left for conditions we have control of, like checking
	  internal consistency of our code, not checking return values of external
	  code.
	  Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
	  gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
	  gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
	  gstspeexenc.c: In function 'gst_speex_enc_encode':
	  gstspeexenc.c:904:19: warning: variable 'written' set but not used
	  pulsesink.c: In function 'gst_pulsesink_change_state':
	  pulsesink.c:2725:9: warning: variable 'res' set but not used
	  pulsesrc.c: In function 'gst_pulsesrc_change_state':
	  pulsesrc.c:1253:7: warning: variable 'e' set but not used

2011-04-16 18:07:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/rtp/server-alsasrc-PCMA.c:
	  examples: fix some warnings in rtp example
	  Caused by -DG_DISABLE_ASSERT

2011-04-16 17:57:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/level/level-example.c:
	  examples: don't put code with side-effects into g_assert()
	  Otherwise things won't work too well when compiling with
	  -DG_DISABLE_ASSERT (as we do for pre-releases and releases).

2011-04-16 16:51:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/tvtime/greedyh.c:
	* gst/matroska/matroska-mux.c:
	  deinterlace, matroska: fix two variable-may-be-used-uninitialized compiler warnings
	  We use -DG_DISABLE_ASSERT for the pre-releases, which makes these
	  warnings pop up in cases that were previously covered by g_assert_not_reached()
	  and the like:
	  tvtime/greedyh.c:801:14: warning: 'scanline' may be used uninitialized in this function
	  matroska-mux.c:501:19: warning: 'context' may be used uninitialized in this function

2011-04-16 14:45:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/apetag/gstapedemux.c:
	  apedemux: Port to 0.11

2011-04-16 13:33:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: fix unused-but-set-variable warnings with gcc-4.6

2011-04-16 13:23:50 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/cairo/cairo_overlay.c:
	  examples: fix 'control reaches end of non-void function' warning in cairo example

2011-04-15 15:47:24 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Address unused but set variable
	  The v4l2object formats list was being obtained into a local variable and
	  then still used from the context. Make use of the local variable.

2011-04-15 15:17:34 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* sys/oss4/oss4-mixer-slider.c:
	* sys/oss4/oss4-mixer-switch.c:
	* sys/oss4/oss4-property-probe.c:
	* sys/oss4/oss4-source.c:
	  oss4: Address unused but set variables
	  GCC 4.6.x complains about such variable usage. Unused but set variables
	  were removed except that gst_oss4_mixer_slider_set_mute () now returns
	  the value from the call to gst_oss4_mixer_set_control_val ().

2011-04-15 15:14:13 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* ext/pulse/pulsesink.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	  jpegenc: pulsesink: raw1394: Address unused but set variables
	  GCC 4.6.x spits warnings about such usage of variables. The variables in
	  raw1394 were marked with G_GNUC_UNUSED as this seemed omre appropriate.
	  The others were removed.

2011-04-15 15:12:44 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	* gst/y4m/gsty4mencode.c:
	  y4mencode: shapewipe: Address unused but set variables
	  GCC 4.6.x complains about such usage.

2011-04-15 15:11:35 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/pipelines/flacdec.c:
	* tests/examples/level/level-example.c:
	* tests/icles/videocrop-test.c:
	* tests/icles/ximagesrc-test.c:
	  tests: Address unused but set variables
	  GCC 4.6.x spits warnings about such usage of variables.

2011-04-15 15:36:41 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/videomixer/blendorc.orc:
	  videomixer: Fix argb/rgba overlay orc code
	  Remove some redundant operations (convubw) and use the correct variable,
	  t2, in the orc_overlay_bgra function.

2011-04-15 15:33:35 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/gstcollectpads2.c:
	* gst/videomixer/videomixer2.c:
	  videomixer: address unused but set variables
	  GCC 4.6.x spits warnings about variables that are set but unused. Such
	  variables have been removed in blend, collectpads2 and videomixer2.

2011-04-15 14:57:20 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtp, rtpmanager: Address unused but set variables
	  GCC 4.6.x spits warnings about variables that are unused but set. Such
	  variables have been removed where trivial but with comments left behind
	  for informational purposes in some cases.
	  gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
	  to always return GST_FLOW_OK instead of the return value of
	  rtp_session_process_rtcp (), so we'll keep it that way.

2011-04-15 11:29:30 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/quicktime/descriptors.c:
	* gst/quicktime/gstrtpxqtdepay.c:
	* gst/quicktime/qtdemux.c:
	  quicktime: Remove unused but set variables
	  GCC 4.6.x spits warnings about such variable usage. Note that some
	  calculations are left as comments for informative purposes.

2011-04-15 11:23:38 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	  matroska: Remove unused but set variables
	  GCC 4.6.x spits warnings about such variable usage.

2011-04-15 11:19:26 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Remove unused but set duration variable
	  GCC 4.6.x spits warnings about such variable usage.

2011-04-15 11:18:19 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flxdemux: Remove unused but set keyframe variables
	  The FIXMEs about the keyframe flag never being used are left for later
	  fixing, at which point the keyframe variables could be added back.

2011-04-15 11:16:42 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/effectv/gstedge.c:
	  edgetv: Remove unused but set height variable
	  GCC 4.6.x spits warnings about such variables.

2011-04-15 18:51:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: update for gst_base_parse_frame_init() API change

2011-02-01 15:57:01 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Use existing functions to parse RTCP FB packets
	  Use existing functions to get the FCI from FB packets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=622553

2011-02-01 16:23:52 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/rtpsession.c:
	  rtpsession: marshal GstBuffer as a MiniObject instead of a pointer
	  https://bugzilla.gnome.org/show_bug.cgi?id=622553

2011-04-14 23:24:56 -0700  David Schleef <ds@schleef.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Better calculation of framerate
	  https://bugzilla.gnome.org/show_bug.cgi?id=647833

2011-04-13 12:37:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: default to dts-method=reorder and presentation-time=true
	  https://bugzilla.gnome.org/show_bug.cgi?id=636699

2011-04-15 12:47:52 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: test various dts-methods

2011-04-15 12:34:05 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: fix corner case buffer handling for reorder method

2011-04-14 13:47:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't leak the SEEKING query

2011-04-14 13:43:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/quicktime/gstqtmoovrecover.c:
	* gst/quicktime/gstqtmoovrecover.h:
	  qtmoovrecover: Don't leak the static recursive mutex

2011-04-14 13:37:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2radio.c:
	  v4l2radio: Free videodev string before replacing it

2011-04-14 13:24:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: Allow webm and matroska caps and don't leak caps

2011-04-14 07:35:29 +0100  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>

	* gst-plugins-good.spec.in:
	  Add parser plugin

2011-04-13 21:58:36 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/Makefile.am:
	* gst/dtmf/gstdtmfcommon.h:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  dtmf: Move duplicate #defines into a common include
	  Centralize duplicated constants so they have the same value.
	  Also standardise minimum tone duration to 250ms and minimum inter-tone
	  interval to 100ms.

2011-03-24 14:34:24 -0700  David Schleef <ds@entropywave.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Add conditionals on WAVE_FORMAT_DOLBY_AC3_SPDIF

2011-04-11 20:09:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/debugutils/gstcapsdebug.c:
	  capsdebug: fix unused-but-set-variable warnings with gcc 4.6

2011-04-11 20:05:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix unused-but-set-variable warning with gcc 4.6
	  Most likely a leftover from when the index parsing code was rewritten.

2011-04-11 19:54:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: fix unused-but-set-variable warning with gcc 4.6

2011-04-11 19:50:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: fix handling of YUV images with 'odd' widths
	  Fixes unused-but-set-variable warnings with gcc 4.6.

2011-04-11 19:49:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: fix unused-but-set-variable warnings with gcc 4.6

2011-04-13 18:11:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	  audiowsinc{band,limit}: Fix check for divison by zero

2011-04-13 18:01:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiowsincband.c:
	  audiowsincband: Fix range of kernel elements (lim -> lim-1)

2011-04-13 18:00:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiowsinclimit.c:
	  audiowsinclimit: Add some more braces to make the code more readable

2011-04-11 18:40:30 -0500  Jordi Burguet-Castell <jordi.burguet-castell@ligo.org>

	* gst/audiofx/audiowsinclimit.c:
	  audiowsinclimit: Fix range of kernel elements (lim -> lim-1) in high/low-pass filters

2011-04-13 17:49:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiowsincband.c:
	  audiowsincband: Add new windowing functions: gaussian, cos and hann

2011-04-11 18:41:43 -0500  Jordi Burguet-Castell <jordi.burguet-castell@ligo.org>

	* gst/audiofx/audiowsinclimit.c:
	  audiowsinclimimt: Add new windows to high/low-pass filters: gaussian, cosine, hann

2011-04-13 16:47:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: set stream-format=byte-stream on h264 caps if there's no codec data
	  https://bugzilla.gnome.org/show_bug.cgi?id=606662

2011-04-13 16:37:07 +0100  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: restrict h264 some more to only accept AU-aligned AVC
	  https://bugzilla.gnome.org/show_bug.cgi?id=606662

2011-04-13 17:11:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: The VBRI header is always at offset 0x20, independent of MPEG version
	  Also clean up advancing of the data pointer a bit.
	  Fixes bug #647659.

2011-04-13 15:18:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	* tests/check/Makefile.am:
	* tests/check/elements/qtmux.c:
	  qtmux: add variant-less video/quicktime to source pad template caps
	  This is needed for automatic transcoding using encodebin. Our typefinder
	  does not always add a variant to the found caps, and encodebin needs
	  an *exact* match to the caps on the source pad template, so we need
	  to add the variant-less video/quicktime caps to the template as well
	  for encodebin to be able to find it. Add unit test for this as well.
	  https://bugzilla.gnome.org/show_bug.cgi?id=642879

2011-04-13 16:17:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: Properly interprete the result of strcmp()

2011-04-13 16:09:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: Don't store image tags inside the vorbiscomments and the flac metadata
	  Instead only store them inside the flac metadata. There's
	  no point in storing them twice and the flac metadata is
	  still the official way to store image tags inside flac.

2011-04-13 12:38:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/.gitignore:
	* tests/check/pipelines/.gitignore:
	  tests: ignore new qtmux-related test binaries

2011-04-13 11:25:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* gst/quicktime/Makefile.am:
	* gst/quicktime/gstqtmuxplugin.c:
	* gst/quicktime/quicktime.c:
	* tests/check/Makefile.am:
	  quicktime: move qtmux plugin from -bad to -good
	  https://bugzilla.gnome.org/show_bug.cgi?id=636699

2011-04-12 16:42:17 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: Remove leftover MAEMO_BROKEN defines
	  Remove defines to work around bugs in old Maemo releases

2011-04-04 12:21:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: more helpful debug error message when no needed duration on input buffers
	  Fixes #646256.

2011-03-21 10:56:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux: Adding GstTagXmpWriter interface
	  Adds GstTagXmpWriter interface support to qtmux

2011-03-22 20:53:08 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: use running time for synchronization
	  See also #432612.

2011-03-10 16:03:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: provide for PTS metadata when so configured
	  ... and not only when sort-of feeling like it.
	  In any case, if it turns out all really is in order,
	  and presumably DTS == PTS, then no ctts will be produced anyway.

2011-03-10 16:02:42 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: also track original PTS buffer timestamp in reorder dts-method

2011-02-21 12:14:59 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  Revert "Check that collectpads exists before removing pad"
	  This reverts commit 6d8740476ccd3a3498dc4f18c19733643825c7b8.
	  Depends on a core commit that was reverted

2011-02-20 23:57:19 -0800  David Schleef <ds@schleef.org>

	* gst/quicktime/gstqtmux.c:
	  Check that collectpads exists before removing pad
	  The core now calls release pad from finalize, at which point
	  the collectpads might have already been freed.

2011-01-13 11:28:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  test: qtmux: Tests qtmux reuse
	  Forces the use of qtmux after it has been put to PLAYING and back
	  to NULL once
	  https://bugzilla.gnome.org/show_bug.cgi?id=639338

2011-01-13 15:27:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: set src pads when starting file
	  ... rather than at _init time, so they are also available following a
	  pad (de)activation cycle.
	  https://bugzilla.gnome.org/show_bug.cgi?id=639338

2011-01-03 17:24:23 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: adjust nasty case timestamp tracking
	  That is, all sorts of problems arise with re-ordered input timestamps that
	  tend to defy automagic handling for every case, so allow for a few variations
	  that can be tried depending on circumstances.
	  Also try to document accordingly.
	  Also fixes #638288.

2010-12-30 21:48:41 +0200  Felipe Contreras <felipe.contreras@nokia.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: get rid of timestamp overprotectiveness
	  Signed-off-by: Felipe Contreras <felipe.contreras@nokia.com>

2011-01-03 16:56:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/atomsrecovery.c:
	* gst/quicktime/gstqtmux.c:
	  qtmux: simplify and fix pts_offset storing
	  In particular, only write a ctts atom if and only if ever a non-zero offset.

2011-01-03 10:43:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: add some more documentation

2010-12-03 15:23:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: remove large-file property
	  Rather, auto-determine if 64-bits fields are needed for a valid result, and
	  stick to plain 32-bits if not needed.
	  API: GstQTMux:large-file (removed)

2010-12-19 12:53:34 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Free AtomInfo structs

2010-12-19 12:50:30 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Free tag string after use

2010-12-19 12:12:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  tagschecking: Fix some more memory leaks

2010-12-17 19:41:25 +0200  Lasse Laukkanen <lasse.laukkanen@digia.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: allow zero duration tracks

2010-12-03 18:09:41 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: add documentation

2010-12-01 10:45:49 +0100  David Hoyt <dhoyt@llnl.gov>

	* gst/quicktime/gstqtmux.c:
	  qtmux: handle msvc ftruncate incompatibility
	  Fixes #636185.

2010-11-27 16:07:19 -0600  Alejandro Gonzalez <agonzalez@dextratech.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: gst_qtmux_check_difference verify before subtract
	  Avoid negative overflow by checking the order of operands
	  on subtraction of unsigned integers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=635878

2010-11-19 17:55:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: remove remnant of obsolete property

2010-11-19 15:18:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: also unit test fragmented file cases

2010-07-30 12:48:29 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: allow specifying trak timescale
	  This is mainly because Smoothstreaming client are broken and don't
	  take the TimeScale property into account.

2010-11-19 17:41:41 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux: include sdtp atoms for ismv fragmented files
	  Based on patch by Marc-André Lureau <mlureau@flumotion.com>

2010-11-19 19:17:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: enable default fragmented file for ismlmux

2010-09-02 13:58:05 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/atoms.h:
	* gst/quicktime/ftypcc.h:
	* gst/quicktime/gstqtmuxmap.c:
	* gst/quicktime/gstqtmuxmap.h:
	  qtmux: add ismlmux, for fragmented isml major brand

2010-11-19 14:44:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: finalize sinkpads list

2010-07-22 19:40:07 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: add moov in streamheader

2010-08-06 13:26:27 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: add streamable property to avoid building fragmented mfra index

2010-11-18 16:48:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: add mfra to fragmented file
	  Based on patch by Marc-André Lureau <mlureau@flumotion.com>

2010-11-15 15:17:59 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: optionally create fragmented file
	  In this mode, an initial empty moov (containing only stream metadata) is written,
	  followed by fragments containing actual data (along with required metadata).
	  New fragments are started either at keyframe (if such are sparse) or when
	  property configured duration exceeded.
	  Based on patch by Marc-André Lureau <mlureau@flumotion.com>
	  Fixes #632911.

2010-11-15 15:12:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	  qtmux: use helper to set atom flags from given uint

2010-11-09 16:49:07 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: refactor configuring and sending of moov
	  Based on patch by Marc-André Lureau <mlureau@flumotion.com>

2010-11-09 15:54:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: refactor extra top-level atom handling
	  Also check a bit more for possible errors, and free proper items in such case.

2010-11-09 15:01:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: refactor slightly using buffer helper

2010-11-05 13:48:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: fix misinforming comment

2010-11-05 12:08:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux: delegate mvex handling to atoms
	  ... which keeps qtmux simpler.

2009-09-28 16:11:35 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux: add mvex/trex in header if fragmented
	  One "trex" is added per "trak". We don't support default values,
	  but the "trex" box is mandatory.

2009-09-28 13:01:30 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/fourcc.h:
	  qtmux: add a couple of fourcc for fragmented mp4

2010-11-05 11:08:01 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: avoid removing temp file when error occurred

2009-09-30 17:16:30 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: truncate buffer file after each send

2009-09-28 16:53:51 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: remove temp file when reset/finalize

2010-10-19 13:43:14 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/quicktime/gstqtmoovrecover.c:
	  various (gst): add missing G_PARAM_STATIC_STRINGS flags
	  Canonicalize property names as needed.

2010-10-13 17:47:29 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: prevent infinite loop when adjusting framerate
	  Fixes #632070.

2010-10-03 23:45:46 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Add G_PARAM_STATIC_STRINGS
	  Add G_PARAM_STATIC_STRINGS to qtmux properties

2010-09-15 17:54:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: Follow xmp serialization guidelines closer
	  qt and isom variants have different ways of serializing
	  xmp, follow these guidelines.
	  Those can be found in Adobe's xmp docs.

2010-08-16 12:36:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: autodetect out-of-order input timestamps and determine DTS accordingly
	  Favour using input buffer timestamps for DTS, but fallback to using buffer
	  duration (accumulation) if input ts detected out-of-order.
	  Fixes #624212.

2010-07-28 16:15:53 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: use caps bitrate at last chance
	  If we didn't get the stream's bitrate from one of the atoms,
	  try getting it from the caps as a last resort.
	  https://bugzilla.gnome.org/show_bug.cgi?id=625496

2010-07-28 16:12:11 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/atoms.c:
	  qtmux: btrt - max bitrate before average
	  According to iso base media file format, the max bitrate
	  is before the avg
	  https://bugzilla.gnome.org/show_bug.cgi?id=625496

2010-07-06 14:48:08 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux: Write 'btrt' atom for H.264 media if possible
	  This writes out the optional 'btrt' atom (MPEG4BitrateBox) for H.264
	  media if either or both of average and maximum bitrate are available for
	  the stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=623678

2010-07-05 14:09:50 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: Write avg/max bitrate to ESDS if available
	  This collects the 'bitrate' and 'maximum-bitrate' tags on the
	  corresponding pad and uses these to populate these fields in the ESDS
	  where applicable.
	  https://bugzilla.gnome.org/show_bug.cgi?id=623678

2010-07-02 12:45:20 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Don't use bogus codec/format tags
	  https://bugzilla.gnome.org/show_bug.cgi?id=623365

2010-06-25 20:19:20 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Write uint tags that don't have a complement
	  Write uint tags that have complements (e.g. track-number/
	  track-count) even when we only have one of them available
	  and set the other one to 0.
	  Fixes #622484

2010-06-21 19:39:54 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Remove the pad from our internal list before calling collectpads
	  Previously we would end up with the collectpaddata structure already freed.
	  This would result in a bogus iteration of mux->sinkpads (all the
	  GstQTPad being freed) and it wouldn't be removed from that list.
	  Finally, due to it not being removed from that list, we would end up
	  calling a bogus gst_qt_mux_pad_reset on those structures => SEGFAULT

2010-05-12 18:50:34 -0700  David Schleef <ds@schleef.org>

	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: Add VP8

2010-05-11 13:15:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  tests: don't fail tagschecking test if qtdemux is not available or too old

2010-03-27 09:46:30 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/quicktime/gstqtmuxplugin.c:
	  qtmux: use GStreamer package name and origin in the plugin info

2010-03-23 17:34:30 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  tests: tagschecking: New tags tests
	  Adds new tags checking tests.

2010-03-25 00:20:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: init debug category before using it

2010-03-22 16:56:03 +0100  Benjamin Otte <otte@redhat.com>

	* gst/quicktime/atoms.c:
	  Add -Wold-style-definition
	  and fix the warnings

2010-03-22 13:16:33 +0100  Benjamin Otte <otte@redhat.com>

	* gst/quicktime/atoms.c:
	* gst/quicktime/gstqtmuxmap.h:
	* tests/check/elements/qtmux.c:
	  Add -Wwrite-strings
	  and fix its warnings

2010-03-21 21:39:18 +0100  Benjamin Otte <otte@redhat.com>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/atomsrecovery.c:
	* gst/quicktime/descriptors.c:
	* tests/check/elements/qtmux.c:
	* tests/check/pipelines/tagschecking.c:
	  Add -Wmissing-declarations -Wmissing-prototypes to configure flags
	  And fix all warnings

2010-03-18 17:30:26 +0100  Benjamin Otte <otte@redhat.com>

	* gst/quicktime/gstqtmoovrecover.c:
	* gst/quicktime/gstqtmux.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple

2010-03-12 11:28:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  tests: tagschecking: Improvements and new geo-location tests
	  Makes some improvements to tagschecking.c, making it use
	  fakesrc instead of videotestsrc and allowing to set input
	  caps so that more muxers can be used. Previously we could
	  only use those that accepted raw video caps.
	  Also adds some tests for geo-location tags

2010-03-12 10:53:36 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Use xmp on mp4mux and gppmux too
	  Do not restrict xmp to qtmux, but use it too
	  on mp4mux and gppmux

2010-03-05 13:33:37 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/pipelines/tagschecking.c:
	  check: tagschecking: tests for tags serialization in muxers
	  Adds a check unit test that aims to test tags serialization
	  and deserialization consistency (in muxers). It provides a
	  basic function that allows one to easily specify tags, a
	  muxer and a demuxer and a test will be done to check if
	  the tags have been consistently muxed and demuxed

2010-02-22 16:45:34 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux: add xmp support
	  Adds xmp metatags adding to qtmux.
	  Fixes #609539

2010-03-11 17:17:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/quicktime/gstqtmoovrecover.c:
	  qtmux: fix GST_ELEMENT_ERROR usage
	  We need to pass (NULL) rather than NULL for empty arguments.

2010-03-10 10:23:23 -0600  Rob Clark <rob@ti.com>

	* gst/quicktime/gstqtmoovrecover.c:
	  qtmux: fix compile error
	  gst/quicktime/gstqtmoovrecover.c:268: warning: format not a string literal and no format arguments
	  https://bugzilla.gnome.org/show_bug.cgi?id=612454

2010-02-22 19:38:15 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: Rename 'avc-sample' to 'avc' in caps
	  Fixes #606662

2010-02-26 11:50:25 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Take lock around use of (non-threadsafe) tagsetter interface.

2010-02-22 16:51:00 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	  qtmux: write all udta children atoms
	  UDTA might have META and other children atoms
	  together, write them all.

2010-02-22 10:48:11 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: Use internal sink pads list
	  Due to GstCollectPads sink pads list being not reliably
	  iteratable (when not inside the collected function) this
	  patch adds a sink pads list to qtmux to be used when iterating
	  sink pads on reset function.
	  Fixes #609055

2010-02-16 17:13:09 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	  qtmux: prevent leaking hdlr name

2010-02-16 16:24:12 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: support for ALAC
	  Fixes #580731.

2010-02-16 14:19:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	  qtmux: refactor building stsd entry 'wave' extension

2010-02-08 11:51:52 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atomsrecovery.c:
	  qtmux: atomsrecovery: Fix compilation problem
	  Fixes a compilation error due to unused function result.

2009-12-12 16:07:15 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/atomsrecovery.c:
	* gst/quicktime/atomsrecovery.h:
	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmoovrecover.c:
	* gst/quicktime/gstqtmoovrecover.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	* gst/quicktime/gstqtmuxplugin.c:
	  qtmux: Adds moov recovery feature
	  Adds a new property to qtmux that sets a path to a file to write
	  and update data about the moov atom (that is not writen till the
	  end of the file). If the pipeline/app crashes during execution it
	  might be possible to recover the movie using the qtmoovrecover element.
	  qtmoovrecover is an element that is also a pipeline. It is not
	  meant to be used with other elements (it has no pads). It is merely
	  a tool/utilitary to recover unfinished qtmux files.
	  Fixes #601576

2010-01-27 19:06:53 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/quicktime/atoms.c:
	  qtmux: for fixed-sample size streams (PCM audio, etc) don't allocate an enormous buffer that we then won't use at all.

2010-01-27 15:37:37 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: handle muxing adpcm correctly.

2010-01-22 13:36:04 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/quicktime/atoms.c:
	  qtmux: Set the mdia hdlr name field to what quicktime uses. Fix writing it since it's not null-terminated. Improves compatibility with some hardware players.

2010-01-22 13:30:07 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: endianness in gstreamer is an int, not boolean.

2010-01-26 17:54:28 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	  qtmux: streamline moov data memory storage
	  In particular, use arrays rather than (double) linked lists.

2010-01-26 13:44:04 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: g_free is NULL safe

2010-01-20 13:30:48 +0100  Benjamin Otte <otte@redhat.com>

	* gst/quicktime/descriptors.c:
	* gst/quicktime/descriptors.h:
	* gst/quicktime/properties.c:
	  [cleanup] Various style and cleanups
	  Various fixes for gtk-doc warnings and making functions without
	  arguments take void as parameter.

2010-01-14 08:09:03 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/gstqtmux.c:
	  qtmux: Actually use new caps info on renegotiation
	  Following the previous qtmux commit, this patch tries
	  to use the new info added to the caps to fill the 'trak'
	  atom's fields and children atoms. This way qtmux will
	  use the late added 'codec_data' when h264parse adds
	  it in the following pipeline:
	  videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \
	  h264parse output-format=0 ! qtmux ! \
	  filesink location=test.mov

2010-01-13 23:33:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/gstqtmux.c:
	  qtmux: Do caps renegotiation when it only adds fields
	  Qtmux can accept caps renegotiation if the new caps is a
	  superset of the old one, meaning upstream added new info to
	  the caps. This patch still doesn't make qtmux update any
	  atoms info from the new info, but at least it doesn't
	  reject the new caps anymore.
	  A pipeline that reproduces this use case is:
	  videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \
	  h264parse output-format=0 ! qtmux ! \
	  filesink location=test.mov

2010-01-13 19:30:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: provide request pads under wider conditions
	  Fixes #606859.

2010-01-13 10:35:00 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: Only accept avc-sample h264
	  qtmux and mp4mux should only accept h264 in avc-sample
	  format

2010-01-11 13:13:41 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  Rename aac's stream-format 'none' to 'raw'
	  Renames aac's stream-format from previous commits from none to
	  raw

2010-01-11 10:34:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: Only accept stream-format='none' aac
	  Only accept raw aac streams (stream-format=none) to avoid
	  generating invalid files.
	  Fixes #604925

2009-12-28 11:34:35 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/quicktime/gstqtmux.h:
	  qtmux: also add .h file changes to unbreak the build

2009-12-27 23:51:50 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/quicktime/gstqtmux.c:
	  qtmux: use correct names from template for request pads
	  The pads where names pad0, pad1, ...

2009-12-27 23:32:58 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/quicktime/gstqtmux.c:
	  qtmux: move errors _new_pad to the end

2009-12-21 13:58:30 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Accept non-paired uint tags
	  Adds support for unpaired unsigned interger tags

2009-12-21 12:05:37 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux: Adds new tags
	  Maps more tags that are already posted by qtdemux
	  Fixes #599759

2009-12-10 22:20:45 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: support more of j2k
	  Reads the new caps added to qtdemux by commit
	  c917d65e6df0b5d585f905c7ad78a8a0a44b2cb0
	  and adds its corresponding atoms.
	  Also adds support for image/x-jpc as it is the same
	  as image/x-jp2, except that the buffers need to be
	  boxed inside a jp2c isom box before muxing. To solve
	  this the QTPads now have a function that (if
	  not NULL) is called when a buffer is collected. This
	  function returns a replacement to the current collected
	  buffer.
	  Fixes #598916

2009-12-10 16:53:19 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: Maps 'classification' tag for 3gpp files
	  Adds the mapping of 'classification' tags to writing of
	  'clsf' atoms for gppmux.
	  Based on a patch by: Lasse Laukkanen <ext-lasse.2.laukkanen@nokia.com>

2009-12-08 17:59:04 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/quicktime/atoms.c:
	* gst/quicktime/gstqtmux.c:
	  qtmux: remove c++ comments and add some more comments.

2009-12-08 17:55:56 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: add ima adpcm support

2009-11-25 21:41:27 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: replace _scale with _scale_round
	  Use the rounding version for improved sync between streams.
	  Small variations in the duration when muxing might lead to
	  cumullative wrong timestamping when demuxing.
	  Fixes #602936

2009-11-24 16:16:56 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: use timestamps for muxing
	  Try to use timestamps even when the stream has out of order
	  timestamps, only fall back to durations when we detect an
	  out of order buffer. Improves sync between streams.

2009-11-19 18:28:52 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: fix missing debug argument
	  Adds a missing debug argument

2009-11-19 11:36:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: fix misinforming debug statement

2009-11-19 11:14:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: ensure writable buffer metadata before setting caps

2009-10-29 08:36:02 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: support for SVQ3
	  Adds support for muxing SVQ3 content. Usually this format
	  has decoder info that must be passed in the 'seqh' field
	  in the caps. It is also good to add the gama atom to make
	  quicktime not crash.
	  Fixes #587922

2009-11-17 09:26:05 -0300  Thiago Sousa Santos <thiagoss@redmoon.(none)>

	* gst/quicktime/gstqtmux.c:
	  qtmux: do not leak a string
	  Frees a string after use. Also does some code organization

2009-11-16 14:57:53 -0300  Thiago Sousa Santos <thiagoss@redmoon.(none)>

	* gst/quicktime/atoms.c:
	  qtmux: do not add size to the pointer variable
	  Do not wrongly add the result of the function to the
	  pointer to the buffer size. Instead, check the result
	  to see if the serialization was ok.
	  Based on a patch by: "Carsten Kroll <car@ximidi.com>"
	  Fixes #602106

2009-11-06 10:34:39 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: handle 'late' streams
	  When muxing streams, some can start later than others. qtmux
	  now handle this by adding an empty edts entry with the
	  duration of the 'lateness' to the stream's trak.
	  It tolerates a stream to be up to 0.1s late.
	  Fixes #586848

2009-11-05 21:35:56 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	  qtmux: adds the EDTS and ELTS atoms to atoms.c
	  These atoms will be useful for signaling streams
	  that start later in the file. As well for adding
	  edit lists if needed sometime later.

2009-11-06 00:46:12 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/gstqtmux.c:
	  qtmux: Adding some ifs for protection
	  Adding somes ifs to protect against warning conditions
	  that might happen when upstream element is not sane
	  Fixes #600895

2009-10-16 10:47:32 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/quicktime/ftypcc.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	* gst/quicktime/gstqtmuxmap.c:
	* gst/quicktime/gstqtmuxmap.h:
	  gppmux: Add support for 3gr6
	  Keep track of the chunk durations to be able to add 3gr6
	  brand if it is a faststart file and the longest chunk is
	  smaller than a sec. Implemented according to 3gpp
	  TS 26.244 v6.4.0 (2005-09)
	  Fixes #584361

2009-10-15 21:11:16 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Only push ftyp later (in faststart mode)
	  In faststart mode, there is no need to send the ftyp
	  right at the beginning of the stream. Waiting and sending it
	  only later (when the moov atom is ready to be sent) provides
	  us with more information about the stream and we can better
	  select the compatible brands.

2009-10-15 17:51:39 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Improve error message
	  Improve error message when we can't get or estimate the
	  timestamp/duration of a buffer

2009-09-29 15:47:13 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/quicktime/atoms.c:
	  qtmux: fix flags_as_uint to flags[]

2009-08-04 12:58:35 +0200  Jan Urbanski <wulczer@wulczer.org>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Don't require endianness field for 8 bit raw audio
	  Fixes bug #590360.

2009-06-25 08:38:21 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/quicktime/atoms.c:
	  qtmux: Remove unused variable.

2009-06-25 08:38:10 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Fix debug statement.

2009-06-11 15:54:42 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmux.h:
	  qtmux: only use (64-bit) extended (mdat) atom size if needed.  Fixes #585319.

2009-06-10 14:46:14 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: set default movie timescale to microsecond units

2009-06-10 13:24:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	  qtmux: compress/optimize stsc writing

2009-06-10 12:42:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: add 3GP style tagging (and refactor appropriately)

2009-06-01 23:00:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux (and variants): handle pixel-aspect-ratio.  Fixes #584358.

2009-06-01 22:42:08 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/ftypcc.h:
	* gst/quicktime/gstqtmuxmap.c:
	  gppmux: enhance ftyp brand heuristic.  Fixes #584360.

2009-05-28 13:56:10 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/fourcc.h:
	* gst/quicktime/gstqtmux.c:
	  qtmux: use different stsd atom type for H263 for ISO and QT variants
	  Fixes #584114.

2009-05-15 01:54:44 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/quicktime/atoms.c:
	  [qtmux] Fixes segfault when adding a blob as first tag.
	  Moves tags data initialization to the function that actually appends
	  the tags to the list. Fixes #582702
	  Also fixes some style caught by the pre-commit hook.

2009-05-10 21:21:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmuxmap.c:
	  gppmux: Add MPEG-4 part 2 to supported formats.  Fixes #581593.

2009-05-07 17:53:42 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  Add ranks to various muxers and encoders in -bad

2009-04-30 14:43:36 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/quicktime/gstqtmuxmap.c:
	  qtmux: changes caps of src pads to video/quicktime, variant=something
	  Take a look at bug #580005 for further info.

2009-04-24 18:53:36 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/quicktime/gstqtmuxmap.c:
	  mp4mux: Changes src caps to application/x-iso-mp4
	  Fixes #580005

2009-03-25 21:24:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: fix reusing element
	  State change to READY and then back to PAUSED should still provide
	  the proper structures as are otherwise freshly available following
	  a request_new_pad.
	  Pointed out by Thiago Santos.

2009-03-23 11:17:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/quicktime/gstqtmux.c:
	  qtmux: fix includes for lseek
	  --

2009-03-20 14:20:16 +0100  LRN <lrn1986 at gmail dot com>

	* gst/quicktime/gstqtmux.c:
	  win32: fix seeking in large files
	  Use _lseeki64() on Windows to seek in large files.
	  Fixes #576021.

2009-03-02 10:57:35 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/quicktime/gstqtmux.c:
	  qtmux: Be a bit more verbose in our debug message when failing to renegotiate

2009-01-28 13:25:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/quicktime/atoms.c:
	* gst/quicktime/atoms.h:
	* gst/quicktime/gstqtmux.c:
	* gst/quicktime/gstqtmuxmap.c:
	  Additional media type support in qtmux (and friends).
	  Support AMR and H263 for both qtmux and gppmux,
	  and add extensions in sample table description.

2009-01-09 21:59:48 +0000  David Schleef <ds@schleef.org>

	  gst/quicktime/gstqtmuxmap.c: Add video/x-qt-part and video/x-m4-part to caps so schroenc/schroparse can use it.  Fixes #5...
	  Original commit message from CVS:
	  * gst/quicktime/gstqtmuxmap.c: Add video/x-qt-part and video/x-m4-part
	  to caps so schroenc/schroparse can use it.  Fixes #566958

2008-12-19 18:53:47 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/quicktime/gstqtmux.c: Do not tempt or suggest to violate gst_collect_pads API specification.
	  Original commit message from CVS:
	  * gst/quicktime/gstqtmux.c: (gst_qt_mux_change_state):
	  Do not tempt or suggest to violate gst_collect_pads API specification.

2008-12-19 18:33:47 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/quicktime/: Dual license qtmux LGPL/MIT.  Fixes #564232.
	  Original commit message from CVS:
	  * gst/quicktime/atoms.c:
	  * gst/quicktime/atoms.h:
	  * gst/quicktime/descriptors.c:
	  * gst/quicktime/descriptors.h:
	  * gst/quicktime/fourcc.h:
	  * gst/quicktime/ftypcc.h:
	  * gst/quicktime/gstqtmux.c:
	  * gst/quicktime/gstqtmux.h:
	  * gst/quicktime/gstqtmuxmap.c:
	  * gst/quicktime/gstqtmuxmap.h:
	  * gst/quicktime/properties.c:
	  * gst/quicktime/properties.h:
	  Dual license qtmux LGPL/MIT.  Fixes #564232.

2008-12-16 16:26:52 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Totally remove the internal taglists and fully use tagsetter. Fixes various tag muxing issues.
	  Original commit message from CVS:
	  * ext/celt/gstceltenc.c:
	  * ext/celt/gstceltenc.h:
	  * ext/metadata/gstmetadatamux.c:
	  * gst/quicktime/gstqtmux.c:
	  * gst/quicktime/gstqtmux.h:
	  Totally remove the internal taglists and fully use tagsetter. Fixes
	  various tag muxing issues.

2008-12-01 16:37:45 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/quicktime/atoms.c: Fix mj2 sample description metadata construction.
	  Original commit message from CVS:
	  * gst/quicktime/atoms.c: (build_jp2h_extension):
	  Fix mj2 sample description metadata construction.

2008-11-18 01:09:09 +0000  David Schleef <ds@schleef.org>

	  gst/quicktime/gstqtmux.c: Quiet a debugging message that I recently added.
	  Original commit message from CVS:
	  * gst/quicktime/gstqtmux.c: Quiet a debugging message that I recently
	  added.

2008-11-15 02:56:31 +0000  David Schleef <ds@schleef.org>

	  gst/quicktime/gstqtmux.*: Use dts from GST_BUFFER_OFFSET_END() for video/x-qt-part.
	  Original commit message from CVS:
	  * gst/quicktime/gstqtmux.c:
	  * gst/quicktime/gstqtmux.h:
	  Use dts from GST_BUFFER_OFFSET_END() for video/x-qt-part.

2008-11-14 21:24:51 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/quicktime/: Revert previous commit.
	  Original commit message from CVS:
	  * gst/quicktime/atoms.c:
	  * gst/quicktime/atoms.h:
	  * gst/quicktime/descriptors.c:
	  * gst/quicktime/descriptors.h:
	  * gst/quicktime/fourcc.h:
	  * gst/quicktime/ftypcc.h:
	  * gst/quicktime/gstqtmux.c:
	  * gst/quicktime/gstqtmux.h:
	  * gst/quicktime/gstqtmuxmap.c:
	  * gst/quicktime/gstqtmuxmap.h:
	  * gst/quicktime/properties.c:
	  * gst/quicktime/properties.h:
	  Revert previous commit.

2008-11-14 20:38:18 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/quicktime/: Dual license LGPL/MIT, as apparently supposed to.
	  Original commit message from CVS:
	  * gst/quicktime/atoms.c:
	  * gst/quicktime/atoms.h:
	  * gst/quicktime/descriptors.c:
	  * gst/quicktime/descriptors.h:
	  * gst/quicktime/fourcc.h:
	  * gst/quicktime/ftypcc.h:
	  * gst/quicktime/gstqtmux.c:
	  * gst/quicktime/gstqtmux.h:
	  * gst/quicktime/gstqtmuxmap.c:
	  * gst/quicktime/gstqtmuxmap.h:
	  * gst/quicktime/properties.c:
	  * gst/quicktime/properties.h:
	  Dual license LGPL/MIT, as apparently supposed to.

2008-11-14 20:17:10 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/quicktime/: Cut detour in sample description extension construction.
	  Original commit message from CVS:
	  * gst/quicktime/atoms.c: (build_esds_extension),
	  (build_mov_aac_extension), (build_jp2h_extension),
	  (build_codec_data_extension):
	  * gst/quicktime/atoms.h:
	  * gst/quicktime/fourcc.h:
	  * gst/quicktime/gstqtmux.c: (gst_qt_mux_audio_sink_set_caps),
	  (gst_qt_mux_video_sink_set_caps):
	  * gst/quicktime/gstqtmuxmap.c: (gst_qt_mux_map_format_to_header):
	  Cut detour in sample description extension construction.
	  Also actually implement ISO JPEG2000 mj2 format.

2008-11-11 19:31:35 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  tests/check/: Add unit test for qtmux.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/qtmux.c: (setup_src_pad),
	  (teardown_src_pad), (setup_qtmux), (cleanup_qtmux),
	  (check_qtmux_pad), (GST_START_TEST), (qtmux_suite), (main):
	  Add unit test for qtmux.

2008-11-11 19:24:12 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/quicktime/gstqtmux.c: Add some more safety/sanity checks in tag manipulation.
	  Original commit message from CVS:
	  * gst/quicktime/gstqtmux.c: (gst_qt_mux_add_metadata_tags):
	  Add some more safety/sanity checks in tag manipulation.

2008-11-08 02:00:58 +0000  Thiago Sousa Santos <thiagossantos@gmail.com>

	  Copy qtmux from revision 148 of the gst-qtmux repository.
	  Original commit message from CVS:
	  patch by: Thiago Sousa Santos <thiagossantos@gmail.com>
	  * configure.ac:
	  * gst/quicktime/Makefile.am:
	  * gst/quicktime/atoms.c:
	  * gst/quicktime/atoms.h:
	  * gst/quicktime/descriptors.c:
	  * gst/quicktime/descriptors.h:
	  * gst/quicktime/fourcc.h:
	  * gst/quicktime/ftypcc.h:
	  * gst/quicktime/gstqtmux.c:
	  * gst/quicktime/gstqtmux.h:
	  * gst/quicktime/gstqtmuxmap.c:
	  * gst/quicktime/gstqtmuxmap.h:
	  * gst/quicktime/properties.c:
	  * gst/quicktime/properties.h:
	  Copy qtmux from revision 148 of the gst-qtmux repository.
	  Fixes #550280.

2011-04-12 18:25:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Android.mk:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* gst/quicktime/LEGAL:
	* gst/quicktime/Makefile.am:
	* gst/quicktime/gstrtpxqtdepay.c:
	* gst/quicktime/gstrtpxqtdepay.h:
	* gst/quicktime/qtatomparser.h:
	* gst/quicktime/qtdemux.c:
	* gst/quicktime/qtdemux.h:
	* gst/quicktime/qtdemux.vcproj:
	* gst/quicktime/qtdemux_dump.c:
	* gst/quicktime/qtdemux_dump.h:
	* gst/quicktime/qtdemux_fourcc.h:
	* gst/quicktime/qtdemux_lang.c:
	* gst/quicktime/qtdemux_lang.h:
	* gst/quicktime/qtdemux_types.c:
	* gst/quicktime/qtdemux_types.h:
	* gst/quicktime/qtpalette.h:
	* gst/quicktime/quicktime.c:
	* po/POTFILES.in:
	  qtdemux: rename directory to quicktime to match plugin name
	  In preparation for qtmux moving to -good.

2011-04-12 11:49:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: simplify framerate fraction calculation

2011-01-24 15:45:28 -0600  Leonardo Sandoval <lsandoval@ti.com>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: add width, height and framerate to caps when present on onMetaData
	  Fixes #640483.

2010-08-24 13:57:55 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Unknown SSRC is not fatal
	  https://bugzilla.gnome.org/show_bug.cgi?id=646966

2010-08-24 13:54:58 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Number of active sources should be updated whenever the status of the source changes to active
	  Forward-ported by Olivier Crête
	  https://bugzilla.gnome.org/show_bug.cgi?id=646965

2010-06-23 11:29:58 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: ignore a BYE if it is sent with our internal SSRC
	  https://bugzilla.gnome.org/show_bug.cgi?id=646964

2010-01-29 09:49:48 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Adds more h264 fields to its caps
	  Adds alignment=au and stream-format=avc to h264 caps
	  Fixes #606662

2011-04-11 12:44:19 +0300  Stefan Kost <ensonic@users.sf.net>

	* configure.ac:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: also handle deprecations for jack 1.9.7
	  Jack 1.9.7 was released 20.Mar.2011, need to handle the deprecated api for this
	  version too.

2011-04-11 00:36:35 -0400  Thibault Saunier <thibault.saunier@collabora.co.uk>

	* gst/dtmf/Makefile.am:
	  android: make it ready for androgenizer
	  Remove the android/ top dir
	  Fixe the Makefile.am to be androgenized
	  To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
	  Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git

2011-04-10 18:56:52 -0400  Thibault Saunier <thibault.saunier@collabora.co.uk>

	* Android.mk:
	* android/NOTICE:
	* android/apetag.mk:
	* android/avi.mk:
	* android/flv.mk:
	* android/gst/rtpmanager/gstrtpbin-marshal.c:
	* android/gst/rtpmanager/gstrtpbin-marshal.h:
	* android/gst/udp/gstudp-enumtypes.c:
	* android/gst/udp/gstudp-enumtypes.h:
	* android/gst/udp/gstudp-marshal.c:
	* android/gst/udp/gstudp-marshal.h:
	* android/icydemux.mk:
	* android/id3demux.mk:
	* android/qtdemux.mk:
	* android/rtp.mk:
	* android/rtpmanager.mk:
	* android/rtsp.mk:
	* android/soup.mk:
	* android/udp.mk:
	* android/wavenc.mk:
	* android/wavparse.mk:
	* gst/alpha/Makefile.am:
	* gst/apetag/Makefile.am:
	* gst/audiofx/Makefile.am:
	* gst/auparse/Makefile.am:
	* gst/autodetect/Makefile.am:
	* gst/avi/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debugutils/Makefile.am:
	* gst/deinterlace/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/equalizer/Makefile.am:
	* gst/flv/Makefile.am:
	* gst/flx/Makefile.am:
	* gst/goom/Makefile.am:
	* gst/goom2k1/Makefile.am:
	* gst/icydemux/Makefile.am:
	* gst/id3demux/Makefile.am:
	* gst/imagefreeze/Makefile.am:
	* gst/interleave/Makefile.am:
	* gst/law/Makefile.am:
	* gst/level/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/monoscope/Makefile.am:
	* gst/multifile/Makefile.am:
	* gst/multipart/Makefile.am:
	* gst/qtdemux/Makefile.am:
	* gst/replaygain/Makefile.am:
	* gst/rtp/Makefile.am:
	* gst/rtpmanager/Makefile.am:
	* gst/rtsp/Makefile.am:
	* gst/shapewipe/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/spectrum/Makefile.am:
	* gst/udp/Makefile.am:
	* gst/videobox/Makefile.am:
	* gst/videocrop/Makefile.am:
	* gst/videofilter/Makefile.am:
	* gst/videomixer/Makefile.am:
	* gst/wavenc/Makefile.am:
	* gst/wavparse/Makefile.am:
	* gst/y4m/Makefile.am:
	  android: Make it ready for androgenizer
	  Remove the android/ top dir
	  Fixe the Makefile.am to be androgenized
	  To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
	  Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git

2011-04-05 21:14:43 +0200  Haakon Sporsheim <haakon.sporsheim@gmail.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: declare frag_offset to hold 32bits.
	  As specified in documenation above and below.
	  https://bugzilla.gnome.org/show_bug.cgi?id=646954

2011-04-09 12:41:48 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: fix wrongly applied patch
	  Obviously recv_rtp_sink does not have much to do with send_rtcp_src...
	  See commit 046ff170.
	  https://bugzilla.gnome.org/show_bug.cgi?id=647263

2011-04-08 15:59:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparsers: update for set_frame_props -> set_frame_rate API change

2011-04-08 00:03:21 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	  tests: hook up audioparser unit tests

2011-04-07 18:30:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: relax sync match a bit when draining
	  ... to at least allow initial caps change (but no further caps jitter).
	  Fixes unit test again after previous change.

2011-04-07 15:21:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	  docs: update for changes in git

2011-04-07 15:20:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	  docs: add audioparsers to docs

2011-04-07 15:07:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstamrparse.h:
	* gst/audioparsers/plugin.c:
	  aacparse, amrparse: gst_fooparse_xyz -> gst_foo_parse_xyz to match GstFooParse
	  See moving-plugins checklist.

2011-04-07 14:43:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/plugin.c:
	  audioparsers: hook up to build

2011-04-07 13:26:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstamrparse.h:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstdcaparse.h:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstmpegaudioparse.h:
	  audioparsers: port to new GstBaseParse in core

2011-04-04 20:55:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: require tighter sync match when draining

2011-04-01 14:47:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstmpegaudioparse.h:
	  mpegaudioparse: Parse encoder delay and encoder padding from the LAME header if present

2011-03-09 23:06:14 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/plugin.c:
	  dcaparse: Bump rank to primary+1
	  Seems to work fine with a reasonably wide range of media, so bumping
	  rank.

2011-03-23 22:02:37 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstdcaparse.h:
	  dcaparse: Expose frame size in caps
	  This exports the size of the frame (number of bytes from one sync point
	  to the next) as the "frame_size" field in caps.

2011-03-09 23:03:10 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstdcaparse.h:
	  dcaparse: Expose block size in caps
	  This sets the "block_size" field on caps as the number of samples
	  encoded in one frame.

2011-03-16 15:53:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: add FIXME for making the base class use xing seek tables better

2011-03-14 18:25:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstdcaparse.h:
	  dcaparse: Add depth and endianness to the caps
	  Some decoders can only handle specific endianness or a fixed
	  depth and this allows better negotiation.
	  Fixes bug #644208.

2011-02-26 13:53:44 -0800  David Schleef <ds@schleef.org>

	* gst/audioparsers/gstaacparse.c:
	  Revert "aacparse: allow parsed frames on sink pad"
	  This reverts commit e49b89d5c5a1244fa0dcb8bb4996e38fb9bff9e5.

2011-02-23 17:25:03 -0800  David Schleef <ds@schleef.org>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: allow parsed frames on sink pad

2010-10-13 16:12:02 -0700  David Schleef <ds@schleef.org>

	* tests/check/elements/parser.c:
	  tests: fix baseparse test

2010-10-13 15:39:55 -0700  David Schleef <ds@schleef.org>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstaacparse.h:
	* gst/audioparsers/gstac3parse.h:
	* gst/audioparsers/gstamrparse.h:
	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	* gst/audioparsers/gstdcaparse.h:
	* gst/audioparsers/gstflacparse.h:
	* gst/audioparsers/gstmpegaudioparse.h:
	  baseparse: Create baseparse library

2011-02-07 14:46:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: tune QUERY_SEEKING response
	  Even if we currently do not have a duration yet, assume seekable if
	  it looks like we'll likely be able to determine it later on
	  (which coincides with needed information to perform seeking).
	  Fixes #641047.

2011-02-08 23:39:24 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: Update min/max bitrate before first posting them
	  This avoids posting an initial min-bitrate of G_UINTMAX and max-bitrate
	  of 0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=641857

2011-02-08 23:50:13 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstmpegaudioparse.h:
	  mpegaudioparse: Post CBR bitrate as nominal bitrate
	  Even if VBR headers are missing, we can't guarantee that a stream is in
	  fact a CBR stream, so it's safer to let baseparse calculate the average
	  bitrate rather than assume a CBR stream. However, in order to make
	  /some/ metadata available before the requisite number of frames have
	  been parsed, this posts the bitrate from the non-VBR headers as the
	  nominal bitrate.
	  https://bugzilla.gnome.org/show_bug.cgi?id=641858

2010-09-06 14:10:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	  amrparse: a valid amr-wb frame should not have reserved frame type index
	  See #639715.

2011-01-27 16:52:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: improve handling of dependent substream frames
	  In particular, timestamps of these should track main-stream timestamps.

2011-01-21 14:53:39 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: tune default duration estimate update interval
	  Rather than a fixed default frame count, estimate frame count to aim for
	  an interval duration depending on fps if available, otherwise use old
	  fixed default.

2011-01-14 15:16:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: reverse playback; mind keyframes for fragment boundary

2011-01-13 15:26:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	  amrparse: properly check for sufficient available data prior to access

2011-01-12 14:40:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: ensure non-empty candidate frames

2011-01-11 15:24:23 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: clarify some debug statements

2011-01-11 15:24:02 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: properly track upstream timestamps
	  ... rather than with a delay.

2011-01-11 15:23:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: need proper frame duration to obtain sensible frame bitrate

2011-01-11 15:22:51 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: proper initial values for index tracking variables

2011-01-11 12:05:13 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: arrange for consistent event handling

2011-01-10 16:59:59 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.h:
	  baseparse: header style cleaning

2011-01-10 17:07:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: provide some more initial frame metadata in parse_frame
	  ... and document accordingly.

2011-01-10 16:56:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	* gst/audioparsers/gstflacparse.c:
	  baseparse: refactor passthrough into format flags
	  Also add a format flag to signal baseparse that subclass/format can provide
	  (parsed) timestamp rather than an estimated one.  In particular, such "strong"
	  timestamp then allows to e.g. determine duration.

2011-01-10 15:34:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  baseparse: introduce a baseparse frame to serve as context
	  ... and adjust subclass parsers accordingly

2011-01-07 16:39:51 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: restrict duration scanning to pull mode and avoid extra set_caps call

2011-01-07 15:58:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: update some documentation
	  Also add some more debug.

2011-01-06 11:41:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: allow increasing min_size for current frame parsing only
	  Also check that subclass actually either directs to skip bytes or
	  increases expected frame size to avoid going nowhere in bogus
	  indefinite looping.

2011-01-14 15:26:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baesparse: fix refactor regression in loop based parsing

2011-01-06 11:16:56 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: pass all available data to subclass rather than minimum
	  Also reduce some adapter calls and add a few debug statements.

2010-12-10 15:59:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: fix reverse playback handling

2010-12-10 14:56:13 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: minor typo and debug statement cleanup

2010-12-10 14:40:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: reduce locking
	  ... which is either already mute and/or implicitly handled by STREAM_LOCK.

2011-01-14 14:08:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: avoid loop in frame locating interpolation

2011-01-19 18:26:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: mind gst_buffer_unref not liking NULL
	  Fixes #639950.

2011-01-14 16:30:11 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  audioparsers: baseparse: Be careful to not lose the event ref
	  Don't unref the event if it hasn't been handled, because the caller
	  assumes it is still valid and might reuse it.
	  I ran into this problem when transcoding an AVI (with mp3 inside)
	  to gpp.
	  https://bugzilla.gnome.org/show_bug.cgi?id=639555

2011-01-13 17:10:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: fix sync word for 14-bit little endian coding
	  Fix copy'n'paste bug that made us look for the raw little endian
	  sync word twice instead of looking for the 14-bit LE sync word
	  as well. Fixes parsing of such streams (see #636234 for sample file).

2011-01-13 16:27:04 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  docs: minor baseparse docs/comment fixes
	  Remove copy'n'paste leftovers.

2011-01-06 12:49:43 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Fix unitialized variable on macosx

2010-12-13 15:17:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: relax bsid checking
	  ... to the widest possible spec interpretation.
	  Fixes #637062.

2010-12-03 18:11:56 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	  audioparsers: update some documentation

2010-12-03 18:11:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: add to documentation

2010-12-03 18:11:09 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: add to documentation

2010-11-08 19:58:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: increase keyframe awareness
	  ... which is not particular relevant for audio parsing, but more so
	  in video cases.  In particular, auto-determine if dealing with video (caps).

2010-12-01 15:28:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	  ac3parse: use proper EAC-3 caps

2010-11-30 15:41:02 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: avoid unexpected stray metadata

2010-11-30 15:40:28 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: use proper _NONE output value when applicable

2010-11-25 18:56:42 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstbaseparse.c:
	  audioparsers: Remove dead assignments

2010-11-25 17:14:23 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/audioparsers/gstbaseparse.c:
	  audioparse: fix possible division-by-zero
	  https://bugzilla.gnome.org/show_bug.cgi?id=635786

2010-11-17 16:23:42 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: use correct offset when adding index entry
	  ... bearing in mind that BUFFER_OFFSET is media specific and may not
	  reflect the basic offset after having been parsed.

2010-11-17 14:30:09 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: enhancements for timestamp marked framed formats
	  That is, as such formats allow subclass to extract position from frame,
	  it is possible to extract duration (if not otherwise provided)
	  from (near) last frame, and a seek can fairly accurately target the required
	  position.
	  Fixes #631389.

2010-11-16 17:06:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: refactor frame scanning peformed by _loop

2010-11-16 18:04:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: slightly optimize sending of pending newsegment events

2010-11-16 17:04:35 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: minor fixes and enhancements
	  Arrange for upstream as well as downstream flushing when seeking.
	  Also determine upstream size as well as seekability.  Adjust some comments
	  to reality and employ debug statement in proper order.

2010-11-17 15:33:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: minor cleanups

2010-11-17 15:24:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix regression in ADIF src caps setting

2010-11-16 12:11:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: parse seektable
	  Fixes #631389 (partially).

2010-11-16 12:08:54 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: minor refactor and enable default baseparse segment clipping

2010-11-09 19:38:25 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: fix silly leak in _reset

2010-10-29 14:08:58 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: use only upstream duration if it provides one

2010-10-25 14:15:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: reflow update_bitrate code
	  ... which makes local variables represent real state better, and avoids
	  triggering unneeded updates/actions.

2010-10-25 14:13:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: add some debug statements

2010-10-19 23:25:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: init variable to make osx build bot happy
	  gstdcaparse.c: In function 'gst_dca_parse_check_valid_frame':
	  gstdcaparse.c:246: warning: 'best_sync' may be used uninitialized in this function

2010-10-19 00:15:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstdcaparse.h:
	* gst/audioparsers/plugin.c:
	  audioparsers: add very basic dts/dca parser
	  Still some issues, e.g. with seekable queries in totem, but also
	  processing already-chunked input (created with matroskademux ! gdppay).

2010-10-14 16:48:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: properly parse e-ac3 frame header
	  Also add a few debug statements.

2010-10-13 11:00:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: tweak setting buffer metadata; avoid timestamp jitter
	  Fixes #631993.

2010-10-12 18:07:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	  aacparse: streamline src caps setting
	  In particular, also set src caps whenever changes in stream warrant doing so.

2010-10-12 10:28:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/flacparse.c:
	  flacparse: Adjust unit tests to new flacparse behaviour
	  Garbage after frames is now included in the frames because flacparse
	  has no easy way to detect the real end of a frame. Decoders are
	  expected to everything after the frame because only decoding the
	  bitstream will reveal the real end of the frame.
	  Fixes bug #631814.

2010-10-12 10:27:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Don't drop the last frame if it is followed by garbage
	  See bug #631814.

2010-10-11 17:49:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: perform bitrate handling and posting after newsegment sending

2010-10-11 17:36:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: immediately post subclass provided bitrate

2010-10-11 17:06:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix parsing with unknown framesizes
	  Fixes #631814 (mostly).

2010-10-07 23:37:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Simplify frame header parsing by using lookup tables
	  Based on a patch by Felipe Contreras.
	  See bug #631200.

2010-10-07 23:28:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: Don't parse the complete FLAC frames but only look for valid frame headers
	  Thanks to Felipe Contreras for the suggestion. This is partially
	  based on his patches and makes flacparse more than 3.5 times faster.
	  Looking for valid frame headers is unlikely to give false positives
	  because every frame header is at least 9 bytes long, contains a
	  14 bit sync code and a 8 bit checksum over the first 8 bytes.
	  Fixes bug #631200.

2010-10-06 18:32:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Really post tags only after the initial newsegment event
	  The first newsegment event will be send by the first call to
	  gst_base_parse_push_buffer() if necessary, posting the tags
	  before that is not a good idea. Instead do it from the
	  GstBaseParse::pre_push_buffer vfunc.

2010-10-05 11:17:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  Revert "baseparse: add skip property"
	  This reverts commit b5a3d60363d837a10f0533c141ec93d10b742312.
	  Reverting this for now, since no one really seems to remember why this
	  property exists or what it could possibly be good for. It seems to have
	  been in the original mp3parse since the beginning of time and was back-
	  ported from there.

2010-10-04 10:41:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Fix uninitialized variable compiler warnings
	  These warnings are wrong, the variables are only used if they were
	  initialized by the bit reader.

2010-09-14 02:48:58 +0300  Felipe Contreras <felipe.contreras@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix picture parsing
	  Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>

2010-10-03 23:54:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Push tags before the header buffers are pushed

2010-08-02 20:50:21 +0300  Felipe Contreras <felipe.contreras@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: trivial caps fix
	  Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>

2010-10-03 23:50:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  audioparser: Let the format string agree with the parameters to fix compiler warning

2010-10-03 15:41:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: Use unchecked versions of the bitreader get functions
	  We didn't check the return values anyway...

2010-09-22 15:44:43 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: Fix debug output
	  We lose the reference to the buffer after gst_pad_push(), so the debug
	  print should happen before.
	  https://bugzilla.gnome.org/show_bug.cgi?id=622276

2010-10-01 12:34:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/flacparse.c:
	* tests/check/elements/parser.c:
	* tests/check/elements/parser.h:
	  audioparsers: add flacparse unit test
	  ... and tweak parser test helper in the process.

2010-09-29 16:12:42 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: support reverse playback
	  ... in pull mode or upstream driven.

2010-09-27 12:16:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: remove done TODOs and update documentation

2010-09-25 14:40:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: use determined seekability in answering SEEKING query

2010-09-25 14:32:06 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: add skip property

2010-09-25 13:59:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/ac3parse.c:
	* tests/check/elements/mpegaudioparse.c:
	  audioparsers: add ac3parse and mpegaudioparse unit test

2010-09-25 13:59:18 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstmpegaudioparse.h:
	* gst/audioparsers/plugin.c:
	  mpegaudioparse: initial version
	  ... adequately equivalent to mp3parse, so lets boldly set it
	  to higher rank.

2010-09-25 14:01:07 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: set minimum frame size at _start
	  ... rather than one time at _init.

2010-09-25 13:50:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/aacparse.c:
	* tests/check/elements/amrparse.c:
	* tests/check/elements/parser.c:
	* tests/check/elements/parser.h:
	  audioparsers: refactor existing unit tests using common helper

2010-09-22 15:07:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: use _set_frame_props to configure frame lead_in and lead_out
	  ... provided a corresponding decoder with sufficient leading and following
	  frames to carry out full decoding for a particular segment.

2010-09-22 14:13:17 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	* gst/audioparsers/gstflacparse.c:
	  baseparse: use _set_duration to configure duration update interval
	  ... as it logically belongs there as one or the other; either subclass
	  can provide a duration, or an estimate must be made (reguarly updated).

2010-09-22 13:55:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: localize use of provided fps information

2010-09-22 12:13:12 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: seek table and accurate seek support

2010-09-21 13:57:10 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: proper and more extended segment and seek handling
	  That is, loop pause handling, segment seek support, newsegment for gaps, etc

2010-09-21 10:57:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: add index support

2010-09-21 09:59:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: refactor state reset

2010-09-20 16:39:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: prevent indefinite resyncing

2010-09-20 13:57:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: specific EOS handling if no output so far

2010-09-20 13:31:57 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: adjust _set_frame_prop documentation and set default as claimed

2010-09-20 13:30:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: fix bitrate copy-and-paste and update heuristic

2010-09-17 18:33:29 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: post duration message if average bitrates is updated

2010-09-17 18:24:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: remove is_seekable vmethod and use a set_seek instead
	  Seekability, like duration, etc is unlikely to change (frequently), and
	  the default assumption covers most cases, so let subclass set when needed.
	  At the same time, allow subclass to indicate if it has seek-metadata (table)
	  available, and possibly have it provide an average bitrate.

2010-09-17 17:35:40 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: remove redundant default is_seekable

2010-09-17 17:21:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: add another hook for subclass prior to pushing buffer
	  ... and allow subclass to perform custom segment clipping, or to
	  emit tags or messages at this time.

2010-09-17 17:19:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: 0 converts to 0 by default

2010-09-16 18:56:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  basepase: refactor conversion using helper function and export default convert

2010-09-16 18:35:47 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: streamline query handling

2010-09-16 11:51:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: cleanup struct and remove unused member

2010-08-16 11:04:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/plugin.c:
	  audioparsers: increase ranks to enable auto-plugging
	  Because we can, and should, have some shakedown testing before having
	  these make it into -good later on ...

2010-09-22 16:07:24 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: Allow chaining of subclass event handlers
	  This allows the child class to chain its event handler with
	  GstBaseParse, so that subclasses don't have to duplicate all the default
	  event handling logic.
	  https://bugzilla.gnome.org/show_bug.cgi?id=622276

2010-08-27 18:35:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: Don't use GST_FLOW_IS_FATAL()
	  Also don't post an error message for UNEXPECTED and do it
	  for NOT_LINKED.

2010-09-06 14:12:00 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: non-TIME seek event is simply not handled

2010-06-15 15:34:05 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: fix seek event ref handling

2010-06-15 15:33:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: prevent arithmetic overflows in pull mode buffer cache handling

2010-06-15 15:32:34 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: fix seek handling
	  Allow a few more seek event type combinations, and really use the result
	  of gst_segment_set_seek to perform the seek.  Also add some debug.

2010-04-12 18:07:29 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/aacparse.c:
	* tests/check/elements/amrparse.c:
	  check: Don't re-declare 'GList *buffers' in the tests
	  It's an external which lives in gstcheck.c. Redeclaring it makes some
	  compilers/architectures think the 'buffers' in the individual tests are
	  a different symbol... and therefore we end up comparing holodecks with
	  oranges.

2010-03-26 18:56:49 +0000  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: Don't emit bitrate tags too early
	  We wait to parse a minimum number of frames (10, arbitrarily) before
	  emiting bitrate tags so that our early estimates are not wildly
	  inaccurate for streams that start with a silence. If the stream ends
	  before that, we just emit the tags anyway.
	  While it _would_ be nicer to be specify the threshold to start pushing
	  the tags in terms of duration, this would introduce more complexity than
	  this merits.
	  https://bugzilla.gnome.org/show_bug.cgi?id=614991

2010-03-26 18:58:35 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: Optionally check the overall frame checksums too before accepting a frame as valid
	  This is optional because it's a quite expensive operation and it's very
	  unlikely that a non-frame is detected as frame after the header CRC check
	  and checking all bits for valid values. The overall frame checksums are
	  mainly useful to detect inconsistencies in the encoded payload.

2010-03-26 18:42:28 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Check the CRC-8 of the headers before accepting a frame as valid
	  This makes false-positives during seeking much less likely and detection of
	  them much faster.

2010-03-26 18:20:24 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: Set the last stop to the buffer starttime if the duration is invalid
	  ...instead of not setting it at all.

2010-03-26 18:19:00 +0100  Joshua M. Doe <josh@joshdoe.com>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: Send NEWSEGMENT event with correct start and position
	  Instead of taking the last stop (which could be buffer endtime instead
	  of starttime) always take the buffer starttime.
	  Fixes bug #614016.

2010-03-26 16:49:01 +0000  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Fix buffer refcount issue
	  When called from the GST_FLAC_PARSE_STATE_HEADERS case,
	  gst_flac_parse_hand_headers() does a gst_buffer_set_caps() on a buffer
	  with refcount > 1. This change handles this case by making the buffer
	  metadata_Writable.
	  https://bugzilla.gnome.org/show_bug.cgi?id=614037

2010-03-25 17:09:17 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  audioparsers: remove unused GstBaseParseClassPrivate structure

2010-03-25 12:55:02 +0000  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Make bitrate estimation more accurate
	  This implements the get_frame_overhead() vfunc so that baseparse can
	  make more accurate bitrate estimates.

2010-03-25 11:48:46 +0000  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Fix bitrate calculation
	  This patch adds the get_frame_overhead() vfunc so that baseparse can
	  accurately calculate the min/avg/max bitrates for aacparse.
	  Note: The bitrate was being incorrectly calculated for ADTS streams
	  (it's not in the header as the code suggests).

2010-03-25 11:22:58 +0000  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  audioparsers: Add bitrate calculation to baseparse
	  This makes baseparse keep a running average of the stream bitrate, as
	  well as the minimum and maximum bitrates. Subclasses can override a
	  vfunc to make sure that per-frame overhead from the container is not
	  accounted for in the bitrate calculation.
	  We take care not to override the bitrate, minimum-bitrate, and
	  maximum-bitrate tags if they have been posted upstream. We also
	  rate-limit the emission of bitrate so that it is only triggered by a
	  change of >10 kbps.

2010-03-22 16:56:03 +0100  Benjamin Otte <otte@redhat.com>

	* tests/check/elements/amrparse.c:
	  Add -Wold-style-definition
	  and fix the warnings

2010-03-21 21:39:18 +0100  Benjamin Otte <otte@redhat.com>

	* tests/check/elements/aacparse.c:
	* tests/check/elements/amrparse.c:
	  Add -Wmissing-declarations -Wmissing-prototypes to configure flags
	  And fix all warnings

2010-03-18 17:30:26 +0100  Benjamin Otte <otte@redhat.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstamrparse.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple

2010-01-14 11:50:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  audioparsers: rename baseparse GType name to avoid possible conflicts

2010-01-12 18:55:53 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Initialize variables.
	  Fixes build on $#@*( macosx

2010-01-11 22:41:57 +0300  ������ ��������� <lrn1986@gmail.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstamrparse.c:
	  win32: Include config.h before anything else. Fix mpegdemux LIBADD
	  Because config.h defines __MSVCRT_VERSION__, which should be defined
	  before inclusion of any system header.
	  Also fixes mpegdemux Makefile.am LIBADD typo.
	  Fixes #606665

2010-01-11 13:20:26 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Also add stream-format to template caps
	  Do not forget to add stream-format to template caps
	  off aacparse

2010-01-11 13:13:41 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* tests/check/elements/aacparse.c:
	  Rename aac's stream-format 'none' to 'raw'
	  Renames aac's stream-format from previous commits from none to
	  raw

2010-01-11 12:10:02 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/aacparse.c:
	  aacparse: update tests to stream-format changes
	  Updates aacparse unit tests to check for stream-format
	  correctness as well.

2010-01-11 10:51:18 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Add stream-format to output caps
	  Adds stream-format field to output caps

2010-01-05 15:05:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstbaseparse.c:
	  audioparsers: documentation fixes

2010-01-05 15:04:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: add documentation

2010-01-05 14:48:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: add documentation

2009-12-21 18:29:43 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: perform additional frame checks when resyncing

2010-01-05 16:35:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix (multiple channel) frame parsing

2010-01-05 16:35:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: declare unparsed input and parsed output

2009-12-21 18:19:23 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: fix scanning for next syncword

2009-12-21 18:18:39 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: adjust seek handling and newsegment sending
	  Perform sanity check on type of seek, and only perform one that is
	  appropriately supported.  Adjust downstream newsegment event
	  to first buffer timestamp that is sent downstream.

2009-12-21 11:59:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: minor refactor cleanup
	  Also add some debug logging.

2009-12-18 21:05:11 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: locate next sync code more efficiently

2009-12-18 21:04:12 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: baseparse takes care of handling leftover pieces

2009-12-18 21:02:40 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: implement leftover draining in pull mode

2009-12-17 12:45:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: set _OFFSET and _OFFSET_END on outgoing buffers

2009-12-17 12:44:20 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	* gst/audioparsers/plugin.c:
	  audioparsers: move 'flacparse' into it

2009-12-16 18:38:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: provide default conversion using bps if no fps available
	  Also store estimated duration as such, rather than pretending otherwise
	  (e.g. set by subclass).

2009-12-18 13:30:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: check for remaining data when draining in push mode

2009-12-18 13:30:07 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  baseparse: fix pull mode cache size comparison

2009-12-18 13:01:17 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: Fix unitialized variable.

2009-12-17 14:46:01 +0000  Christian Schaller <christian.schaller@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	  Update spec file and fix ac3parser header listing in Makefile.am

2009-12-11 10:25:16 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/audioparsers/gstbaseparse.c:
	  audioparse: fix a format string as reported on irc.

2009-11-23 16:34:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: ensure sufficient data available for parsing

2009-10-29 15:19:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: extract and use some more details for Enhanced Ac-3 streams

2009-10-29 15:18:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	  baseparse: custom bufferflag indicates not to count frame in stats

2009-10-28 14:08:43 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: perform additional frame checks when resyncing

2009-10-28 14:07:17 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: inform base parser of frame duration

2009-10-27 16:16:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: improve src caps settings

2009-11-27 17:59:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	* gst/audioparsers/plugin.c:
	  ac3parse: initial version
	  MARGINAL rank for now; might take some time for some (useful)
	  framed=true/false to appear here and there.

2009-11-26 18:34:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstamrparse.h:
	  amrparse: use (default) time handling of baseparser class

2009-11-26 18:15:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstamrparse.h:
	* gst/audioparsers/plugin.c:
	  audioparsers: move 'amrparse' into it

2009-11-27 17:27:32 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstbaseparse.c:
	  audioparsers: reference GstBaseParse now lives here

2009-11-28 18:13:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/Makefile.am:
	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	* gst/audioparsers/gstbaseparse.c:
	* gst/audioparsers/gstbaseparse.h:
	* gst/audioparsers/plugin.c:
	  audioparsers: rename 'aacparse' plugin to generic 'audioparsers' plugin

2009-11-26 17:04:43 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/Makefile.am:
	* gst/aacparse/gstaacparse.c:
	* gst/aacparse/plugin.c:
	  aacparse: separate plugin registration and rename plugin

2009-11-26 17:04:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstaacparse.c:
	  aacparse: ensure sufficient data available before accessing

2009-11-05 14:31:40 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstaacparse.c:
	* gst/aacparse/gstaacparse.h:
	  aacparse: use (default) time handling of baseparser class

2009-10-29 15:19:35 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstaacparse.c:
	  aacparse: fixup comments to C-style

2009-10-29 16:05:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: reset passthrough mode to default (disabled) on activation

2009-10-29 15:16:59 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: ensure buffer metadata is writable

2009-10-28 14:06:13 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	* gst/aacparse/gstbaseparse.h:
	  baseparse: fix/enhance DISCONT marking
	  In particular, consider DISCONT == !sync, and allow subclass to query
	  sync state, as it may want to perform additional checks depending
	  on whether sync was achieved earlier on.
	  Also arrange for subclass to query whether leftover data is being drained.

2009-11-23 15:48:25 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	* gst/aacparse/gstbaseparse.h:
	  baseparse: add timestamp handling, and default conversion
	  In particular, (optionally) provide baseparse with a notion of frames per second
	  (and therefore also frame duration) and have it track frame and byte counts.
	  This way, subclass can provide baseparse with fps and have it provide default
	  buffer time metadata and conversions, though subclass can still install
	  callbacks to handle such itself.

2009-10-28 12:02:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: documentation fixes

2009-10-28 12:00:08 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: use_fixed_caps for src pad
	  After all, stream is as-is, and there is little molding to downstream's
	  taste that can be done.  If subclass can and wants to do so, it can
	  still override as such.

2009-11-20 17:32:13 +0100  Julien Moutte <julien@fluendo.com>

	* gst/aacparse/gstbaseparse.c:
	  aacparse: Fix compilation warnings

2009-10-11 11:22:11 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/aacparse/gstaacparse.c:
	* gst/aacparse/gstbaseparse.c:
	  aacparse: fix warnings in macosx snow leopard

2009-09-25 17:02:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstaacparse.c:
	* gst/aacparse/gstbaseparse.c:
	* gst/aacparse/gstbaseparse.h:
	  aacparse: forego (bogus) parsing of already parsed (raw) input

2009-08-07 13:07:17 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: prevent infinite loop when draining

2009-08-07 13:06:28 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: fix minor memory leak

2009-07-14 14:08:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	* gst/aacparse/gstbaseparse.h:
	  aacparse: Add function for the baseparse subclass to push buffers downstream
	  Also handle the case gracefully where the subclass decides to drop
	  the first buffers and has no caps set yet. It's still required to
	  have valid caps set when the first buffer should be passed downstream.

2009-07-14 14:07:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: Fix seek event leaking

2009-06-18 12:13:28 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstaacparse.c:
	  aacparse: ADIF: do not send bogus timestamps, leave to downstream (decoder)

2009-06-01 15:53:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/aacparse/gstaacparse.c:
	  aacparse: fix sample rate extraction from codec data
	  In one case we extracted the sample rate index from the codec data
	  and saved it as sample rate rather than getting the real sample
	  rate from the table. Fix that, and also make sure we don't access
	  non-existant table entries by adding a small helper function that
	  guards against out-of-bounds access in case of invalid input data.

2009-06-01 14:02:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/aacparse/gstaacparse.c:
	  aacparse, amrparse: remove bogus gst_pad_fixate_caps() calls

2009-06-01 13:56:18 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: propagate return value of GstBaseParse::set_sink_caps()
	  gst_base_parse_sink_setcaps() presumably should fail if the subclass
	  returns FALSE from its ::set_sink_caps() function.

2009-06-01 13:47:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: don't try to GST_LOG an already-freed caps string
	  The proper way to log caps is via GST_PTR_FORMAT anyway.

2009-06-01 13:05:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/aacparse/gstaacparse.c:
	* tests/check/elements/aacparse.c:
	  aacparse: set channels and rate on output caps, and keep codec_data
	  Create output caps from input caps, so we maintain any fields we
	  might get on the input caps, such as codec_data or rate and channels.
	  Set channels and rate on the output caps if we don't have input caps
	  or they don't contain such fields. We do this partly because we can,
	  but also because some muxers need this information. Tagreadbin will
	  also be happy about this.

2009-05-26 19:43:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: fix debug category

2009-04-27 22:39:15 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: fix (regression in) newsegment handling
	  (aacparse, amrparse, flacparse).  Fixes #580133.

2009-04-07 04:53:02 +0300  René Stadler <mail@renestadler.de>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: Fix slightly broken buffer-in-segment check (aacparse, amrparse, flacparse)

2009-04-05 03:50:19 +0300  René Stadler <mail@renestadler.de>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: Fix push mode seeking (aacparse, amrparse)
	  Sending the flush-start event forward before taking the stream lock actually
	  works, in contrast to deadlocking in downstream preroll_wait (hunk 1).
	  After that we get the chain function being stuck in a busy loop. This is fixed
	  by updating the minimum frame size inside the synchronization loop because the
	  subclass asks for more data in this way (hunk 2).
	  Finally, this leads to a very probable crash because the subclass can find a
	  valid frame with a size greater than the currently available data in the
	  adapter. This makes the subsequent gst_adapter_take_buffer call return NULL,
	  which is not expected (hunk 3).

2009-03-31 16:07:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: Delay newsegment as long as possible.
	  If newsegment is sent (too) early, caps may not yet be fixed/set,
	  and downstream may not have been linked.

2009-03-19 01:17:25 +0200  René Stadler <mail@renestadler.de>

	* gst/aacparse/gstaacparse.c:
	  aacparse: Fix busyloop when seeking. Fixes #575388
	  The problem is that after a discont, set_min_frame_size(1024) is called when
	  detect_stream returns FALSE. However, detect_stream calls check_adts_frame
	  which sets the frame size on its own to something larger than 1024. This is the
	  same situation as in the beginning, so the base class ends up calling
	  check_valid_frame in an endless loop.

2009-03-19 00:32:40 +0200  René Stadler <mail@renestadler.de>

	* gst/aacparse/gstaacparse.c:
	  aacparse: Refactor check_valid_frame to expose broken code
	  Just moving code around and removing an unhelpful/misleading comment.

2009-02-27 11:24:37 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: revert last change and properly fix
	  Baseparse internaly breaks the semantics of a _chain function by calling it with
	  buffer==NULL. The reson I belived it was okay to remove it was that there is
	  also an unchecked access to buffer later in _chain. Actually that code is wrong,
	  as it most probably wants to set discont on the outgoing buffer.

2009-02-26 11:02:06 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/aacparse/gstbaseparse.c:
	  baseparse: remove checks for buffer==NULL
	  Accordifn to docs for GstPadChainFunction buffer cannot be NULL. If we would
	  leave the check, we would also need more such check below.

2009-02-11 00:15:43 +0200  René Stadler <mail@renestadler.de>

	* gst/aacparse/gstaacparse.c:
	  aacparse: Fix license specified in plugin details.

2009-01-30 18:18:10 +0000  Jan Schmidt <jan.schmidt@sun.com>

	* gst/aacparse/gstbaseparse.c:
	  Fix the return value of the default parse_frame function.
	  Fix the return value of the default parse_frame function in both
	  copies of GstBaseParse

2009-01-23 16:00:10 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/aacparse/gstaacparse.c:
	  Log aac details found in codec_data.

2008-11-13 17:24:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/aacparse/gstaacparse.c: Don't autoplug aacparse until it works.
	  Original commit message from CVS:
	  * gst/aacparse/gstaacparse.c: (plugin_init):
	  Don't autoplug aacparse until it works.

2008-11-13 15:20:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/: Add unit tests for new parsers.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/aacparse.c:
	  * tests/check/elements/amrparse.c:
	  Add unit tests for new parsers.

2008-11-13 14:21:39 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: Fix baseparse type name.
	  Original commit message from CVS:
	  * gst/aacparse/gstbaseparse.c:
	  * gst/amrparse/gstbaseparse.c:
	  Fix baseparse type name.

2008-11-13 12:59:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add two new baseparse based parsers (aac and amr) from Bug #518857.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/aacparse/Makefile.am:
	  * gst/aacparse/gstaacparse.c:
	  * gst/aacparse/gstaacparse.h:
	  * gst/aacparse/gstbaseparse.c:
	  * gst/aacparse/gstbaseparse.h:
	  * gst/amrparse/Makefile.am:
	  * gst/amrparse/gstamrparse.c:
	  * gst/amrparse/gstamrparse.h:
	  * gst/amrparse/gstbaseparse.c:
	  * gst/amrparse/gstbaseparse.h:
	  Add two new baseparse based parsers (aac and amr) from Bug #518857.

2011-03-20 01:08:38 +0100  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Make src_query MT-safe
	  It is possible that the element might be going down while the event arrives

2011-04-08 15:22:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Unref event if the parent element disappeared

2011-04-08 15:22:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Unref event if the parent element disappeared

2011-03-21 16:04:34 +0100  Havard Graff <havard.graff@tandberg.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Make upstream events MT-safe

2011-03-21 16:04:34 +0100  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Make upstream events MT-safe

2011-04-08 15:20:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtp: Unref events if the parent element disappeared

2011-01-06 18:24:36 +0100  Ole André Vadla Ravnås <oravnas@cisco.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpmanager: fix pad callbacks so they handle when parent goes away
	  1) We need to lock and get a strong ref to the parent, if still there.
	  2) If it has gone away, we need to handle that gracefully.
	  This is necessary in order to safely modify a running pipeline. Has been
	  observed when a streaming thread is doing a buffer_alloc() while an
	  application thread sends an event on a pad further downstream, and from
	  within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
	  while the streaming thread has its buffer_alloc() in progress.

2010-11-26 15:20:04 +0100  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: make iterate_internal_links MT-safe

2011-04-08 14:35:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  Revert "Pulsesink: Allow chunks up to bufsize instead of segsize"
	  This reverts commit 1e2c1467ae042a3c6bb1a6bc0c07aeff13ec5edb.
	  The commit causes pulsesink to ignore the latency-time baseaudiosink property.

2011-04-08 11:13:07 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/rtp/gstrtpspeexpay.c:
	  rtpspeexpay: Do not transmitt samples with GAP flag
	  If we get GAP samples, there is no need to transmitt it.
	  In some situations, microphone is muted, we can drop net traffick
	  usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s

2011-04-08 11:11:58 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* ext/speex/gstspeexenc.c:
	  speexenc: Use speex intern silence detection
	  Speex has build in silence detection. If speex_encode_int returns 0,
	  than there is silence and sample do not need to be transmitted.
	  This work only if vbr=1 and dtx=1 optionas are enabled.
	  So if we get 0, we add GAP flag to the sample.

2011-04-07 19:04:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	  rtp: port some pay/depayloaders

2011-04-05 19:15:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udpsink: handle scather gather from buffers
	  Iterate the memory blocks on the buffer and send them using sendmsg.

2011-04-05 17:26:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtpdec.c:
	  rtpdec: reset structure before use

2011-04-05 17:20:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/rtsp/gstrtspsrc.c

2011-04-05 17:12:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle * control correctly
	  Parse session control attributes when no media control attribute is
	  present. Threat * control attributes as an empty string, just like the
	  spec says.
	  Fixes #646800

2011-04-05 17:06:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	  rtsp/udp: port to 0.11

2011-04-05 14:28:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Add support for A-Law and µ-Law
	  Fixes bug #646567.

2011-04-05 09:44:01 +0200  Jon Nordby <jononor@gmail.com>

	* configure.ac:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: Fix build with jack 0.120.1
	  9544622674c0d0a3147a9b51145159b02eec68e9 checked
	  for 0.120.2 and later, but the deprecation was introduced in
	  0.120.1

2011-04-05 11:13:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavisubtitle.c:
	  avi: more porting to 0.11

2011-04-05 12:05:19 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2radio.h:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2xoverlay.c:
	  docs: fix docuemntation warnings (and reindent)

2011-04-04 19:17:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	  avi: port to 0.11 API

2011-04-04 17:34:17 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	  videomixer: update orc dist files

2011-04-04 15:57:10 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 1ccbe09 to c3cafe1

2011-03-01 14:08:12 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Always call pa_stream_new_with_proplist()
	  pa_stream_new_with_proplist() can take a NULL proplist, so we don't need
	  to concern ourselves with whether it's NULL or not.

2011-04-04 11:33:10 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: perform post-flush state tricks downstream to upstream
	  ... so downstream is set when upstream resumes data flow.

2011-04-04 11:27:29 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: distribute new base_time to manager children following flush seek
	  ... by forcing a state changed to PLAYING, which should otherwise be a
	  no-op as elements should already be in that state.
	  In particular, jitterbuffer needs new base_time as soon as possible to perform
	  proper timing (e.g. eos timeout handling) and can't wait for the new base_time
	  that will be distributed when the whole pipeline returns to PLAYING.
	  See bug #646397.

2011-04-04 11:35:59 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Revert "jitterbuffer: reset element base_time upon flush"
	  This reverts commit f84b8a69cba9c538f5546869cb4ef454ad5efb9d.
	  Fixes bug #646397.

2011-04-04 10:31:44 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	  flv: Specify the only possible stream-format for h264 in the pad templates.

2011-04-04 10:07:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Check for invalid (empty) classification info entity strings
	  Otherwise the classification string can be empty and gst_tag_list_add() will
	  complain or have a \0 in the first four bytes, which is wrong too.

2011-04-04 10:01:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Year 0 is not a valid year for GDate and the proleptic gregorian calendar

2011-04-01 13:18:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: Add support for writing METADATA_BLOCK_PICTURE blocks for GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE

2011-04-01 11:33:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer2.c:
	  videomixer[2]: Use orc_memset() instead of memset()

2011-01-19 18:06:45 -0700  Lane Brooks <dirjud@gmail.com>

	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Add transparent background option for alpha channel formats

2011-01-19 12:07:17 -0700  Lane Brooks <dirjud@gmail.com>

	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/blendorc.orc:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer2: Add transparent background option for alpha channel formats
	  This option allows the videomixer2 element to output a valid alpha
	  channel when the inputs contain a valid alpha channel. This allows
	  mixing to occur in multiple stages serially.
	  The following pipeline shows an example of such a pipeline:
	  gst-launch videotestsrc background-color=0x000000 pattern=ball ! video/x-raw-yuv,format=\(fourcc\)AYUV ! videomixer2 background=transparent name=mix1 ! videomixer2 name=mix2 ! ffmpegcolorspace ! autovideosink  videotestsrc ! video/x-raw-yuv,format=\(fourcc\)AYUV ! mix2.
	  The first videotestsrc in this pipeline creates a moving ball on a
	  transparent background. It is then passed to the first videomixer2.
	  Previously, this videomixer2 would have forced the alpha channel to
	  1.0 and given a background of checker, black, or white to the
	  stream. With this patch, however, you can now specify the background
	  as transparent, and the alpha channel of the input will be
	  preserved. This allows for further mixing downstream, as is shown in
	  the above pipeline where the a second videomixer2 is used to mix in a
	  background of an smpte videotestsrc. So the result is a ball hovering
	  over the smpte test source. This could, of course, have been
	  accomplished with a single mixer element, but staged mixing is useful
	  when it is not convenient to mix all video at once (e.g. a pipeline
	  where a foreground and background bin exist and are mixed at the final
	  output, but the foreground bin needs an internal mixer to create
	  transitions between clips).
	  Fixes bug #639994.

2011-03-31 13:25:00 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: also uncork during EOS waiting (and after EOS is rendered)
	  Pulsesink was recently changed to defer uncorking until there is data
	  to write. This condition will however never occur when EOS in being
	  rendered (since that marks the end of data). Changing to PAUSED state
	  while EOS is being waited on results in a hang: pausing corks the
	  stream, which will never be undone since there is no more data when
	  going back to PLAYING. If pulsesink is the clock provider, deadlock
	  ensues since time doesn't continue in corked state and the clock id
	  for EOS wait never fires.
	  Fixes #645961.

2011-03-29 16:33:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  rtpbin: Don't try to request the same request pad twice

2011-03-28 23:46:47 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: fix issues with large metadata blocks when streaming unframed flac
	  Parse metadata blocks when handling unparsed flac in push mode. This
	  works around a bunch of issues with the flac decoder when handling
	  metadata blocks that are larger than the max. flac framesize, which
	  coverart blocks often are. We need to have all the data for these
	  blocks available when we pass data to libflac.
	  http://gstreamer-devel.966125.n4.nabble.com/Flac-files-that-will-playback-but-not-stream-td3338198.html#a3395276
	  https://bugzilla.gnome.org/show_bug.cgi?id=566769

2011-03-28 21:05:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	  plugins: port to new memory API

2011-03-28 20:50:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11-fdo

2011-03-27 21:39:50 +0200  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Do not build an index if upstream is not seekable
	  An index is not useful if upstream cannot handle seeks and building it
	  for infinite files, for instance FLV streams, results in a memory leak.

2011-03-27 01:19:58 +0300  Alexey Chernov <4ernov@gmail.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2radio.h:
	  v4l2: new v4l2radio element to control analog radio devices
	  https://bugzilla.gnome.org/show_bug.cgi?id=640118

2011-03-25 22:22:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 193b717 to 1ccbe09

2011-03-25 14:56:06 +0200  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From b77e2bf to 193b717

2011-03-25 12:53:43 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/cairo/Makefile.am:
	  cairo: fix the name of the *-marshall.list file to unbreak make distcheck

2011-03-25 09:31:03 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From d8814b6 to b77e2bf

2011-03-25 09:06:16 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 6aaa286 to d8814b6

2011-03-25 00:10:56 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum: refactor processing loop for block based operation
	  Previously the chain function was working sample frame based. In each cycle it
	  was checking if it is time to run a fft or if it is time to send a message.
	  Now we changed the data transform functions to work on a block of data and
	  calculate the max length until either {end-of-data, do-fft, do-msg}. This allows
	  us also to avoid the duplicated code for the single and multi-channel case (as
	  the transformers have the same signature now).

2011-03-24 23:47:33 +0200  Stefan Kost <ensonic@users.sf.net>

	* configure.ac:
	  jack: unbreak the build for jack2 users
	  Jack2 (versions 1.X.X) does only have that API in svn. Limmit the use of the new
	  API for jack1 versions.

2011-03-24 18:49:19 +0200  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 6aec6b9 to 6aaa286

2011-03-24 14:14:09 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: fix the error accumulation and frames_todo handling
	  Even though we wrap around the accumulated second, we still need to add the
	  error in the same cycle. Increase the todo in the same conditional as afterwards
	  the accumulated error will be below one second.

2011-03-24 13:53:12 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: fix broken code resulting for a wrong splitup of changes

2011-03-22 16:29:53 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum: simplify the have_interval calculation
	  Move some of the conditions to the places where the dependent variables change.

2011-03-22 16:26:45 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: use local var for input_data function
	  Avoid dereferencing the input_data from the instance from within an inner loop.

2011-03-23 16:34:16 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexdec.h:
	  speexdec: Get and use streamheader from the caps if possible
	  This allows playback of streams where the streamheader buffers
	  were dropped from the stream for some reason.

2011-03-22 19:36:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: use running time for synchronization
	  Fixes #432612.

2011-03-22 19:36:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: use running time for synchronization
	  Fixes #432612.

2011-03-22 19:35:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: use running time for synchronization
	  See bug #432612.

2011-03-22 12:53:22 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* configure.ac:
	  configure.ac: redundant uses of AC_MSG_RESULT()
	  cleaned the redundant uses of AC_MSG_RESULT() in configure.ac

2011-03-18 19:34:57 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* autogen.sh:
	  autogen: wingo signed comment

2011-03-16 10:43:47 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	  jackaudiosink: Fix typo from 9544622674c0d0a3147a9b51145159b02eec68e9

2011-03-16 09:38:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: Mark tag mapping tables as static const

2011-03-16 09:37:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Use ARTIST instead of AUTHOR for GST_TAG_ARTIST

2011-03-16 09:35:50 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Use ARTIST Matroska tag instead of AUTHOR for GST_TAG_ARTIST
	  AUTHOR only existed in an old version of the spec and ARTIST is
	  the new replacement for this. We are still reading both to still
	  be compatible with old files.
	  Fixes bug #644875.

2011-03-15 20:19:48 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/videofilter.c:
	  tests: enable more formats in videofilter unit test, check more resolutions

2011-03-14 19:14:07 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Fix buffer overflow bug for odd resolutions and Y422 colorspaces
	  https://bugzilla.gnome.org/show_bug.cgi?id=644773

2011-03-15 19:36:01 +0200  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: silence warning message when appropriate
	  If we did not know how many frames to expect, then we get an unexpected
	  end of stream when trying to decode more frames that are there, if there
	  are leftover bits to pad to the next byte

2011-03-14 19:14:07 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Add support for YUY2, UVYV and YVYU colorspaces
	  https://bugzilla.gnome.org/show_bug.cgi?id=644773

2011-03-15 09:43:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/videofilter.c:
	  tests: in videofilter unit test also check with 'odd' widths and heights
	  And only use one test suite.

2011-03-14 19:28:07 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: Always process the number of frames per packet as specified in the header
	  Looking at the remaining bits in the bitstream after decoding a
	  single frame can't be used as loop condition. The remaining
	  bits might not give a complete frame and the speex decoder will
	  then output nothing but access uninitialized memory, which leads
	  to valgrind warnings.
	  Fixes bug #644669.

2011-03-14 15:46:50 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: return TRUE from sink pad event function for tag events, which are handled
	  https://bugzilla.gnome.org/show_bug.cgi?id=644730

2011-03-12 00:44:31 +0530  Philip Jägenstedt <philipj@opera.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Better fix for deadlock on failed connect
	  This reverts the previous fix that would cause a double-unlock when the
	  stream connect failed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=644510

2011-03-11 23:06:31 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Fix deadlock if connecting to PA fails
	  Commit dd4ec22e introduced a deadlock in the failure path while trying
	  to connect to PulseAudio. This makes sure we drop the lock on the
	  resource mutex to avoid this.
	  https://bugzilla.gnome.org/show_bug.cgi?id=644510

2011-03-11 16:59:10 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/Makefile.am:
	  tests: order state-test blacklist and add jack elements
	  Jack audio src/sink elements recently got moved from bad and should be excluded
	  from the test (like the other device specific source and sinks).
	  Fixes #644288

2011-03-11 13:47:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Chain up to the parent class' ::send_event for non-seek events

2011-03-11 13:46:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Fix refcount issues with the seek event
	  Fixes bug #642963.

2011-03-11 09:54:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  docs: fix pulsesink gtk-doc markup

2011-03-11 10:29:08 +0100  Philippe Normand <pnormand@igalia.com>

	* configure.ac:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: fix build against jack 0.120.2
	  jack_port_get_total_latency() has been deprecated in favor of
	  jack_port_get_latency_range().
	  https://bugzilla.gnome.org/show_bug.cgi?id=644477

2011-03-10 14:29:25 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: more comments and tune and logging

2011-03-10 14:15:42 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: avoid unneccesary extra fft runs
	  Before it was possible that we run an extra fft when the time for sending a new
	  message is due. Only do this if we have not run the fft for the interval at all.

2011-03-10 14:12:01 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: only scale the vectors that we are processing
	  Phase is not produced by default, so lets not scale it unconditionally to save a
	  few cycles.

2011-03-10 14:10:25 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum: put number of channels to instance variable
	  When freeing data the format might have changed. Thus we need to remember for
	  which format we allocated memory.

2011-03-10 10:27:14 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: update doc review stamp

2011-03-10 10:22:29 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum: use function pointers for data readers
	  Don't check the format for each sample frame to read. We can make that decission
	  in _setup already. This is still not ideal as we call the function per frame.
	  Ideally we determine how many samples we can copy and have a loop in the input
	  reader. As an alternative we might also consider to use the fft variants for the
	  various formats and not convert to float for all cases - we would still need to
	  mix or deinterleave though.

2011-03-09 17:07:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: improve recovery from failed seek
	  In case server-side fails to perform seek, i.e. PLAY at non-zero requested
	  position, recovery so far would arrange for streaming to continue, albeit
	  having lost position tracking in the process.  So, query position prior
	  to seek and use upon failed seek.

2011-03-09 16:51:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: handle position query

2011-03-09 16:57:28 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum:  multi-channel support
	  Add a boolean multi-channel property with a default of FALSE. When set to TRUE
	  the element won't mix all input channels to mono, but instead run a FFT on each
	  channel. In that case the result message would contain a 2 dimensional array
	  of channel x data for magnitude and phase.
	  API: GstSpectrum:multi-channel
	  https://bugzilla.gnome.org/show_bug.cgi?id=593482

2011-03-09 16:55:56 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: more xrefs in the docs

2011-03-09 12:41:15 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: factor out the code that accumulated samples into the ring-buffer
	  Use a separate function to read a sample frame into a ringbuffer slot. In the
	  future we can use format-specific function pointer to avoid the reoccuring
	  format checks.

2011-03-09 12:38:52 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: pull format to temp var to improve readability of lines using it

2011-03-09 12:20:11 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: code cleanup for copying data to ring-buffer
	  Rename fp to is_float and restructure if-else part for handling the different formats.

2011-03-09 11:40:48 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum: add a GstSpecrtumChannel context structure
	  We now keep the fft data that is related to one channel in a separate structure
	  to prepare for multichannel support. We also refactor the code to operate more
	  often on the channel context.

2011-03-09 11:18:19 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: call the instance var spectrum instead of filter

2011-03-09 11:14:37 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: don't value we already took from the gvalue

2011-03-08 17:26:17 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  configure.ac

2011-03-08 17:02:30 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/debugutils/efence.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/ximage/ximageutil.c:
	  meta: update for new API

2011-03-08 16:28:27 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge ad-hoc release branch '0.10.28'

=== release 0.10.28 ===

2011-03-08 15:47:52 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.28
	  Ad-hoc release to fix build issue with newer kernels.

2011-03-03 00:16:47 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/v4l2_calls.h:
	  v4l2: remove unnecessary linux/videodev.h include
	  Causes compilation issues with newer kernel headers where the old
	  v4l interface has been removed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=643716

2011-03-08 10:14:20 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  tests/examples/cairo/Makefile.am

2011-03-07 16:56:43 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: also estimate eos if very near eos

2011-03-07 16:56:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: avoid trying to buffer more than is available.
	  That is, in case of short (or near eos of) stream, deadlock (until timeout)
	  would occur trying to buffer more than is yet forthcoming.

2011-03-07 11:01:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reset element base_time upon flush
	  ... to arrange for properly scheduled timeout (following seek).

2011-03-07 10:54:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/cairo/cairo_overlay.c:
	  cairooverlay: Add a bus handler to the example to handle EOS/ERROR/WARNING
	  Also clean up the pipeline properly.

2011-03-07 10:47:23 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/Makefile.am:
	  examples: Always dist the cairo example

2011-03-07 10:46:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/cairo/Makefile.am:
	  cairooverlay: Use LDADD instead of LDFLAGS for libs and add $(GST_LIBS)

2011-03-05 23:22:58 +0000  Jon Nordby <jononor@gmail.com>

	* tests/examples/Makefile.am:
	* tests/examples/cairo/Makefile.am:
	* tests/examples/cairo/cairo_overlay.c:
	  cairooverlay: Remove unnecessary gtk/gtk-x11 use in example.
	  This removes code, and allows the example to be used on any platform.
	  Fixes bug #643981.

2011-03-04 18:37:38 -0800  David Schleef <ds@schleef.org>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Use #ifdefs for V4L2_PIX_FMT_PJPG
	  It's only recently added to kernel headers.

2011-02-23 16:50:43 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: tune output max buffer size to material
	  ... to avoid ending up with tons of short time buffers for e.g. high sample
	  rate audio.

2011-03-04 17:04:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/cairo/Makefile.am:
	  examples: don't use hardcodec 0.10

2011-03-04 16:30:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-03-04 15:50:01 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: add a doc example for setting stream-properties

2011-03-04 15:42:19 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix the xml in the docs

2011-03-03 00:16:47 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/v4l2_calls.h:
	  v4l2: remove unnecessary linux/videodev.h include
	  Causes compilation issues with newer kernel headers where the old
	  v4l interface has been removed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=643716

2011-03-02 23:21:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* tests/examples/Makefile.am:
	* tests/examples/cairo/Makefile.am:
	* tests/examples/cairo/cairo_overlay.c:
	  cairooverlay: The example always requires gtk-x11
	  Check for gtk-x11 and only build the example if it's available.

2011-03-02 23:14:36 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairooverlay.h:
	  cairooverlay: Some minor cleanup

2011-03-02 23:09:21 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	  docs: Update inspected plugin data

2011-01-28 02:14:04 +0200  Jon Nordby <jononor@gmail.com>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* ext/cairo/.gitignore:
	* ext/cairo/Makefile.am:
	* ext/cairo/gstcairo-marshal.list:
	* ext/cairo/gstcairo.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairooverlay.h:
	* tests/examples/Makefile.am:
	* tests/examples/cairo/.gitignore:
	* tests/examples/cairo/Makefile.am:
	* tests/examples/cairo/cairo_overlay.c:
	  cairooverlay: Add generic Cairo overlay video element.
	  Allows applications to connect to the "draw" signal of
	  the element and do their custom drawing there.
	  Includes an example application demonstrating usage.
	  Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=595520

2011-03-02 13:00:31 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/monoscope/monoscope.c:
	  monoscope: don't leak the monoscope_state data
	  The monoscope_close() implementation was empty.

2011-03-02 12:59:35 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/monoscope/monoscope.c:
	  monoscope: we have 64 colors, don't access colors[64]
	  Fixes remaining invalid read.

2011-03-02 10:25:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: arrange for non-fatal error when parsing non-vital parts

2011-03-02 10:56:33 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/monoscope/convolve.c:
	  monoscope: stack needs to be size+1 as we put a end-marker into it
	  Valgrind is still complaining about one bad read, but this takes care of the
	  crash mentioned in the comment and in bug #564122.

2011-03-01 22:40:19 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	  example: fix the variable name for the ip-address
	  Fix the name in the launch pipeline and use a value of "localhost" by default.

2011-02-28 19:16:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	  configure.ac: cygwin/mingw; enable plugin linking to static lib
	  Useful for DirectX plugin(s).
	  Fixes #642507.

2011-02-28 19:13:41 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	  configure.ac: export plugin description more platform independent
	  Fixes #642504.

2011-02-28 18:32:54 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 1de7f6a to 6aec6b9

2011-02-28 13:29:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-02-28 13:28:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: use NetAddress metadata

2011-02-28 13:14:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstudp.c:
	* gst/udp/gstudpsrc.c:
	  udp: implement NetAddress with metadata

2011-02-28 10:16:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: register metadata

2011-02-27 19:43:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/debugutils/efence.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/ximage/gstximagesrc.c:
	* sys/ximage/ximageutil.c:
	* sys/ximage/ximageutil.h:
	  meta: fix for new API

2011-02-25 16:29:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/debugutils/efence.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/ximage/gstximagesrc.c:
	* sys/ximage/ximageutil.c:
	* sys/ximage/ximageutil.h:
	  metadata: use metadata for private buffer data
	  Use buffer metadata to store element private data.

2011-02-24 13:51:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	* sys/ximage/ximageutil.c:
	* sys/ximage/ximageutil.h:
	  miniobject: port to 0.11
	  Use buffer private data instead of subclassing.

2011-02-24 13:50:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/pulse/Makefile.am:
	* tests/examples/v4l2/Makefile.am:
	* tests/icles/Makefile.am:
	  build: don't hardcode version number

2011-02-24 13:03:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/taglib/gstid3v2mux.cc:
	  id3: use boxed type instead of miniobject

2011-02-24 13:00:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/debugutils/efence.c:
	* gst/replaygain/Makefile.am:
	* gst/rtpmanager/rtpsession.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstudp.c:
	* gst/udp/gstudpsrc.c:
	  miniobject: use buffer private field for extra data
	  Use the owner private field to store extra buffer data instead of using
	  subclassing.

2011-02-24 12:23:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: add duration when extimating QoS time
	  When we need to decide on the next QoS time, take into account the duration of
	  the buffers.

2011-02-28 11:58:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  configure.ac

2011-02-23 17:41:22 +0100  Philip Jägenstedt <philipj@opera.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: release pa_shared_resource_mutex before pa_threaded_mainloop_wait
	  Not doing so can result in a deadlock when two threads enter
	  gst_pulseringbuffer_open_device at the same time, as
	  pa_threaded_mainloop_wait releases the mainloop lock while waiting,
	  allowing another thread to take it, resulting in a deadlock as two
	  threads waits for the lock the other is holding.
	  https://bugzilla.gnome.org/show_bug.cgi?id=643087

2011-02-23 17:18:19 +0100  Philip Jägenstedt <philipj@opera.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: s/ressource/resource/
	  https://bugzilla.gnome.org/show_bug.cgi?id=643087

2011-02-25 20:12:35 -0800  David Schleef <ds@schleef.org>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: remove accidental debug message
	  in previous commit

2011-02-25 19:35:51 -0800  David Schleef <ds@schleef.org>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Add support for 2Vuy and r210

2011-02-24 14:08:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Add support for NV21 colorspace

2011-02-24 14:00:37 +0100  Carsten Kroll <car@ximidi.com>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Add support for NV12 colorspace
	  Fixes bug #642961.

2011-02-24 13:56:04 +0100  Carsten Kroll <car@ximidi.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: First try if upstream handles TIME seeks before handling them here
	  Fixes bug #642963.

2010-11-08 14:25:59 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Simplify setcaps
	  The current code never uses upstream negotiation so the code can be
	  significantly simplified.

2011-01-24 12:48:18 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/tvtime/greedy.c:
	  deinterlace: Port greedyl to GstDeinterlaceSimpleMethod
	  The main goal of this change is to reuse the complex but now neatly
	  written scanline pointer calculation code from the simple methods.

2011-02-22 15:20:11 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/id3demux/gstid3demux.c:
	  Revert "id3demux: ensure a taglist before adding the container tag"
	  This reverts commit a86bab66893bb1a3323a756410573c117b8219ef. The issue is
	  fixed with commit ff5e5a8f0daa1fdf89792d0726ea063bbd99db18 instead.

2011-02-22 15:19:00 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/id3demux/id3tags.c:
	  id3demux: return ID3TAGS_BROKEN_TAG for unsupported versions
	  This prevents us for trying to work with a NULL taglist.

2011-02-22 14:15:27 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix unitialized variable.

2011-02-22 14:01:27 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: ensure sane parameters when parsing superindex

2011-02-22 14:00:11 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: check for NULL audio stream format header when parsing stream

2011-02-22 14:52:18 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  rtp-examples: move capsfilter behind converters
	  We need to have the capsfilter behin the converters to make the converters
	  convert from the formats v4l2src can do to what we request with the
	  capsfilter.

2011-02-22 14:50:59 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-PCMA.sh:
	* tests/examples/rtp/server-alsasrc-PCMA.sh:
	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  rtp-examples: fix ascii-art
	  Some boxes where misaligned due to long "audiotetssrc" name. Trim trailing
	  whitespace.

2011-02-22 13:29:26 +0100  Blaise Gassend <blaise at willowgarage dot com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: handle NULL demux elements
	  When using gstrtpbin with ignore-pt=true, the free_stream function tries to
	  call gst_element_set_locked_state and gst_element_set_state on a stream->demux
	  which is NULL.
	  fixes #642412

2011-01-24 12:18:39 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: small clean-ups
	  Improve debug output by printing the buffer pointer when
	  popping a buffer and simplify code to use scanlines.bottom_field
	  as appropriate.
	  https://bugzilla.gnome.org/show_bug.cgi?id=642691

2011-01-24 12:18:39 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix assigned method_id when using fallback
	  https://bugzilla.gnome.org/show_bug.cgi?id=642691

2011-02-21 17:17:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix setting the SDES property
	  Only the sdes veriable is protected with the object lock.
	  Use the right object when setting the sdes property.

2011-02-21 12:09:07 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	* gst/avi/gstavimux.c:
	* gst/flv/gstflvmux.c:
	* gst/interleave/interleave.c:
	* gst/matroska/matroska-mux.c:
	* gst/videomixer/videomixer.c:
	  Revert "Check that collectpads exists before removing pad"
	  This reverts commit 8e6b876e76c94410db160afe5eb30f21452e419f.
	  Depends on a core commit that was reverted

2011-02-21 00:55:49 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/icydemux/gsticydemux.c:
	  icydemux: fix tag list handling issues that might have caused crashes
	  Fix slightly confused tag handling in some places: make it clear when
	  we're taking ownership of a tag list and when not. For example,
	  gst_icydemux_tag_found() was taking ownership when the source pad
	  existed, but otherwise not (leak). Also, gst_event_parse_tag() does
	  not return a newly-allocated taglist, but a tag list that belongs to
	  the tag event, so don't give ownership of it away.
	  While we're at it, some minor clean-ups: don't re-invent g_strndup()
	  and simplify gst_icydemux_parse_and_send_tags() a bit, and don't
	  leak the tag list in case no valid tags where found.
	  https://bugzilla.gnome.org/show_bug.cgi?id=641330

2011-02-20 23:39:41 -0800  David Schleef <ds@schleef.org>

	* ext/cairo/gsttextoverlay.c:
	* gst/avi/gstavimux.c:
	* gst/flv/gstflvmux.c:
	* gst/interleave/interleave.c:
	* gst/matroska/matroska-mux.c:
	* gst/videomixer/videomixer.c:
	  Check that collectpads exists before removing pad
	  The core now calls release pad from finalize, at which point
	  the collectpads might have already been freed.

2011-02-19 15:48:22 -0800  David Schleef <ds@schleef.org>

	* ext/libpng/gstpngdec.c:
	  pngdec: Handle 16-bit-per-channel images

2011-02-18 10:12:47 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avidemux: stream->current_total is accumulated byte size and not time
	  Use timestamp for the stream index as well.

2011-02-15 19:33:45 -0800  David Schleef <ds@schleef.org>

	* gst/udp/gstmultiudpsink.c:
	  udpsink: warn when packet is too large

2011-02-17 17:59:25 -0800  David Schleef <ds@schleef.org>

	* gst/matroska/Makefile.am:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska.c:
	  matroskaparse: New element
	  Copied from demux.  Duplicates much code, also some dead code
	  remaining.

2011-02-17 17:57:55 -0800  David Schleef <ds@schleef.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Earlier debug category initialization

2011-01-22 00:13:16 -0800  David Schleef <ds@schleef.org>

	* gst/flv/gstflvmux.c:
	  flvmux: don't set duration for live stream

2011-01-06 15:44:24 -0800  David Schleef <ds@schleef.org>

	* gst/debugutils/Makefile.am:
	* gst/debugutils/negotiation.c:
	  debugutils: remove bitrotten negotiation element
	  Wasn't enabled, didn't work, and planned features have been
	  superceded by capsfilter and capsdebug.

2010-09-17 12:10:38 -0700  David Schleef <ds@schleef.org>

	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtp/gstrtpvrawpay.h:
	  rtpvrawpay: Implement interlacing

2011-02-17 17:57:42 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avidemux: also add the frame-type for the stream index

2011-02-17 17:56:29 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avidemux: get the index writer id when the pad has a parent
	  Otherwise the index writer has a weired name, as the pad has no parent yet.

2011-02-17 14:00:48 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	  avidemux, flvdemux: formatting cleanup
	  Trim trailing whitespaces and fix the formatting of double negation.

2011-02-17 13:57:37 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	  avidemux, flvdemux: mark delta-units in the index
	  We need to use the 'delta' flag for delta units and not the 'none' flag.

2011-02-17 11:58:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/icles/.gitignore:
	  .gitignore: ignore moved equalizer test binary

2011-02-17 12:46:14 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: mark delta-unit in the index
	  We need to use the delta flag fro delta units and not none. Print more details
	  to the debug log.

2011-02-17 12:44:01 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: formatting cleanup
	  Trim trailing whitespaces and fix the formatting of double negation.

2011-02-16 17:09:20 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: rework _request_new_pad to handle explict req-pad-names
	  Don't ignore explicit pad-names.

2011-02-16 17:06:51 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	  avimux: rework _request_new_pad to handle explict req-pad-names
	  Don't ignore explicit pad-names. Rearrange the code and the error handling a
	  bit. Add a FIXME-0.11 for the bad pad-names.

2011-02-16 15:28:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/icles/Makefile.am:
	  icles: Add equalizer-test to the build system

2011-02-16 15:23:50 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/icles/equalizer-test.c:
	  [MOVED FROM BAD 5/5] equalizer-test: Initialize debug category after gst_init() to fix segfault

2007-11-07 15:36:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 4/5] tests/icles/equalizer-test.c: Fix gain ranges for the latest equalizer changes.
	  Original commit message from CVS:
	  * tests/icles/equalizer-test.c: (do_slider_fiddling):
	  Fix gain ranges for the latest equalizer changes.

2007-05-21 14:01:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD 3/5] ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBa...
	  Original commit message from CVS:
	  * ChangeLog:
	  ChangeLog surgery.
	  * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN,
	  _GstIirEqualizerBand, object, _GstIirEqualizerBandClass,
	  parent_class, gst_iir_equalizer_band_set_property,
	  gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type,
	  gst_iir_equalizer_child_proxy_get_child_by_index,
	  gst_iir_equalizer_child_proxy_get_children_count,
	  gst_iir_equalizer_child_proxy_interface_init, setup_filter,
	  gst_iir_equalizer_compute_frequencies, plugin_init):
	  * tests/icles/equalizer-test.c:
	  Add fixme and comment for example.

2007-03-14 16:33:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD 2/5] tests/icles/equalizer-test.c: Port the example to new equalizer api.
	  Original commit message from CVS:
	  * tests/icles/equalizer-test.c: (equalizer_set_band_value),
	  (equalizer_set_all_band_values),
	  (equalizer_set_band_value_and_wait),
	  (equalizer_set_all_band_values_and_wait), (do_slider_fiddling),
	  (main):
	  Port the example to new equalizer api.

2007-02-03 23:35:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  [MOVED FROM BAD 1/5] Fix up to use the newly ported (actually working) GstAudioFilter.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/equalizer/Makefile.am:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init),
	  (gst_iir_equalizer_class_init), (gst_iir_equalizer_init),
	  (setup_filter), (gst_iir_equalizer_compute_frequencies),
	  (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
	  (gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup),
	  (plugin_init):
	  * gst/equalizer/gstiirequalizer.h:
	  Fix up to use the newly ported (actually working) GstAudioFilter.
	  Bump core/base requirements to CVS for this.
	  * tests/icles/.cvsignore:
	  * tests/icles/Makefile.am:
	  * tests/icles/equalizer-test.c: (check_bus),
	  (equalizer_set_band_value), (equalizer_set_all_band_values),
	  (equalizer_set_band_value_and_wait),
	  (equalizer_set_all_band_values_and_wait), (do_slider_fiddling),
	  (main):
	  Add brain-dead interactive test for equalizer.

2011-02-15 15:59:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add PJPG mapping
	  Adds mapping of progressive jpeg format

2011-02-15 16:30:20 +0100  Andy Wingo <wingo@oblong.com>

	  plug qtdemux refcount leaks
	  * gst/qtdemux/qtdemux.c (gst_qtdemux_src_convert): Unref the qtdemux; we
	  weren't doing so before.
	  (gst_qtdemux_handle_src_event, gst_qtdemux_chain): Fix some error
	  cases which would leak a ref to the qtdemux.

2011-02-14 20:20:08 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Add URI query handler
	  Fixes bug #642337.

2011-02-14 17:49:54 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: avoid sorting NULL array of cluster positions

2011-02-14 16:46:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	  theorapay: handle 0 sized packets
	  Handle 0 sized packets (repeat frame) in the payloader and depayloader.
	  Fixes #641827

2011-02-14 15:21:29 +0200  Tuukka Pasanen <tuukka.pasanen@ilmi.fi>

	* gst/debugutils/gsttaginject.c:
	  taginject: resend tags when they are changed
	  Allow setting new tags on the property while running and send them.
	  Fixes #640249

2011-02-14 12:53:27 +0200  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From f94d739 to 1de7f6a

2011-02-07 23:32:53 +0100  Miguel Angel Cabrera Moya <madmac2501@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix minor leaks when handling server requests.
	  https://bugzilla.gnome.org/show_bug.cgi?id=640163

2011-02-14 00:49:00 +0000  Heath Nielson <heathn@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: extract MusicBrainz tags
	  Extract MusicBrainz tags added by MusicBrainz's Picard
	  tagger application. These tags (esp. the album id) are
	  helpful for rhythmbox et.al. to automatically downloads
	  cover art.
	  https://bugzilla.gnome.org/show_bug.cgi?id=642205

2011-02-14 00:38:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: refactor iTunes tag parsing a bit

2011-02-10 23:52:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-plugins-good.doap:
	  doap: update mailing list location

2011-02-10 18:11:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: propagate error during expose_streams
	  ... as it may occur during initial parsing of fragmented file.

2011-02-10 18:00:11 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: avoid skipping exposing a stream following a removed stream

2011-02-10 11:56:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: store cluster positions provided by SeekHead
	  ... and use those, if available, to locate a cluster rather than scanning.

2011-02-09 16:22:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: properly resume cluster scanning
	  ... rather than getting offset tracking messed up, and then likely
	  failing a subsequent assert.

2011-02-08 10:07:43 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/id3demux/gstid3demux.c:
	  id3demux: ensure a taglist before adding the container tag
	  In the case of id3v1 also don't return NULL on empty tags, but also create a new
	  taglist and add the container tag for consistency.

2011-02-07 17:08:47 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: strip trailing spaces

2011-02-07 17:07:42 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/rtsp/gstrtspsrc.c:
	  rtpsrc: set multiple properties in one go
	  There is no need for separate g_object_set() calls here.

2011-02-03 16:10:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* tests/check/elements/deinterlace.c:
	  deinterlace: Handle image caps without asserting
	  Images might have framerate=0/1 in the caps, which caused an
	  assertion on deinterlace. I don't know of interlaced image formats
	  but deinterlace might be hardcoded on some generic pipelines and
	  it shouldn't assert.
	  The fix was to set field_duration to 0 if the input has a framerate
	  with a 0 numerator.
	  This patch also adds checks for this situation on the unit tests.
	  https://bugzilla.gnome.org/show_bug.cgi?id=641400

2011-02-04 12:33:09 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/udp/gstudpsrc.c:
	  docs: fix parameter name in udpsrc docs
	  It is "buffer-size" and not "buffer". Also trim trailing whitespace.

2011-02-03 23:42:59 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: fix interlaced set_format configuration
	  Commit 6c8268dbfd5c88fac28c882ef2e4598a6522e2d6 broke recording
	  from interlaced v4l2 source (e.g. typical tv capture card) since
	  V4L2_FIELD_SEQ_TB (with fields stored separately) does not map
	  to currently defined interlaced format (fields stored interleaved).
	  Besides this mismatch, hardware might quite likely not support or
	  appreciate this field value, since querying supported formats mapped
	  _INTERLACED field formats to interlaced=true caps (so the latter should
	  not be mapped to field value that is not known to be supported).

2011-02-03 18:25:00 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/pipelines/lame.c:
	  tests: add unit test for lamemp3enc negotiation issue
	  https://bugzilla.gnome.org/show_bug.cgi?id=641151

2011-02-03 18:18:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: implement sinkpad get_caps() function to proxy rate and channels restrictions from downstream
	  The element downstream of mp3enc might only accept certain sample rates or channels,
	  make sure we relay any restrictions that do exist to upstream when it does a
	  get_caps() on the sink pad. That way upstream elements like audioresample or
	  audioconvert can pick a sample rate / channel configuration that will be accepted,
	  instead of just negotiating to the highest, which might then be rejected.
	  https://bugzilla.gnome.org/show_bug.cgi?id=641151

2011-02-02 18:27:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  source: fix type of ntpnstime

2011-02-02 18:21:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpbin: Get and use the NTP time when receiving RTCP
	  When we receive an RTCP packet, get the current NTP time in nanseconds so that
	  we can correctly calculate the round-trip time.

2011-02-01 19:40:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	  directsound: arrange for definition of _swab on Cygwin
	  gstdirectsoundsink.c: In function 'gst_directsound_sink_write':
	  gstdirectsoundsink.c:557: error: implicit declaration of function '_swab'
	  gstdirectsoundsink.c:557: error: nested extern declaration of '_swab'

2010-10-06 21:17:28 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheoradepay.h:
	  rtptheoradepay: Request new keyframe on lost packets
	  Theora can only use the last frame (or the keyframe) as a reference, so in
	  practice. If we receive a buffer that references an unknown codebook, request
	  new headers. It probably means that headers were lost.

2010-08-27 14:11:53 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Add action signal to request early RTCP

2010-08-27 16:11:06 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Add callback to get the current time

2010-10-19 22:21:54 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Don't relay more than one PLI request per RTT
	  Drop PLI requests if one was relay in the last RTT, the other side may
	  just not have received the keyframe yet.

2010-06-23 16:43:24 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Send GstForceKeyUnit event in response to received RTCP PLI

2010-11-24 15:27:46 -0500  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: Fallback for FIR to PLI if PLI isn't available

2010-06-22 19:56:50 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Implement sending PLI packets in response to GstForceKeyUnit

2010-06-22 13:33:32 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsource: Retain RTCP Feedback packets for a specified amount of time

2010-09-07 13:35:16 +0300  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Make rtcp buffer metadata writable after processing it
	  Functions that process the rtcp buffer could decide to keep a ref
	  on the buffer for further processing. So make the metadata writable
	  only after they are done.

2010-06-17 17:34:19 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Emit signal on incoming RTCP FB packet

2011-02-01 18:17:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: fix compilation

2010-06-15 18:39:47 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Add method to request early RTCP packet
	  Implement the early mode defined in RFC 4585. In this mode, RTCP feedback
	  packets are sent early to notifier.

2010-06-01 19:28:01 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpsession: Add property for minimum interval between Regular RTCP messages
	  This can be changed according to RFC 4585

2010-06-14 18:40:33 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Emit signal when sending a compound RTCP packet
	  This allows users to add extra RTCP packets to the compound
	  RTCP packet.

2010-06-19 19:11:06 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: Tag upstream custom events with payload type

2010-06-18 19:12:40 -0400  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Tag upstream custom events with SSRC

2010-10-01 17:19:16 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Emit "on-ssrc-validated" when validating by RTCP
	  Emit "on-ssrc-validated" if the SSRC is validated by receiving
	  a RTCP SDES packet.

2011-02-01 16:38:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kpay.c:
	  j2kpay: skip EPH packets
	  Include EPH markers into the previous chunk of packets.

2011-01-31 17:56:18 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	  rtppcmapay: Rename the class to have the right name
	  It was name pmca instead of pcma and made debug logs hard to search.

2011-01-31 05:58:36 +0100  David Henningsson <david.henningsson@canonical.com>

	* ext/pulse/pulsesink.c:
	  Pulsesink: Allow chunks up to bufsize instead of segsize
	  By allowing larger chunks to be sent, PulseAudio will have a
	  lower CPU usage. This is especially important on low-end machines,
	  where PulseAudio can crash if packets are coming in at a higher
	  rate than PulseAudio can process them.
	  Signed-off-by: David Henningsson <david.henningsson@canonical.com>

2011-01-31 13:44:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: simplify template caps
	  We can merge all the YUV variants into one single structure.

2011-01-27 15:35:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  win32: fix DEFAULT_AUDIOSINK, should be direct*sound*sink
	  https://bugzilla.gnome.org/show_bug.cgi?id=640705

2011-01-27 16:02:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: initialize local variable to please mingw32 compiler

2011-01-26 22:21:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpnetutils.h:
	* gst/udp/gstudpsrc.c:
	  udp: use socklen_t where appropriate rather than custom type
	  In particular, fixes Cygwin build where socklen_t is defined as int
	  in line with native win32 api definition.

2011-01-27 12:16:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: mind rounding issues when converting from global time to mov time
	  In particular, this avoids missing the intended keyframe when first converting
	  from the frame's mov time to global segment time, and then back from global
	  time to mov time when activating the segment.

2011-01-26 08:48:43 +0000  Ognyan Tonchev <ognyan.tonchev@axis.com>

	* gst/matroska/ebml-write.c:
	* tests/check/elements/matroskamux.c:
	  matroskamux: don't leak ebml writer caps when re-using matroskamux
	  https://bugzilla.gnome.org/show_bug.cgi?id=640542

2011-01-25 21:56:19 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: don't divide by 0

2011-01-18 14:48:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: pull mode should always report seekable
	  ... as it no longer requires an index, but can seek by scanning as well.

2011-01-10 12:34:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: support some more mpeg-4 fourcc variants

2011-01-10 12:34:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: simplify retrieving stsd child entry atom

2011-01-24 18:27:52 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Don't consider 0 fcc_handler as uncompressed.
	  Just avoids a warning

2011-01-20 12:14:08 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: take configured start time into account
	  when creating the newsegment event, take the configured start time
	  into account.

2011-01-24 15:11:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix printf format warning on mingw32
	  Make win32 build bot happy again, and nicefy output while we're at it.
	  qtdemux.c: In function 'qtdemux_parse_trun':
	  qtdemux.c:2162:3: error: format '%lu' expects type 'long unsigned int', but argument 9 has type 'guint32'

2011-01-24 13:39:58 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/rtp/client-H263p-AMR.sh:
	* tests/examples/rtp/client-H263p-PCMA.sh:
	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-PCMA.sh:
	  examples: autoaudisink -> autoaudiosink in RTP examples

2011-01-24 00:32:41 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development

=== release 0.10.27 ===

2011-01-21 12:54:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.27

2011-01-20 14:10:55 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: don't leak codec data buffer in byte-stream=true mode
	  https://bugzilla.gnome.org/show_bug.cgi?id=640063

2011-01-20 13:41:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't leak url string
	  https://bugzilla.gnome.org/show_bug.cgi?id=640064

2011-01-20 11:45:47 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Gracefully handle mov files misusing the WAVE atoms
	  Check that the WAVEHEADER node is present instead of blindly using it.
	  If not present we won't be able to provide a more refined caps, but at
	  least we won't crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=640028

2011-01-20 00:07:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: fix accidental breakage of navigation interface support

2011-01-18 12:58:29 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.26.4 pre-release

2011-01-12 14:03:57 -0800  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: rewrite how neighboring scan lines are calculated
	  Old code was difficult to understand exactly how the neighboring
	  scan lines are calculated, and it appeared that some were off by
	  +2 or -2, depending on the field flag.  Fixes #639321.

2011-01-18 09:33:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavisubtitle.c:
	  avisubtitle: set caps on srcpad to fix issue with discoverer
	  Set caps from the start so discoverer doesn't blow up on
	  seeing no negotiated caps between elements on preroll,
	  which might happen if no subtitle buffers have been
	  pushed yet at the time. See file from bug #603308.

2011-01-17 20:09:16 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Uncork stream while flushing the ringbuffer
	  After starting the ringbuffer, we wait for enough data to arrive before
	  uncorking the stream. This will cause the pipeline to stall if we get an
	  EOS (or otherwise need to flush the stream) before sufficient data
	  becomes available. This patch makes sure that the stream is uncorked
	  while flushing to avoid this problem.
	  Fixes issue with a webkit unit test testing reverse playback of
	  an MP4 H.264/AAC file.
	  https://bugzilla.gnome.org/show_bug.cgi?id=639740

2011-01-14 14:51:51 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: avoid creating caps from string when possible
	  Fixes #639516.

2011-01-14 14:48:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: set src pad caps when starting file
	  Fixes #639516.

2011-01-12 20:38:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: define V4L2_FIELD_INTERLACED_{TB,BT} if not available in header
	  Older kernels don't have these, and there's no easy way to check for the
	  existance of enums that doesn't involve a configure check, so just define
	  these if the V4L2_CAP_VIDEO_OUTPUT_OVERLAY define is not there, which was
	  added in the same commit as the TB/BT enum. Fixes compilation on CentOS 5.
	  https://bugzilla.gnome.org/show_bug.cgi?id=639339

2011-01-11 23:18:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.26.3 pre-release

2011-01-11 22:42:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update docs

2011-01-11 23:39:12 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Make corking during pause synchronous
	  This makes the call to pa_stream_cork() during ringbuffer pause()
	  synchronous, which makes sure that the clock does not advance after we
	  take a snapshot for start_time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=639240

2011-01-11 19:33:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/da.po:
	* po/gl.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/sl.po:
	* po/sv.po:
	* po/tr.po:
	  po: update translations

2011-01-11 15:50:28 +0200  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From e572c87 to f94d739

2011-01-10 16:36:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From ccbaa85 to e572c87

2011-01-10 14:53:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 46445ad to ccbaa85

2011-01-07 13:24:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.26.2 pre-release

2011-01-07 13:06:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update translations

2011-01-07 02:32:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: fix compiler warnings caused by -DG_DISABLE_ASSERT

2011-01-07 02:06:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	  matroska: don't put essential function calls into g_assert()
	  g_assert() will expand to NOOPs if -DG_DISABLE_ASSERT is passed.

2011-01-07 01:35:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: don't put functional code like ioctl calls into g_return_if_fail()
	  These macros will expand to NOOPs given the right defines. Also,
	  g_return_if_fail() and friends are meant to be used to catch programming
	  errors (like invalid input to functions), not runtime error handling.

2011-01-07 01:11:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	  tests: never disable g_assert() and cast checks for the unit tests
	  The unit tests are riddled with g_assert() and friends, make sure we
	  don't disable assert and cast checks for the unit tests even if
	  this has been specified for the rest of the code base, e.g. via
	  --disable-glib-asserts.

2011-01-06 12:29:21 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	  rtp: Fix unitialized variables on macosx

2011-01-06 12:28:58 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/qtdemux/qtdemux_dump.c:
	  qtdemux: Fix unitialized variables on macosx

2011-01-05 17:49:16 -0800  David Schleef <ds@schleef.org>

	* gst/debugutils/gstcapsdebug.c:
	  capsdebug: Add capdebug debug category

2010-12-11 12:42:10 -0800  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Change the default to linear
	  The previous default, greedyh, takes 4 times as long as MPEG-2
	  video decoding, and is unlikely fast enough on any current CPU
	  to play 1080i video in real-time.  greedyl isn't much faster.
	  linear was chosen over vfir, since the quality advantage of vfir
	  is minimal compared to the occasional visual artifacts and slower
	  processing.

2011-01-05 18:32:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't confuse return values
	  Return a return value of the right type.

2011-01-05 16:24:13 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_dump.c:
	  qtdemux: Fix unitialized variables on macosx

2011-01-05 15:03:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  vrawdepay: fix length check
	  Add some more debugging.
	  Add the length check so we don't cause unneeded warnings.

2011-01-05 12:04:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multiudpsink: add buffer-size property
	  Add buffer-size property to configure the kernel send buffer.

2011-01-03 20:16:22 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove unused variables when debug-logging disabled

2011-01-03 20:06:35 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: remove unused variables when debug-logging disabled

2011-01-03 18:05:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/libcaca/gstcacasink.c:
	  cacasink: fix masks and strides
	  Use the right endianness to read the masks.
	  Use the right strides for the bitmap.
	  Fixes #638569

2011-01-03 01:18:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: undo presumably accidental enablement of the GstXOverlay interface
	  Looks like this got enabled by accident when adding it to v4l2sink,
	  so undo this for now. Not sure it makes much sense in a GStreamer
	  context with current hardware.

2011-01-03 15:40:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: increase udp buffer size
	  Set a bigger UDP buffer size by default to reduce packet loss with
	  high bitrate streams.

2011-01-02 19:19:27 -0800  David Schleef <ds@schleef.org>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: send stream headers in key-frame mode

2011-01-02 19:43:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jack/Makefile.am:
	* ext/jack/README:
	* ext/jack/gstjack.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: fix up element details and some other minor clean-ups

2011-01-02 19:23:51 +0000  Erich Schubert <erich@debian.org>

	* gst/id3demux/id3v2frames.c:
	  id3demux: fix parsing of ID3v2.4 genre frames with multiple genres
	  We'd only extract the first genre (multiple times) instead of all
	  genres.
	  https://bugzilla.gnome.org/show_bug.cgi?id=638535

2011-01-02 17:40:41 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: template caps had lists with one value, just use value directly

2011-01-02 17:07:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: make get_type functions thread-safe
	  Because we can (shouldn't be needed with other workarounds still there).

2011-01-02 15:27:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	  docs: update plugin docs

2011-01-02 15:25:41 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* .gitignore:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-jack.xml:
	* ext/Makefile.am:
	* gst-plugins-good.spec.in:
	* tests/examples/Makefile.am:
	* tests/examples/jack/Makefile.am:
	  jack: new jackaudiosrc and jackaudiosink elements, moved from gst-plugins-bad
	  https://bugzilla.gnome.org/show_bug.cgi?id=621929

2010-10-19 16:23:23 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  various (ext): add missing G_PARAM_STATIC_STRINGS flags
	  Canonicalize property names as needed.

2010-09-09 14:49:06 -0400  Tristan Matthews <le.businessman@gmail.com>

	* ext/jack/Makefile.am:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: added translatable text for server not found error

2010-09-06 17:17:54 -0400  Tristan Matthews <le.businessman@gmail.com>

	* tests/examples/jack/Makefile.am:
	* tests/examples/jack/jack_client.c:
	  examples: add test to demonstrate jack_client_t usage

2010-09-06 16:11:31 -0400  Tristan Matthews <le.businessman@gmail.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudioclient.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: added client property

2010-06-17 16:26:07 -0400  Tristan Matthews <tristan@sat.qc.ca>

	* ext/jack/gstjackbin.c:
	  jack: removed unused file gstjackbin.c
	  This is a 0.8 leftover.

2010-05-13 12:55:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjackaudiosrc.c:
	  jacksrc: make sure we always read nframes
	  Error out when we are asked to read a different size that what was configured as
	  the jack period size because that would mean something else is wrong.
	  Fixes #618409

2010-05-11 17:56:31 -0400  Tristan Matthews <tristan@sat.qc.ca>

	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: improve process_cb

2010-04-27 10:48:32 -0400  Tristan Matthews <tristan@tristan-laptop.(none)>

	* ext/jack/Makefile.am:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackutil.c:
	* ext/jack/gstjackutil.h:
	  jack: implement multichannel support correctly for jackaudiosrc
	  Fixes parts of bug #616541.

2010-04-27 11:21:16 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackringbuffer.h:
	  jack: remove empty dispose and finalize methods

2010-04-27 10:59:00 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: don't leak caps
	  Add dispose methods to clear caps.

2010-04-27 10:34:24 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: don't use GST_DEBUG_FUNCPTR for gobject vmethods

2010-03-24 15:59:53 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jack/gstjackaudiosrc.c:
	  jack: fix element name in section doc blob

2010-03-22 16:56:03 +0100  Benjamin Otte <otte@redhat.com>

	* ext/jack/gstjackaudiosrc.c:
	  Add -Wold-style-definition
	  and fix the warnings

2010-03-21 21:39:18 +0100  Benjamin Otte <otte@redhat.com>

	* ext/jack/gstjack.h:
	  Add -Wmissing-declarations -Wmissing-prototypes to configure flags
	  And fix all warnings

2010-03-18 17:30:26 +0100  Benjamin Otte <otte@redhat.com>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple

2009-10-12 09:06:37 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: ensure segtotal is at least 2
	  Not only adjust buffer-time and avoid segtotal=0, but instead ensure segtotal is
	  atleast 2. Do same change on jacksrc. We could also check the latency and buffer
	  time configured by the client and adjust buffer-time so that we get to the same
	  number of segments.

2009-10-12 00:51:27 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jack/gstjackaudiosink.c:
	  jack: don't crash in ringbuffer with SIGFPE on small buffer-times
	  Jack overrides user-specified latency-time with the one it gets from jack
	  itself. It also needs to adjust buffer-time somewhat to avoid segtotal being 0

2009-05-11 16:12:54 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jack/gstjackaudioclient.c:
	* ext/jack/gstjackaudiosink.c:
	  jack: when stopping playback, do one more cycle to flush the port. Fixes #582167
	  The gst_jack_audio_client_set_active() flags the port as deactivating and uses
	  a GCond to wait until the jack_process_cb() has run once more and cleared the
	  flag. This way the client zero's the buffer. This happens if one manyally go
	  to PAUSED and then to READY, while leting the mainloop run inbetween.

2009-03-16 11:21:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: Add new connection mode
	  Add a new connection mode to jacksrc and jacksink. In this new auto-force
	  connection mode jack will create as many ports as requested/needed in the
	  pipeline and will then connect as many physical ports as possible, possibly
	  leaving some ports unconnected.
	  Also get rid of some leftover g_print.
	  Fixes #575284.

2008-11-23 17:50:08 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/jack/: Query port latencies for sink/src delays.
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosink.c:
	  * ext/jack/gstjackaudiosrc.c:
	  Query port latencies for sink/src delays.
	  * ext/jack/gstjackbin.c:
	  No printf please.

2008-11-04 12:42:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Don't install static libs for plugins. Fixes #550851 for -bad.
	  Original commit message from CVS:
	  * ext/alsaspdif/Makefile.am:
	  * ext/amrwb/Makefile.am:
	  * ext/apexsink/Makefile.am:
	  * ext/arts/Makefile.am:
	  * ext/artsd/Makefile.am:
	  * ext/audiofile/Makefile.am:
	  * ext/audioresample/Makefile.am:
	  * ext/bz2/Makefile.am:
	  * ext/cdaudio/Makefile.am:
	  * ext/celt/Makefile.am:
	  * ext/dc1394/Makefile.am:
	  * ext/dirac/Makefile.am:
	  * ext/directfb/Makefile.am:
	  * ext/divx/Makefile.am:
	  * ext/dts/Makefile.am:
	  * ext/faac/Makefile.am:
	  * ext/faad/Makefile.am:
	  * ext/gsm/Makefile.am:
	  * ext/hermes/Makefile.am:
	  * ext/ivorbis/Makefile.am:
	  * ext/jack/Makefile.am:
	  * ext/jp2k/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/lcs/Makefile.am:
	  * ext/libfame/Makefile.am:
	  * ext/libmms/Makefile.am:
	  * ext/metadata/Makefile.am:
	  * ext/mpeg2enc/Makefile.am:
	  * ext/mplex/Makefile.am:
	  * ext/musepack/Makefile.am:
	  * ext/musicbrainz/Makefile.am:
	  * ext/mythtv/Makefile.am:
	  * ext/nas/Makefile.am:
	  * ext/neon/Makefile.am:
	  * ext/ofa/Makefile.am:
	  * ext/polyp/Makefile.am:
	  * ext/resindvd/Makefile.am:
	  * ext/sdl/Makefile.am:
	  * ext/shout/Makefile.am:
	  * ext/snapshot/Makefile.am:
	  * ext/sndfile/Makefile.am:
	  * ext/soundtouch/Makefile.am:
	  * ext/spc/Makefile.am:
	  * ext/swfdec/Makefile.am:
	  * ext/tarkin/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/timidity/Makefile.am:
	  * ext/twolame/Makefile.am:
	  * ext/x264/Makefile.am:
	  * ext/xine/Makefile.am:
	  * ext/xvid/Makefile.am:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/dshow/Makefile.am:
	  * gst/aiffparse/Makefile.am:
	  * gst/app/Makefile.am:
	  * gst/audiobuffer/Makefile.am:
	  * gst/bayer/Makefile.am:
	  * gst/cdxaparse/Makefile.am:
	  * gst/chart/Makefile.am:
	  * gst/colorspace/Makefile.am:
	  * gst/dccp/Makefile.am:
	  * gst/deinterlace/Makefile.am:
	  * gst/deinterlace2/Makefile.am:
	  * gst/dvdspu/Makefile.am:
	  * gst/festival/Makefile.am:
	  * gst/filter/Makefile.am:
	  * gst/flacparse/Makefile.am:
	  * gst/flv/Makefile.am:
	  * gst/games/Makefile.am:
	  * gst/h264parse/Makefile.am:
	  * gst/librfb/Makefile.am:
	  * gst/mixmatrix/Makefile.am:
	  * gst/modplug/Makefile.am:
	  * gst/mpeg1sys/Makefile.am:
	  * gst/mpeg4videoparse/Makefile.am:
	  * gst/mpegdemux/Makefile.am:
	  * gst/mpegtsmux/Makefile.am:
	  * gst/mpegvideoparse/Makefile.am:
	  * gst/mve/Makefile.am:
	  * gst/nsf/Makefile.am:
	  * gst/nuvdemux/Makefile.am:
	  * gst/overlay/Makefile.am:
	  * gst/passthrough/Makefile.am:
	  * gst/pcapparse/Makefile.am:
	  * gst/playondemand/Makefile.am:
	  * gst/rawparse/Makefile.am:
	  * gst/real/Makefile.am:
	  * gst/rtjpeg/Makefile.am:
	  * gst/rtpmanager/Makefile.am:
	  * gst/scaletempo/Makefile.am:
	  * gst/sdp/Makefile.am:
	  * gst/selector/Makefile.am:
	  * gst/smooth/Makefile.am:
	  * gst/smoothwave/Makefile.am:
	  * gst/speed/Makefile.am:
	  * gst/speexresample/Makefile.am:
	  * gst/stereo/Makefile.am:
	  * gst/subenc/Makefile.am:
	  * gst/tta/Makefile.am:
	  * gst/vbidec/Makefile.am:
	  * gst/videodrop/Makefile.am:
	  * gst/videosignal/Makefile.am:
	  * gst/virtualdub/Makefile.am:
	  * gst/vmnc/Makefile.am:
	  * gst/y4m/Makefile.am:
	  * sys/acmenc/Makefile.am:
	  * sys/cdrom/Makefile.am:
	  * sys/dshowdecwrapper/Makefile.am:
	  * sys/dshowsrcwrapper/Makefile.am:
	  * sys/dvb/Makefile.am:
	  * sys/dxr3/Makefile.am:
	  * sys/fbdev/Makefile.am:
	  * sys/oss4/Makefile.am:
	  * sys/qcam/Makefile.am:
	  * sys/qtwrapper/Makefile.am:
	  * sys/vcd/Makefile.am:
	  * sys/wininet/Makefile.am:
	  * win32/common/config.h:
	  Don't install static libs for plugins. Fixes #550851 for -bad.

2008-09-17 13:59:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Fix compiler warnings on OS/X
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosink.c: (jack_process_cb):
	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
	  Fix compiler warnings on OS/X

2008-08-07 13:15:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed.
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosrc.c:
	  Try committing this once again. Now properly renamed.

2008-08-07 09:09:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: docs/plugins/inspect/plugin-jack.xml
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-jack.xml
	  Add new element to docs.
	  * ext/jack/gstjack.h
	  Add missing file.
	  * ext/jack/gstjackaudiosrc.c:
	  * ext/jack/gstjackaudiosrc.h:
	  Rename jackaudiosrc to jack_audio_src.

2008-08-07 08:47:40 +0000  Tristan Matthews <tristan@sat.qc.ca>

	  ext/jack/: Add a jackaudiosrc. Refactor sink slightly for better code reuse.
	  Original commit message from CVS:
	  patch by: Tristan Matthews <tristan@sat.qc.ca>
	  * ext/jack/Makefile.am:
	  * ext/jack/gstjack.c:
	  * ext/jack/gstjackaudioclient.c:
	  * ext/jack/gstjackaudiosink.c:
	  * ext/jack/gstjackaudiosink.h:
	  * ext/jack/gstjackaudiosrc.c:
	  * ext/jack/gstjackaudiosrc.h:
	  * ext/jack/gstjackringbuffer.h:
	  Add a jackaudiosrc. Refactor sink slightly for better code reuse.
	  Fixes #545197.

2008-06-13 11:59:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.prerequisites:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-alsaspdif.xml:
	  * docs/plugins/inspect/plugin-amrwb.xml:
	  * docs/plugins/inspect/plugin-app.xml:
	  * docs/plugins/inspect/plugin-bayer.xml:
	  * docs/plugins/inspect/plugin-bz2.xml:
	  * docs/plugins/inspect/plugin-cdaudio.xml:
	  * docs/plugins/inspect/plugin-cdxaparse.xml:
	  * docs/plugins/inspect/plugin-dtsdec.xml:
	  * docs/plugins/inspect/plugin-dvb.xml:
	  * docs/plugins/inspect/plugin-dvdspu.xml:
	  * docs/plugins/inspect/plugin-faac.xml:
	  * docs/plugins/inspect/plugin-faad.xml:
	  * docs/plugins/inspect/plugin-fbdevsink.xml:
	  * docs/plugins/inspect/plugin-festival.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-flvdemux.xml:
	  * docs/plugins/inspect/plugin-freeze.xml:
	  * docs/plugins/inspect/plugin-gsm.xml:
	  * docs/plugins/inspect/plugin-gstinterlace.xml:
	  * docs/plugins/inspect/plugin-gstrtpmanager.xml:
	  * docs/plugins/inspect/plugin-h264parse.xml:
	  * docs/plugins/inspect/plugin-interleave.xml:
	  * docs/plugins/inspect/plugin-jack.xml:
	  * docs/plugins/inspect/plugin-ladspa.xml:
	  * docs/plugins/inspect/plugin-metadata.xml:
	  * docs/plugins/inspect/plugin-mms.xml:
	  * docs/plugins/inspect/plugin-modplug.xml:
	  * docs/plugins/inspect/plugin-mpeg2enc.xml:
	  * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
	  * docs/plugins/inspect/plugin-mpegtsparse.xml:
	  * docs/plugins/inspect/plugin-mpegvideoparse.xml:
	  * docs/plugins/inspect/plugin-musepack.xml:
	  * docs/plugins/inspect/plugin-musicbrainz.xml:
	  * docs/plugins/inspect/plugin-mve.xml:
	  * docs/plugins/inspect/plugin-mythtv.xml
	  * docs/plugins/inspect/plugin-nas.xml:
	  * docs/plugins/inspect/plugin-neon.xml:
	  * docs/plugins/inspect/plugin-nsfdec.xml:
	  * docs/plugins/inspect/plugin-nuvdemux.xml:
	  * docs/plugins/inspect/plugin-oss4.xml
	  * docs/plugins/inspect/plugin-rawparse.xml:
	  * docs/plugins/inspect/plugin-real.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-rfbsrc.xml:
	  * docs/plugins/inspect/plugin-sdl.xml:
	  * docs/plugins/inspect/plugin-sdp.xml:
	  * docs/plugins/inspect/plugin-selector.xml:
	  * docs/plugins/inspect/plugin-sndfile.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-spcdec.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-speexresample.xml:
	  * docs/plugins/inspect/plugin-stereo.xml:
	  * docs/plugins/inspect/plugin-subenc.xml
	  * docs/plugins/inspect/plugin-timidity.xml:
	  * docs/plugins/inspect/plugin-tta.xml:
	  * docs/plugins/inspect/plugin-vcdsrc.xml:
	  * docs/plugins/inspect/plugin-videosignal.xml:
	  * docs/plugins/inspect/plugin-vmnc.xml:
	  * docs/plugins/inspect/plugin-wildmidi.xml:
	  * docs/plugins/inspect/plugin-x264.xml:
	  * docs/plugins/inspect/plugin-xvid.xml:
	  * docs/plugins/inspect/plugin-y4menc.xml:
	  * ext/amrwb/gstamrwbdec.c:
	  * ext/amrwb/gstamrwbenc.c:
	  * ext/amrwb/gstamrwbparse.c:
	  * ext/dc1394/gstdc1394.c:
	  * ext/directfb/dfbvideosink.c:
	  * ext/ivorbis/vorbisdec.c:
	  * ext/jack/gstjackaudiosink.c:
	  * ext/mpeg2enc/gstmpeg2enc.cc:
	  * ext/mplex/gstmplex.cc:
	  * ext/musicbrainz/gsttrm.c:
	  * ext/mythtv/gstmythtvsrc.c:
	  * ext/theora/theoradec.c:
	  * ext/timidity/gsttimidity.c:
	  * ext/timidity/gstwildmidi.c:
	  * gst-libs/gst/app/gstappsink.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/dvdspu/gstdvdspu.c:
	  * gst/festival/gstfestival.c:
	  * gst/freeze/gstfreeze.c:
	  * gst/interleave/deinterleave.c:
	  * gst/interleave/interleave.c:
	  * gst/modplug/gstmodplug.cc:
	  * gst/nuvdemux/gstnuvdemux.c:
	  Add missing elements to docs. Fix doc-markup: use convinience syntax
	  for examples (produces valid docbook), add several refsec2 when we
	  have several titles. Fix some types.

2008-06-12 14:49:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Do not use short_description in section docs for elements. We extract them from element details and there will be war...
	  Original commit message from CVS:
	  * ext/dc1394/gstdc1394.c:
	  * ext/ivorbis/vorbisdec.c:
	  * ext/jack/gstjackaudiosink.c:
	  * ext/metadata/gstmetadatademux.c:
	  * ext/mythtv/gstmythtvsrc.c:
	  * ext/theora/theoradec.c:
	  * gst-libs/gst/app/gstappsink.c:
	  * gst/bayer/gstbayer2rgb.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/rawparse/gstaudioparse.c:
	  * gst/rawparse/gstvideoparse.c:
	  * gst/rtpmanager/gstrtpbin.c:
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/rtpmanager/gstrtpptdemux.c:
	  * gst/rtpmanager/gstrtpsession.c:
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  * gst/selector/gstinputselector.c:
	  * gst/selector/gstoutputselector.c:
	  * gst/videosignal/gstvideoanalyse.c:
	  * gst/videosignal/gstvideodetect.c:
	  * gst/videosignal/gstvideomark.c:
	  * sys/oss4/oss4-mixer.c:
	  * sys/oss4/oss4-sink.c:
	  * sys/oss4/oss4-source.c:
	  Do not use short_description in section docs for elements. We extract
	  them from element details and there will be warnings if they differ.
	  Also fixing up the ChangeLog order.

2008-05-26 17:52:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/jack/gstjackaudiosink.c: Include the element name in the port name to avoid duplicate port names.
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosink.c:
	  (gst_jack_audio_sink_allocate_channels):
	  Include the element name in the port name to avoid duplicate port names.

2008-04-06 20:18:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jack/gstjackaudiosink.c: Work around missing bits of thread-safety on older GLibs some more to avoid assertions w...
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init):
	  Work around missing bits of thread-safety on older GLibs some
	  more to avoid assertions when starting up multiple playbin
	  objects concurrently (see #512382).

2008-03-13 14:25:20 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values where possible. Fixes bug #522212.
	  Original commit message from CVS:
	  * ext/alsaspdif/alsaspdifsink.c:
	  * ext/gsm/gstgsm.c:
	  * ext/jack/gstjack.c:
	  * ext/libmms/gstmms.c:
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/shout/gstshout.c:
	  * ext/timidity/gsttimidity.c:
	  * ext/timidity/gstwildmidi.c:
	  * gst/nuvdemux/gstnuvdemux.c:
	  * gst/tta/gsttta.c:
	  Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead
	  of hardcoding values where possible. Fixes bug #522212.

2007-07-18 07:42:47 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/jack/gstjackaudiosink.c: Add stdlib include here too.
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
	  (gst_jack_ring_buffer_acquire):
	  Add stdlib include here too.

2007-04-04 07:36:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/jack/gstjackaudiosink.c: Try t better name clients. properly handle return codes when re- establishing links.
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
	  (gst_jack_ring_buffer_acquire):
	  Try t better name clients. properly handle return codes when re-
	  establishing links.

2007-03-18 17:57:48 +0000  Paul Davis <paul@linuxaudiosystems.com>

	  ext/jack/gstjackaudioclient.c: Don't need to take the connection lock, it will not be used and could cause deadlocks.
	  Original commit message from CVS:
	  Based on patch by: Paul Davis <paul at linuxaudiosystems dot com>
	  * ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection):
	  Don't need to take the connection lock, it will not be used and could
	  cause deadlocks.

2007-03-08 15:24:52 +0000  Paul Davis <paul@linuxaudiosystems.com>

	  ext/jack/: Make an object to manage client connections to the jack server which we will use in the future to run sele...
	  Original commit message from CVS:
	  Includes patch by: Paul Davis <paul at linuxaudiosystems dot com>
	  * ext/jack/Makefile.am:
	  * ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init),
	  (jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb),
	  (jack_shutdown_cb), (connection_find),
	  (gst_jack_audio_make_connection), (gst_jack_audio_get_connection),
	  (gst_jack_audio_unref_connection),
	  (gst_jack_audio_connection_add_client),
	  (gst_jack_audio_connection_remove_client),
	  (gst_jack_audio_client_new), (gst_jack_audio_client_free),
	  (gst_jack_audio_client_get_client),
	  (gst_jack_audio_client_set_active):
	  * ext/jack/gstjackaudioclient.h:
	  Make an object to manage client connections to the jack server which we
	  will use in the future to run selected jack elements with the same jack
	  connection.
	  Make some stuff a bit more threadsafe.
	  Activate the jack client ASAP.
	  * ext/jack/gstjackaudiosink.c:
	  (gst_jack_audio_sink_allocate_channels),
	  (gst_jack_audio_sink_free_channels), (jack_process_cb),
	  (gst_jack_ring_buffer_open_device),
	  (gst_jack_ring_buffer_close_device),
	  (gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
	  (gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
	  (gst_jack_audio_sink_getcaps):
	  * ext/jack/gstjackaudiosink.h:
	  Use new client object to manage connections.
	  Don't remove and recreate all ports, try to reuse them.

2007-01-12 10:25:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/jack/gstjackaudiosink.*: Improve docs.
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosink.c: (jack_sample_rate_cb),
	  (jack_buffer_size_cb), (jack_shutdown_cb),
	  (gst_jack_ring_buffer_acquire):
	  * ext/jack/gstjackaudiosink.h:
	  Improve docs.

2006-12-06 16:57:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/jack/.cvsignore: Ignore old files as requested by the build slave.
	  Original commit message from CVS:
	  * ext/jack/.cvsignore:
	  Ignore old files as requested by the build slave.

2006-11-30 11:59:04 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/Makefile.am: Fix build.
	  Original commit message from CVS:
	  * ext/Makefile.am:
	  Fix build.
	  * ext/jack/gstjackaudiosink.c: (jack_process_cb),
	  (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
	  (gst_jack_ring_buffer_acquire):
	  Small cleanups.

2006-11-30 11:49:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Added fully functional jackaudiosink.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/Makefile.am:
	  * ext/jack/Makefile.am:
	  * ext/jack/gstjack.c: (plugin_init):
	  * ext/jack/gstjack.h:
	  * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type),
	  (gst_jack_ring_buffer_class_init), (jack_process_cb),
	  (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
	  (gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose),
	  (gst_jack_ring_buffer_finalize),
	  (gst_jack_ring_buffer_open_device),
	  (gst_jack_ring_buffer_close_device),
	  (gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
	  (gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause),
	  (gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay),
	  (gst_jack_connect_get_type), (gst_jack_audio_sink_base_init),
	  (gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
	  (gst_jack_audio_sink_set_property),
	  (gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps),
	  (gst_jack_audio_sink_create_ringbuffer):
	  * ext/jack/gstjackaudiosink.h:
	  Added fully functional jackaudiosink.

2006-04-08 21:48:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
	  Original commit message from CVS:
	  * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init):
	  * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init):
	  * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init):
	  * ext/arts/gst_arts.c: (gst_arts_class_init):
	  * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init):
	  * ext/audiofile/gstafsink.c: (gst_afsink_class_init):
	  * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init):
	  * ext/audioresample/gstaudioresample.c:
	  * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init):
	  * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init):
	  * ext/divx/gstdivxdec.c: (gst_divxdec_class_init):
	  * ext/hermes/gsthermescolorspace.c:
	  (gst_hermes_colorspace_class_init):
	  * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init):
	  * ext/jack/gstjack.c: (gst_jack_class_init):
	  * ext/jack/gstjackbin.c: (gst_jack_bin_class_init):
	  * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init):
	  * ext/libfame/gstlibfame.c: (gst_fameenc_class_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init):
	  * ext/nas/nassink.c: (gst_nassink_class_init):
	  * ext/shout/gstshout.c: (gst_icecastsend_class_init):
	  * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
	  * ext/sndfile/gstsf.c: (gst_sf_class_init):
	  * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init),
	  (gst_swfdec_class_init):
	  * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init):
	  * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init):
	  * gst/chart/gstchart.c: (gst_chart_class_init):
	  * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init):
	  * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init):
	  * gst/festival/gstfestival.c: (gst_festival_class_init):
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
	  * gst/filter/gstiir.c: (gst_iir_class_init):
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
	  * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init):
	  * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init):
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  (gst_system_encode_class_init):
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  (gst_mp1videoparse_class_init):
	  * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init):
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  (gst_mp3parse_class_init):
	  * gst/overlay/gstoverlay.c: (gst_overlay_class_init):
	  * gst/passthrough/gstpassthrough.c: (passthrough_class_init):
	  * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init):
	  * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init):
	  * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init):
	  * gst/smooth/gstsmooth.c: (gst_smooth_class_init):
	  * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init):
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
	  * gst/stereo/gststereo.c: (gst_stereo_class_init):
	  * gst/switch/gstswitch.c: (gst_switch_class_init):
	  * gst/tta/gstttadec.c: (gst_tta_dec_class_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_class_init):
	  * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init):
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init):
	  * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init):
	  * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init):
	  * sys/cdrom/gstcdplayer.c: (cdplayer_class_init):
	  * sys/directsound/gstdirectsoundsink.c:
	  (gst_directsoundsink_class_init):
	  * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init):
	  * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init):
	  * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init):
	  * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init):
	  * sys/v4l2/gstv4l2colorbalance.c:
	  (gst_v4l2_color_balance_channel_class_init):
	  * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init),
	  (gst_v4l2_tuner_norm_class_init):
	  * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init):
	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)

2006-04-01 10:09:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/jack/gstjack.c:
	  rework build; add translations for v4l2
	  Original commit message from CVS:
	  rework build; add translations for v4l2

2005-10-12 14:29:55 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition
	  Original commit message from CVS:
	  * examples/indexing/indexmpeg.c: (main):
	  * ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio),
	  (gst_artsdsink_close_audio), (gst_artsdsink_change_state):
	  * ext/artsd/gstartsdsink.h:
	  * ext/audiofile/gstafparse.c: (gst_afparse_open_file),
	  (gst_afparse_close_file):
	  * ext/audiofile/gstafparse.h:
	  * ext/audiofile/gstafsink.c: (gst_afsink_open_file),
	  (gst_afsink_close_file), (gst_afsink_chain),
	  (gst_afsink_change_state):
	  * ext/audiofile/gstafsink.h:
	  * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
	  (gst_afsrc_close_file), (gst_afsrc_change_state):
	  * ext/audiofile/gstafsrc.h:
	  * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init):
	  * ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init):
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_init):
	  * ext/jack/gstjack.h:
	  * ext/jack/gstjackbin.c: (gst_jack_bin_init),
	  (gst_jack_bin_change_state):
	  * ext/musepack/gstmusepackdec.c: (gst_musepackdec_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init):
	  * ext/nas/nassink.c: (gst_nassink_open_audio),
	  (gst_nassink_close_audio), (gst_nassink_change_state):
	  * ext/nas/nassink.h:
	  * ext/polyp/polypsink.c: (gst_polypsink_init):
	  * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state):
	  * ext/sdl/sdlvideosink.h:
	  * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init):
	  * ext/sndfile/gstsf.c: (gst_sf_set_property),
	  (gst_sf_change_state), (gst_sf_release_request_pad),
	  (gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop):
	  * ext/sndfile/gstsf.h:
	  * ext/swfdec/gstswfdec.c: (gst_swfdec_init):
	  * ext/tarkin/gsttarkindec.c: (gst_tarkindec_init):
	  * gst/apetag/apedemux.c: (gst_ape_demux_init):
	  * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init):
	  * gst/festival/gstfestival.c: (gst_festival_change_state):
	  * gst/festival/gstfestival.h:
	  * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
	  * gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init),
	  (gst_multifilesink_set_location), (gst_multifilesink_open_file),
	  (gst_multifilesink_close_file), (gst_multifilesink_next_file),
	  (gst_multifilesink_pad_query), (gst_multifilesink_handle_event),
	  (gst_multifilesink_chain), (gst_multifilesink_change_state):
	  * gst/multifilesink/gstmultifilesink.h:
	  * gst/videodrop/gstvideodrop.c: (gst_videodrop_init):
	  * sys/cdrom/gstcdplayer.c: (cdplayer_init):
	  * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init),
	  (dxr3audiosink_open), (dxr3audiosink_close),
	  (dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3),
	  (dxr3audiosink_change_state):
	  * sys/dxr3/dxr3audiosink.h:
	  * sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open),
	  (dxr3spusink_close), (dxr3spusink_chain),
	  (dxr3spusink_change_state):
	  * sys/dxr3/dxr3spusink.h:
	  * sys/dxr3/dxr3videosink.c: (dxr3videosink_init),
	  (dxr3videosink_open), (dxr3videosink_close),
	  (dxr3videosink_write_data), (dxr3videosink_change_state):
	  * sys/dxr3/dxr3videosink.h:
	  * sys/glsink/glimagesink.c: (gst_glimagesink_init):
	  * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state),
	  (gst_qcamsrc_open), (gst_qcamsrc_close):
	  * sys/qcam/gstqcamsrc.h:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  * sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get),
	  (gst_vcdsrc_open_file), (gst_vcdsrc_close_file),
	  (gst_vcdsrc_change_state), (gst_vcdsrc_recalculate):
	  * sys/vcd/vcdsrc.h:
	  renamed GST_FLAGS macros to GST_OBJECT_FLAGS
	  moved bitshift from macro to enum definition

2005-09-05 17:20:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackbin.c:
	  Fix up all the state change functions.
	  Original commit message from CVS:
	  Fix up all the state change functions.

2004-08-03 14:28:12 +0000  Benjamin Otte <otte@gnome.org>

	  fixes for G_DISABLE_ASSERT and friends
	  Original commit message from CVS:
	  * examples/dynparams/filter.c: (ui_control_create):
	  * examples/gstplay/player.c: (print_tag):
	  * ext/alsa/gstalsa.c: (gst_alsa_request_new_pad):
	  * ext/gdk_pixbuf/gstgdkanimation.c:
	  (gst_gdk_animation_iter_may_advance):
	  * ext/jack/gstjack.c: (gst_jack_request_new_pad):
	  * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list),
	  (tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event):
	  * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value):
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value):
	  * ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain):
	  * gst-libs/gst/media-info/media-info-test.c: (print_tag):
	  * gst/sine/demo-dparams.c: (main):
	  * gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
	  * testsuite/alsa/formats.c: (create_pipeline):
	  * testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get):
	  fixes for G_DISABLE_ASSERT and friends
	  * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
	  (mp3_type_frame_length_from_header), (mp3_type_find),
	  (plugin_init):
	  require mp3 typefinding to have at least MIN_HEADERS valid headers
	  add typefinding for AAC adts files

2004-05-21 23:28:57 +0000  Stéphane Loeuillet <gstreamer@leroutier.net>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	  second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in ...
	  Original commit message from CVS:
	  second batch :
	  remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
	  (in gst-plugins/ext/ this time)

2004-03-15 19:32:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackbin.c:
	  don't mix tabs and spaces
	  Original commit message from CVS:
	  don't mix tabs and spaces

2004-03-15 16:32:54 +0000  Johan Dahlin <johan@gnome.org>

	  *.h: Revert indenting
	  Original commit message from CVS:
	  * *.h: Revert indenting

2004-03-14 22:34:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackbin.c:
	  gst-indent
	  Original commit message from CVS:
	  gst-indent

2004-01-12 03:40:18 +0000  David Schleef <ds@schleef.org>

	* ext/jack/gstjack.c:
	  Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
	  Original commit message from CVS:
	  Remove all usage of gst_pad_get_caps(), and replace it with
	  gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().

2003-12-22 01:47:09 +0000  David Schleef <ds@schleef.org>

	* ext/jack/gstjack.c:
	  Merge CAPS branch
	  Original commit message from CVS:
	  Merge CAPS branch

2003-12-13 16:59:51 +0000  Benjamin Otte <otte@gnome.org>

	* ext/jack/gstjackbin.c:
	  removed GST_*_CAST. Disabling of type checking is done in glib.
	  Original commit message from CVS:
	  removed GST_*_CAST. Disabling of type checking is done in glib.

2003-12-04 10:37:38 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  remove copyright field from plugins
	  Original commit message from CVS:
	  remove copyright field from plugins

2003-11-07 12:47:02 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/jack/gstjackbin.c:
	  Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro...
	  Original commit message from CVS:
	  Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files

2003-11-01 23:43:13 +0000  Iain Holmes <iain@prettypeople.org>

	* ext/jack/gstjack.c:
	  Jack fixed too
	  Original commit message from CVS:
	  Jack fixed too

2003-10-29 03:15:55 +0000  David Schleef <ds@schleef.org>

	* ext/jack/gstjack.h:
	  change gst/bytestream.h to gst/bytestream/bytestream.h
	  Original commit message from CVS:
	  change gst/bytestream.h to gst/bytestream/bytestream.h

2003-10-28 20:52:41 +0000  Benjamin Otte <otte@gnome.org>

	* ext/jack/gstjack.h:
	  merge TYPEFIND branch. Major changes:
	  Original commit message from CVS:
	  merge TYPEFIND branch. Major changes:
	  - totally reworked type(find) system
	  - all typefind functions are in gst/typefind now
	  - more typefind functions then before
	  - some plugins might fail to compile now because I don't have them installed and they
	  a) require bytestream or
	  b) haven't had their typefind fixed.
	  Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies

2003-10-08 16:08:19 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
	  Original commit message from CVS:
	  /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.

2003-10-01 13:14:50 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/jack/gstjack.h:
	  New typefind system: bytestream is now part of the core all plugins have been modified to use this new typefind syste...
	  Original commit message from CVS:
	  New typefind system:
	  * bytestream is now part of the core
	  * all plugins have been modified to use this new typefind system
	  * asf typefinding added
	  * mpeg video stream typefiding removed because it's broken
	  * duplicate typefind entries removed
	  * extra id3 typefinding added, because we've seen 4 types of files
	  (riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs
	  to work. Instead, I've added an id3 element and let it redo typefiding
	  after the id3 header. this needs a hack because spider only typefinds
	  once. We can remove this hack once spider supports multiple typefinds.
	  * with all this, mp3 typefinding is semi-rewritten
	  * id3 typefinding in flac/vorbis is removed, it's no longer needed
	  * fixed spider and gst-typefind to use this, too.
	  * Other general cleanups

2003-09-30 12:56:27 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackbin.c:
	  conform to the buffer-frames props entry -- much nicer now...
	  Original commit message from CVS:
	  conform to the buffer-frames props entry -- much nicer now...

2003-08-10 00:01:58 +0000  David Schleef <ds@schleef.org>

	* ext/jack/Makefile.am:
	  Remove redundant plugindir definition
	  Original commit message from CVS:
	  Remove redundant plugindir definition

2003-07-19 23:25:25 +0000  Leif Johnson <leif@ambient.2y.net>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	  + changes for new float caps without slope/intercept + some category changes for plugins
	  Original commit message from CVS:
	  + changes for new float caps without slope/intercept
	  + some category changes for plugins

2003-07-06 20:49:52 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/jack/gstjack.c:
	  New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri...
	  Original commit message from CVS:
	  New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs

2003-07-01 02:27:06 +0000  David Schleef <ds@schleef.org>

	* ext/jack/gstjack.c:
	  fix type punning
	  Original commit message from CVS:
	  fix type punning

2003-06-29 19:46:13 +0000  Benjamin Otte <otte@gnome.org>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackbin.c:
	  compatibility fix for new GST_DEBUG stuff.
	  Original commit message from CVS:
	  compatibility fix for new GST_DEBUG stuff.
	  Includes fixes for missing includes for config.h and unistd.h
	  I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.

2003-06-13 21:21:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/jack/gstjack.c:
	  Removed ugly caps fixed flag hack, will be done automatically in core soon
	  Original commit message from CVS:
	  Removed ugly caps fixed flag hack, will be done automatically in
	  core soon

2003-03-04 15:34:20 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackbin.c:
	  update for the latest jack cvs and non-cothreaded gst scheduler
	  Original commit message from CVS:
	  update for the latest jack cvs and non-cothreaded gst scheduler

2003-02-05 20:38:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ext/jack/gstjack.c:
	  Changed caps->fixed to use FLAG_SET
	  Original commit message from CVS:
	  Changed caps->fixed to use FLAG_SET

2003-01-10 13:38:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/jack/gstjack.c:
	  PadConnect -> PadLink
	  Original commit message from CVS:
	  PadConnect -> PadLink

2003-01-10 10:22:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/jack/gstjack.c:
	  another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t...
	  Original commit message from CVS:
	  another batch of connect->link fixes
	  please let me know about issues
	  and please refrain of making them yourself, so that I don't spend double
	  the time resolving conflicts

2002-12-08 14:50:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/jack/Makefile.am:
	  parallel install fixes
	  Original commit message from CVS:
	  parallel install fixes

2002-09-29 18:12:18 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackbin.c:
	  licenses again
	  Original commit message from CVS:
	  licenses again

2002-09-18 19:02:52 +0000  Christian Schaller <uraeus@gnome.org>

	* ext/jack/gstjack.c:
	  plugins part of license field patch
	  Original commit message from CVS:
	  plugins part of license field patch

2002-09-10 09:31:40 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/jack/gstjack.c:
	  This updates all plugins to the new API for gst_pad_try_set_caps
	  Original commit message from CVS:
	  This updates all plugins to the new API for gst_pad_try_set_caps

2002-09-09 23:27:38 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/jack/gstjack.c:
	  removing warnings as approved by wim
	  Original commit message from CVS:
	  removing warnings as approved by wim

2002-08-23 04:04:11 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackbin.c:
	  fix jack input port connection
	  Original commit message from CVS:
	  fix jack input port connection

2002-07-09 17:39:17 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  compile fixen, and prepare to move MAINTAINER_MODE to as-version.m4
	  Original commit message from CVS:
	  compile fixen, and prepare to move MAINTAINER_MODE to as-version.m4

2002-07-02 23:35:07 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackbin.c:
	  make jack work in all its full duplex glory
	  Original commit message from CVS:
	  make jack work in all its full duplex glory

2002-06-12 03:32:02 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjackbin.c:
	  working jack elements (fixed a problem in upstream jack) random other fixen...
	  Original commit message from CVS:
	  * working jack elements (fixed a problem in upstream jack)
	  * random other fixen...

2002-05-15 19:08:49 +0000  Steve Baker <steve@stevebaker.org>

	* ext/jack/gstjack.c:
	  use new bytestream api
	  Original commit message from CVS:
	  use new bytestream api

2002-05-13 18:08:33 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackbin.c:
	  update to new jack api
	  Original commit message from CVS:
	  update to new jack api

2002-05-05 19:39:17 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  add some includes
	  Original commit message from CVS:
	  add some includes

2002-05-05 01:08:05 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackbin.c:
	  better initialization. it doesn't work over here, though.
	  Original commit message from CVS:
	  better initialization. it doesn't work over here, though.

2002-05-04 21:38:56 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjackbin.c:
	  a commit so that jack will build without errors on Uraeus's system ;)
	  Original commit message from CVS:
	  a commit so that jack will build without errors on Uraeus's system ;)

2002-05-04 20:53:35 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  set caps once we know the sample rate of the system
	  Original commit message from CVS:
	  set caps once we know the sample rate of the system

2002-05-04 18:57:44 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackbin.c:
	  some jack fixes, alsa touchups, and add rtp by default to the build if there are any problems building rtp, we're mov...
	  Original commit message from CVS:
	  some jack fixes, alsa touchups, and add rtp by default to the build
	  if there are any problems building rtp, we're moving it back to experimental ;)

2002-04-20 21:42:51 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br...
	  Original commit message from CVS:
	  * a hack to work around intltool's brokenness
	  * a current check for mpeg2dec
	  * details->klass reorganizations
	  * an element browser that uses details->klass
	  * separated cdxa parse out from the avi directory

2002-04-16 17:14:05 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/Makefile.am:
	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	* ext/jack/gstjackbin.c:
	  Finally we're on to a proper jack setup, with a specialized bin and elements that can only go in a jack bin. I had to...
	  Original commit message from CVS:
	  Finally we're on to a proper jack setup, with a specialized bin and elements
	  that can only go in a jack bin. I had to fix the parser first to do this, but
	  to run it, the syntax is like so:
	  gst-launch jackbin.( filesrc ! mad ! jacksink )
	  But of course it's not fully functional yet. Sigh.

2002-04-11 20:42:26 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind.
	  Original commit message from CVS:
	  GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE
	  same with *factory and typefind.
	  also, some -Werror fixes.

2002-03-30 21:07:51 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  alphabetization fixen a jack caps fix
	  Original commit message from CVS:
	  * alphabetization fixen
	  * a jack caps fix

2002-03-30 19:31:13 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  add notify back to filesrc, it's needed for MVC applications remove notify printouts from gst-launch cleanup in gst-p...
	  Original commit message from CVS:
	  * add notify back to filesrc, it's needed for MVC applications
	  * remove notify printouts from gst-launch
	  * cleanup in gst-plugins configure.ac
	  * some jack updates
	  * remove SELF_ITERATING flag in favor of SEF_SCHEDULABLE (not a clear name,
	  but it's what we have for the moment)
	  * improve parsing of request pad names, no more sscanf
	  * fixes to the fastscheduler Makefile.am

2002-03-20 21:45:04 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/gstjack.c:
	  s/Gnome-Streamer/GStreamer/
	  Original commit message from CVS:
	  s/Gnome-Streamer/GStreamer/

2002-03-19 04:10:06 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/Makefile.am:
	* ext/jack/gstjack.c:
	  removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with...
	  Original commit message from CVS:
	  * removal of //-style comments
	  * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
	  and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.

2002-03-19 01:39:43 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/Makefile.am:
	  s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi...
	  Original commit message from CVS:
	  s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/
	  @-substitued variables variables are defined as make variables automagically,
	  and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag

2002-03-18 04:41:35 +0000  Andy Wingo <wingo@pobox.com>

	* ext/jack/Makefile.am:
	* ext/jack/README:
	* ext/jack/gstjack.c:
	* ext/jack/gstjack.h:
	  s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way added jack ...
	  Original commit message from CVS:
	  * s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way
	  * added jack element, doesn't quite work right yet but i didn't want to lose the work -- it does build, register,
	  and attempt to run though
	  * imposed some restrictions on the naming of request pads to better allow for reverse parsing
	  * added '%s' to reverse parsing
	  * added new bin flag to indicate that it is self-iterating, and some lame code in gst-launch to test it out
	  * fixen on launch-gui
	  * added pkg-config stuff for the editor's libs

2011-01-02 11:37:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: mark v4l2sink as experimental and build only if --enable-experimental is passed
	  It's not really of 'good' quality yet, but there's a lot of
	  code shared with v4l2src, so not so easy to move it elswhere.
	  https://bugzilla.gnome.org/show_bug.cgi?id=612244

2011-01-02 01:24:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/v4l2_calls.c:
	  Revert "v4l2: add norm property"
	  This reverts commit 9e1d419d07337e6db2cc3936472be205ce927e54.
	  Reverting this since it adds unreviewed and bad API to v4l2src
	  (property of type enum, with seemingly random and unsorted values).

2011-01-01 23:26:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tools/.gitignore:
	* tools/Makefile.am:
	* tools/README.filterstamp:
	* tools/filterstamp.sh:
	* tools/gst-launch-ext-m.m:
	* tools/gst-launch-ext.1.in:
	* tools/gst-visualise-m.m:
	* tools/gst-visualise.1.in:
	  tools: remove unused left-over directory
	  These are all in -base/tools.

2010-12-31 13:57:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4adepay.h:
	  mp4adepay: improve timestamps on outgoing packets
	  Improve parsing of the samplerate.
	  Parse the framelen so that we can calculate timestamps.
	  When interpollate the incomming timestamp on outgoing buffers when there are
	  multiple subframes.
	  fixes #625825

2010-12-31 00:12:53 -0800  David Schleef <ds@schleef.org>

	* gst/dtmf/tone_detect.c:
	  dtmf: Fix build failure caused by previous commit

2010-12-30 18:20:47 -0800  David Schleef <ds@schleef.org>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/tone_detect.c:
	* gst/dtmf/tone_detect.h:
	  dtmf: build fixes for MSVC
	  Use gint16 and G_PI.

2010-12-30 18:19:47 -0800  David Schleef <ds@schleef.org>

	* gst/dtmf/tone_detect.c:
	  dtmf: reindent

2010-12-31 02:16:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/cairo/gsttimeoverlay.c:
	* gst/videofilter/gstvideobalance.c:
	  cairo, videofilter: use gst/math-compat.h header for rint

2010-12-30 14:30:27 -0800  David Schleef <ds@schleef.org>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Check for HAVE_RINT instead
	  Also change M_PI to G_PI for giggles.

2010-12-30 14:21:37 -0800  David Schleef <ds@schleef.org>

	* ext/cairo/gstcairorender.c:
	  cairo: Don't use #ifdefs inside macros

2010-12-30 14:20:52 -0800  David Schleef <ds@schleef.org>

	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/effectv/gstop.c:
	* gst/equalizer/gstiirequalizer.c:
	* gst/goom/convolve_fx.c:
	* gst/goom/ifs.c:
	* gst/goom/lines.c:
	* gst/goom/tentacle3d.c:
	* tests/examples/audiofx/firfilter-example.c:
	* tests/examples/audiofx/iirfilter-example.c:
	  Change M_PI to G_PI

2010-12-30 12:07:52 -0800  David Schleef <ds@schleef.org>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: use G_OS_WIN32 for windows check

2010-12-30 16:24:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	  mp4adepay: fix timestamps on buffers

2010-12-30 16:22:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmpvpay.c:
	  mpvpay: fix flushing and discont
	  Fix flushing and disconts.
	  Clean up in state changes.

2010-12-29 23:38:18 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: increase allowed max. block size for push mode from 10M to 15M
	  It was an arbitrary limit from the start, meant as a basic sanity check,
	  so may just as well increase it a little. Would be good to provide
	  progress reporting while completing the block in any case..
	  https://bugzilla.gnome.org/show_bug.cgi?id=637060

2010-12-29 23:09:04 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: assume matroska if no doctype is specified
	  https://bugzilla.gnome.org/show_bug.cgi?id=638019

2010-12-04 13:43:11 -0600  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  v4l2: add interlaced support

2010-10-02 14:45:14 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/gstv4l2xoverlay.h:
	  v4l2sink: add navigation support

2010-04-04 06:43:41 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: add norm property
	  Based on a patch by Guennadi Liakhovetski.

2010-07-13 10:03:51 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	  v4l2: cleanup get/set input/output
	  output devices should use get/set output, and in either case we should
	  not print a warning message if the ioctl fails but the device does not
	  claim to support the tuner interface

2010-06-10 11:15:46 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/gstv4l2xoverlay.h:
	  v4l2xoverlay: add support to create window
	  If xoverlay is available, v4l2sink should create a window for the overlay to
	  display in.
	  The window automatically tries to make itself as large as possible.
	  This works well on a small screen, but perhaps should first attempt to use
	  the size of the video that is played (no scaling).

2010-04-04 06:41:28 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: special handling for cases gst_buffer_make_metadata_writable()
	  Special case check for sub-buffers:  In certain cases, places like
	  GstBaseTransform, which might check that the buffer is writable before copying
	  metadata, timestamp, and such, will find that the buffer has more than one
	  reference to it.  In these cases, they will create a sub-buffer with an offset=0
	  and length equal to the original buffer size.
	  This could happen in two scenarios: (1) a tee in the pipeline, and (2) because
	  the refcnt is incremented in gst_mini_object_free() before the finalize function
	  is called, and decremented after it returns..  but returning this buffer to the
	  buffer pool in the finalize function, could wake up a thread blocked in
	  _buffer_alloc() which could run and get a buffer w/ refcnt==2 before the thread
	  originally unref'ing the buffer returns from finalize function and decrements
	  the refcnt back to 1!
	  This is related to issue #545501

2010-04-04 06:39:52 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: fix race condition
	  The size of the buffer would be zero'd out in gst_v4l2_buffer_finalize()
	  after the buffer is qbuf'd or pushed onto the queue of available buffers..
	  leaving a race condition where the thread waiting for the buffer could awake
	  and set back a valid size before the finalizing thread zeros out the length.
	  This would result that the newly allocated buffer has length of zero.

2010-04-04 06:39:08 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	  v4l2sink: add properties to control crop

2010-04-04 06:37:16 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2xoverlay.c:
	  v4l2: re-enable x-overlay support

2010-12-25 11:52:36 -0600  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: fix for PAUSED->READY->PAUSED state transitions
	  When v4l2sink goes to PAUSED->READY it only stops streaming, so the state
	  should be set to STATE_PENDING_STREAMON in case the element transitions
	  back to PLAYING.

2010-04-04 06:28:51 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	  v4l2sink: add "min-queued-bufs" property

2010-04-04 06:26:50 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2sink: Add support for blocking dequeue.
	  We'd prefer to throttle the decoder if we run out of buffers, to keep a bound
	  on memory usage.  Also, for OMAP4 it is a requirement of the decoder to not
	  alternate between memory alloced by the display driver and malloc'd userspace
	  memory.

2010-04-04 06:24:41 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: clear flags before reusing buffer from buffer pool
	  note: this really only affects v4l2sink since gst_v4l2_buffer_pool_get() is
	  only called once per buffer in the v4l2src case (in
	  gst_v4l2src_buffer_pool_activate())

2010-04-04 06:23:31 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: don't render preroll buffers
	  Most v4l2 drivers will get upset when you queue the same buffer twice in a
	  row without first dequeueing it.
	  Rendering of pre-roll buffers can be re-introduced later, but will require
	  tracking the state of the buffer, and avoiding to re-QBUF if the buffer has
	  already been passed to the driver.

2010-04-04 06:22:43 -0500  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Improve behavior for shared buffers.
	  When the decoder is using pad_alloc(), v4l2sink would behave badly if
	  the number of buffers ('queue-size' property) was not high enough to
	  account for all the buffers needed by the decoder, and other elements
	  (such as queues) between the decoder and v4l2sink.  This patch
	  slightly increases the default number of buffers, and changes v4l2sink
	  to drop frames rather than return an error in case the number of
	  buffers is not high enough.

2010-11-15 15:58:28 +0100  Andy Wingo <wingo@oblong.com>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  add "client" property
	  * ext/pulse/pulsesrc.c (gst_pulsesrc_class_init, gst_pulsesrc_init)
	  (gst_pulsesrc_set_property, gst_pulsesrc_get_property)
	  (gst_pulsesrc_open): Add a "client" property, as in pulsesink.
	  Fixes #634914

2010-12-29 15:54:46 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: serialise/deserialise floats without changing locale
	  Use g_ascii_dtostr() and g_ascii_strtod() to serialise/deserialise
	  floating point numbers, instead of ugly hacks that switch locale
	  before and after calling libc functions (which is not a good idea
	  in a multi-threaded application).

2010-12-29 14:40:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	  rtpjpegdepay: fix framerate parsing for locales that use a comma as floating point
	  atof() converts strings according to the current locale, but the
	  framerate string will likely always use a dot as floating point
	  separator, so use g_ascii_strtod() instead (but also canonicalise
	  the string before, so we can handle both formats as input).

2010-12-27 13:11:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: use the right variable
	  Use the right variable for specifying that we sent a receiver report.

2010-12-23 16:42:29 -0600  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: fix typo

2010-12-23 16:03:00 -0600  Rob Clark <rob@ti.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: add stream-format and alignment properties for h264

2010-12-22 11:41:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  gstpay: fix klass, add RTP as a use case

2010-12-12 15:10:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  gstdepay: cleanup the cache

2010-12-12 05:10:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstdepay.h:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  gstpay/depay: add generic gstreamer payloader
	  Add the beginnings of a generic GStreamer buffers payloader.

2010-12-23 17:06:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4gpay.c:
	  mp4gpay: reset state on flush-stop

2010-12-23 16:26:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  mp4gdepay: flush state on flush-stop

2010-12-23 16:25:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: on-npt-stop is a manager signal

2010-12-23 15:24:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: improve RTP session handling
	  Store the RTP session in the stream so that we can more efficiently
	  perform actions on the stream based on RTP signals.

2010-12-23 13:55:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: include last send RB block
	  Only report RB values for non-internal sources.
	  Report not only the RB blocks we last received from but also the last RB
	  block we sent to a source.

2010-12-23 13:52:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: remember last sent RB values.

2010-12-23 13:00:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: include all stats and document
	  Include all possible stats of a source in the stats structure because we might
	  be interested in what happened in the past.
	  Document the stats property and the fields.

2010-12-23 12:59:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/client-PCMA.c:
	  examples: add example RTP stats
	  Add some more RTP examples for how to retrieve RTP stats in a receiver.

2010-12-23 12:58:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: also emit RTCP activity on SR
	  Also emit RTCP activity signals when we receive an SR packet without RB blocks,
	  such as from a sender that is not receiving anything.

2010-12-23 11:10:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  docs: add some more gstrtpbin docs

2010-12-22 21:27:11 +0100  Edward Hervey <bilboed@bilboed.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: remote is a boolean (and not uint) property

2010-12-22 19:58:21 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't use gst_pad_alloc_buffer()
	  Using this in a demuxer will cause deadlocks if there's
	  a pad with a pending pad-block downstream, no matter if
	  there is a queue between the pad or not. Queues pass
	  bufferalloc downstream from the same thread and only
	  act as a thread boundary for events and buffers.

2010-12-22 14:14:08 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix subtitle pad template, we only handle kate for now

2010-12-16 11:44:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  docs: update rtspsrc docs, rtpbin is not in -bad any more

2010-12-22 11:42:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: unlock before emitting signals

2010-12-21 22:34:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpac3pay.h:
	  rtpac3pay: add AC3 payloader

2010-12-21 22:17:19 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpac3depay.c:
	  ac3depay: fix debug category description

2010-12-21 22:16:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmpapay.c:
	  mpapay: add debug category

2010-12-20 14:49:02 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/jpegenc.c:
	  jpegenc: Adds another test case
	  Adds a test for jpegenc to check that is possible to negotiate and
	  push buffers with different resolution one after another.
	  https://bugzilla.gnome.org/show_bug.cgi?id=637686

2010-12-21 13:37:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: sink pad's getcaps shouldn't use the src pad getcaps
	  Instead of using get_allowed_caps on the srcpad, the sinkpad getcaps
	  should use the getcaps of the srcpad's peer. This way the srcpad
	  can keep using fixed_caps and sinkpad getcaps exposes all caps
	  that can be negotiated
	  https://bugzilla.gnome.org/show_bug.cgi?id=637686

2010-12-21 16:58:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: add RTP hint to the klass

2010-12-21 16:49:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: fix rank of payloaders and depayloaders
	  Set the payloaders and depayloaders to a reasonable rank.

2010-12-21 15:24:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  vrawdepay: reset depayloader state
	  Reset the depayloader state on flush-stop.

2010-12-21 15:07:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	  mp4pay: use vmethod for intercepting events

2010-12-21 13:55:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  theorapay: clear packet on flush-stop

2010-12-21 13:49:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbispay.c:
	  vorbispay: clear packet on flush-stop

2010-12-21 12:31:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  mp4gdepay: reset depayloader state

2010-12-21 12:29:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  h264pay: flush adapter on flush-stop

2010-12-20 18:49:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmpapay.c:
	  mpapay: flush last packets on EOS

2010-12-20 17:47:05 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 169462a to 46445ad

2010-12-20 16:51:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmpapay.c:
	  mpapay: reset payloader on state change

2010-12-20 16:05:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmpapay.c:
	  mpapay: reset payloader on flush
	  Reset the payloader on a flush event.
	  Handle DISCONT better.

2010-12-20 15:54:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: get better buffering level
	  When the jitterbuffer contains -1 timestamps, make sure we still calculate the
	  buffer fill level by skipping the -1 buffers.
	  Try to be more resilient to weird input timestamps.

2010-12-20 11:10:22 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: provide a clock.
	  since we are using the clock for sync, we need to also provide a clock for good
	  measure. The reason is that even if downstream elements provide a clock, we
	  don't want to have that clock selected because it might not be running yet.

2010-12-20 10:49:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: copy buffering stats
	  when we create an aggregate buffering message, copy the buffering stats form the
	  last message. At least we get correct buffering mode then.

2010-12-19 11:02:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/pipelines/wavenc.c:
	  wavenc: Fix memory leaks in the unit test

2010-12-19 10:58:16 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstradioac.c:
	* gst/effectv/gstradioac.h:
	  radioactv: Prevent use of uninitialized values
	  Fixes bug #618652.

2010-12-19 10:22:29 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/debugutils/gstcapsdebug.c:
	  capsdebug: Don't leak pad templates created from static pad templates

2010-11-29 12:36:06 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	  ximagesrc: change from XGetImage to XGetSubImage dependant on a property
	  ximagesrc: change from XGetImage to XGetSubImage dependant on a property
	  to avoid unnecessary performance hits by default.

2010-11-28 16:04:35 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: use XGetSubImage instead of XGetImage, works with remote X
	  ximagesrc: use XGetSubImage instead of XGetImage, works with remote X
	  (on my setup anyway...)

2010-11-27 17:15:32 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: fix various width/height calculations being off by one,
	  ximagesrc: fix various width/height calculations being off by one,
	  and make it so a single pixel width/height can be captured (except
	  the top left one, as 0,0,0,0 is reserved for full screen as per
	  the property comments).

2010-12-17 19:19:35 -0600  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2object.c:
	  fix compile errors on macosx
	  with i686-apple-darwin10-gcc-4.2.1:
	  gstv4l2object.c: In function 'gst_v4l2_object_get_nearest_size':
	  gstv4l2object.c:1988: warning: format '%u' expects type 'unsigned int', but argument 12 has type 'gint *'
	  gstv4l2object.c:1988: warning: format '%u' expects type 'unsigned int', but argument 13 has type 'gint *'

2010-12-17 15:38:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: determine output h264 layout using caps negotiation
	  ... thereby (partially) deprecating properties currently controlling whether
	  or not byte-stream output or NAL/AU alignment (though properties still determine
	  fallback if nothing specified in caps).
	  Fixes #606662.

2010-12-16 18:55:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kpay.c:
	  j2kpay: handle EOC correctly
	  Don't include the next 2 bytes when we are at the end of the data and there are
	  no more bytes left.

2010-12-16 15:15:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: flush remaining buffered samples on EOS
	  ... which can make a difference between all or nothing when dealing
	  with short streams and relatively large ringbuffer segment.

2010-12-16 10:04:19 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Change classification to Filter/Effect/Video/Deinterlace

2010-12-15 18:21:34 +0100  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: Initialize all fields
	  Makes sad compliers happy

2010-12-15 16:22:54 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kpay.c:
	  j2kpay: cleanup header construction
	  Use a simpler way of constructing the header that doesn't depend on
	  the endianness.

2010-12-15 13:30:50 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: depend on -base from git for new rtp base depayloader features
	  This is ok in this case, since the plan is to release core/base again
	  along with good/ugly/bad in the next cycle.

2010-12-15 14:55:58 +0200  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 20742ae to 169462a

2010-12-15 13:12:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kdepay.h:
	  j2kdepay: add support for buffer lists

2010-12-14 18:12:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: fix average RTCP packet size some more.
	  Fix stupid error in averaging macro.
	  Include udp headers in packet length estimation.

2010-12-14 17:15:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.c:
	  rtpbin: correctly calculate RTCP packet size

2010-12-14 15:27:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kpay.c:
	  j2kpay: stop scanning when we reached the end
	  Stop scanning for markers when we reached the end of the data.

2010-12-13 16:23:24 +0200  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 011bcc8 to 20742ae

2010-12-13 12:56:12 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: avoid leaking sink events
	  Avoid leaking the newsegment event when it has the wrong format.

2010-12-12 14:53:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4vpay.c:
	  mp4vpay: we can also accept xvid caps

2010-12-12 01:39:06 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Avoid infinite loop draining frames
	  When the pipeline is flushed just as we're draining history,
	  don't loop infinitely, just discard the history and abort.

2010-12-11 17:39:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: add "max-errors" property to ignore decoding errors
	  Add property to ignore decoding errors. Default is to ignore a few
	  decoding errors if the input is packetized, but error out immediately
	  if the input is not packetized.
	  Ignoring errors for packetized input most likely doesn't work
	  properly yet, so don't do that for now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=623063

2010-05-28 15:27:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: free/malloc instead of realloc, avoids memcpy

2010-12-11 17:49:03 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Check if there's actually a seek table before parsing it

2010-12-11 17:46:17 +0100  Kishore Arepalli <kishore.arepalli@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Implement CONVERT and FORMATS query
	  Fixes bug #636784.

2010-07-01 00:22:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: put unrecognised RIFF format IDs into the unknown caps
	  Extra info can't hurt. Field names aren't necessarily consistent with
	  what's used elsewhere though (e.g. avidemux), but then neither are the
	  caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=623178

2010-10-29 22:50:14 +0100  Jan Schmidt <thaytan@noraisin.net>

	* ext/pulse/pulsemixerctrl.c:
	* ext/pulse/pulsemixerctrl.h:
	  pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS
	  Add the mixer flag and send notifications when either the volume or muted
	  status changes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=618389

2010-02-08 21:41:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: mark DISCONT when resuming PLAY
	  In particular, when streaming interleaved, this arranges for setting a new
	  timestamp on outgoing buffer so downstream can appropriate reset
	  to a change in (rtp)time.

2010-12-02 16:08:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response

2010-10-25 11:51:06 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add and use auto buffering mode
	  ... which selects BUFFER for a non-live stream, and otherwise SLAVE.
	  Fixes #633088.

2010-12-06 12:16:12 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kdepay.h:
	  j2kdepay: make the depayloader more resilient
	  Use 3 adapters, one to accumulate paketization units, another on to accumulate
	  tiles and a last one to accumulate the final frame.
	  Don't just blindly flush the adapter on DISCONT but only discard the current
	  packetization unit.
	  When we dropped jpeg2000 packets between SOP markers, adjust the SOT header with
	  the new lenght.

2010-12-09 13:49:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix flow return aggregation

2010-12-08 11:35:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix handling near end-of-file corner cases
	  Also, relax some error handling to not bail out completely when something
	  feels amiss, but consider this EOF and continue with was obtained so far.

2010-12-07 17:19:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fragmented support; fix offset handling and relax error raising
	  In particular, accept unknown stream in track fragment, and only error out
	  if that raises problems later on with respect to offset tracking.
	  Fixes #620283.

2010-12-07 15:39:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/pipelines/lame.c:
	  check: don't use deprecated method

2010-12-07 13:11:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/Makefile.am:
	* gst/flv/gstflvdemux.c:
	  flvdemux: use aac codec-data to adjust samplerate if needed
	  Based on patch by Fabien Lebaillif-Delamare <fabien@arq-media.com>
	  Fixes #636621.

2010-12-07 11:43:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: don't uncork in _start
	  Don't uncork in the _start method just yet but wait until we have written some
	  samples to pulseaudio. This avoid underruns on pulseaudio and less crackling
	  noises when starting.

2010-12-07 11:47:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2010-12-07 11:43:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: don't uncork in _start
	  Don't uncork in the _start method just yet but wait until we have written some
	  samples to pulseaudio. This avoid underruns on pulseaudio and less crackling
	  noises when starting.

2010-12-07 11:42:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: use _object_ref_sink() when we can

2010-12-07 11:40:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: don't abuse the class lock
	  Use a new static lock to protect the probed device list instead of the object
	  class lock.

2010-12-06 19:59:49 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix compiler warnings on OSX.

2010-12-06 18:17:24 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: add debug to notify when skipping to jpeg header

2010-12-06 18:16:19 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: discard incomplete image
	  ... as determined when finding SOI next image before an EOI.
	  Based on patch by David Hoyt <david.hoyt@llnl.gov>
	  Fixes #635734.

2010-12-06 17:45:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: avoid infinite loop when resyncing
	  Fixes #635734 (partly).

2010-12-06 17:28:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11

2010-12-06 17:27:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* android/apetag.mk:
	* android/avi.mk:
	* android/flv.mk:
	* android/icydemux.mk:
	* android/id3demux.mk:
	* android/qtdemux.mk:
	* android/rtp.mk:
	* android/rtpmanager.mk:
	* android/rtsp.mk:
	* android/soup.mk:
	* android/udp.mk:
	* android/wavenc.mk:
	* android/wavparse.mk:
	* configure.ac:
	  more 0.10 -> 0.11 changes

2010-12-06 15:21:53 +0100  David Hoyt <dhoyt@llnl.gov>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: pass along eos if received before buffer arrives
	  Fixes #636172.

2010-10-20 11:05:49 +0200  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: try to write timestamps in all the outgoing buffers
	  Fixes #632654.

2010-12-06 12:21:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  configure: start 0.11 branch

2010-12-06 12:17:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/debugutils/progressreport.c:
	* gst/debugutils/progressreport.h:
	  progressreport: optionally determine progress using buffer metadata
	  Based on patch by Leo Singer <lsinger at caltech.edu>
	  Fixes #629418.

2010-12-05 14:39:19 +0100  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/interleave.c:
	  check: Fixup the shutting down order
	  First bring down everything to NULL before attempting to unlink
	  or unref anything.
	  Avoids the tests just hanging there for ever waiting to acquire a
	  lock that doesn't exist anymore.

2010-11-04 19:31:45 +0100  Janne Grunau <janne.grunau@collabora.co.uk>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2src: set top field first for interlaced buffers if v4l2 exports it
	  https://bugzilla.gnome.org/show_bug.cgi?id=634393

2010-11-04 18:36:09 +0100  Janne Grunau <janne.grunau@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: check field information and set interlaced caps accordingly
	  Reject the format if the field type is not supported.
	  https://bugzilla.gnome.org/show_bug.cgi?id=634391

2010-12-03 17:42:14 +0100  Benjamin Gaignard <benjamin.gaignard@stericsson.com>

	* Android.mk:
	* android/NOTICE:
	* android/apetag.mk:
	* android/avi.mk:
	* android/flv.mk:
	* android/gst/rtpmanager/gstrtpbin-marshal.c:
	* android/gst/rtpmanager/gstrtpbin-marshal.h:
	* android/gst/udp/gstudp-enumtypes.c:
	* android/gst/udp/gstudp-enumtypes.h:
	* android/gst/udp/gstudp-marshal.c:
	* android/gst/udp/gstudp-marshal.h:
	* android/icydemux.mk:
	* android/id3demux.mk:
	* android/qtdemux.mk:
	* android/rtp.mk:
	* android/rtpmanager.mk:
	* android/rtsp.mk:
	* android/soup.mk:
	* android/udp.mk:
	* android/wavenc.mk:
	* android/wavparse.mk:
	  Add build system for Android

2010-03-26 13:51:58 +0100  Guillaume Emont <gemont@igalia.com>

	* gst/debugutils/gstnavseek.c:
	  navseek: add basic support to change playback rate
	  The following keys will now be interpreted by navseek:
	  'f' means fast forward: the stream gets played at rate 2.0
	  'r' means rewind: the stream gets played at rate -2.0
	  'n' means normal: the stream gets played at rate 1.0
	  Fixes #631516.

2010-12-01 13:12:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: add support for e(a)c-3 audio

2010-11-19 12:44:35 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: avoid sending EOS event twice

2010-11-19 12:44:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: remove dead code trying to update stream duration
	  On the one hand, it insufficiently checks whether it only updates a dummy
	  segment.  On the other hand, only doing this at the time the last sampled is
	  prepared (and sent downstream) is too little too late.

2010-11-09 10:58:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fragmented support; handle ismv sample flags

2010-11-08 11:41:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fragmented support; handle ismv stbl atoms
	  ... or lack of some thereof, such as mandatory stsz.  Shuffle some code
	  in _stbl_init to detect this early enough.

2010-11-08 11:39:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fragmented support; compensate for ismv offset handling
	  ... or lack thereof, which according to specs would put media data in
	  unlikely position.

2010-11-04 14:07:56 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: fragmented support for push mode

2010-11-04 10:17:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: fragmented support; proper and incremental moof parsing
	  That is, parse each moof in one pass (considering all contained streams'
	  metadata), and do so incrementally as needed for playback rather than
	  an initial complete scan of all moof (though all moov sample metadata
	  is fully parsed at startup).

2010-11-04 10:06:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: refactor stream freeing

2010-11-04 10:05:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: delegate linear search for sample to binary search when possible
	  Also arrange for parsing a sample prior to taking a reference to it,
	  which requires less memory layout assumptions for correctness.

2010-11-01 15:52:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fragmented support; handle moov samples and proper stream duration

2010-11-01 13:40:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fragmented support; consider mvex and handle flags and offset fields

2010-10-28 16:49:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fragmented support; forego check for short streams
	  ... as some bogus files may indicate streams of 0 duration in moov,
	  while indicating the complete movie duration in mvhd (the latter should
	  be in mehd).

2010-10-28 16:46:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_types.h:
	  qtdemux: fragmented support; code cleanups and optimizations in atom parsing
	  Avoid extra allocation in _parse_trun, add more checks for parsing errors,
	  add or adjust some debug statement, fix comments, sprinkle some branch
	  prediction.

2010-09-13 23:19:44 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: parse_moof should return TRUE on success

2010-09-10 22:41:03 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix iteration bug
	  Avoid infinite loop when iterating traf

2010-09-10 21:32:26 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Refactor trun parsing
	  The allocation of the samples can be placed out of the loop.
	  Makes the code clearer.
	  Also avoid relying on traf information as it is placed on the
	  end of the file and might not be acessible on push mode.

2010-09-10 00:29:26 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Remove parsing of unused atom
	  sdtp atom is parsed but not used, so we don't have to
	  parse it.

2010-11-09 11:45:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: tweak wam support
	  ... with some comment and portability macros.

2009-09-23 18:47:42 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	* gst/qtdemux/qtdemux_types.c:
	  qtdemux: support wma & vc-1
	  https://bugzilla.gnome.org/show_bug.cgi?id=596321

2010-03-11 09:56:04 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: parse fmp4 samples information
	  The fragmented mp4 format stores the tracks and samples information in the
	  'moof' boxes, which are appended before each fragment (fragment->'moof'+'mdat').
	  The 'mfra' box stores the offset of each 'moof' box and their presentation
	  time. The location of this box can be retrieved from the 'mfro' box, which is
	  located at the end of the file.
	  The 'mfra' box is parsed to get the offset of each 'moof' box and their
	  presentation time.
	  Each 'moof' box can contain information for one or more tracks inside
	  'tfhd' boxes. For each track in a 'moof', we have a 'trun' box, which
	  contains information of each sample (offset and duration) used to build
	  the samples table.
	  Based on patch by Marc-André Lureau <mlureau@flumotion.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=596321

2010-03-11 15:34:49 +0100  Marc-André Lureau <mlureau@flumotion.com>

	* gst/qtdemux/qtatomparser.h:
	* gst/qtdemux/qtdemux_dump.c:
	* gst/qtdemux/qtdemux_dump.h:
	* gst/qtdemux/qtdemux_fourcc.h:
	* gst/qtdemux/qtdemux_types.c:
	* gst/qtdemux/qtdemux_types.h:
	  qtdemux: add fragmented mp4 fourccs
	  Adds fourcc's for tfra, tfhd, trun, sdtp, trex, mehd and
	  their dumps
	  https://bugzilla.gnome.org/show_bug.cgi?id=596321

2010-03-11 10:24:56 +0100  Marc-André Lureau <mlureau@flumotion.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: parse the track id from the track header
	  Signed-off-by: Andoni Morales Alastruey <amorales@flumotion.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=596321

2010-03-11 14:10:12 +0100  Marc-André Lureau <mlureau@flumotion.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: allow pulling atoms with unknown size
	  Signed-off-by: Andoni Morales Alastruey <amorales@flumotion.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=596321

2010-07-14 20:13:55 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/qtdemux/qtdemux_dump.c:
	  qtdemux: make qtdemux_dump_mvhd parse version 1 correctly
	  Versions 0 and 1 of mvhd have different sizes of its values
	  (32bits/64bits). This patch makes it dump them correctly.
	  Also use the right node in the parameter and not the root node.
	  https://bugzilla.gnome.org/show_bug.cgi?id=596321

2010-11-19 12:45:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskademux: minor cleanups in setting streamheader on caps

2010-11-02 17:04:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: normalize empty Cues to no Cues
	  ... to trigger indexless seeking.

2010-10-26 11:15:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: add workaround for buggy list size
	  Fixes truncated extra-data in hdrl/strl/strf due to buggy containing
	  list size not accounting for padding in contained chunks.

2010-12-02 16:11:01 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: do not hold custom PAD_LOCK when pushing downstream

2010-12-02 16:10:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: reset session manager base time when flushing
	  ... as rtpbin uses running time to handle rtpjitterbuffer's buffer mode pauses.

2010-12-01 16:51:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: include range request for all streams with non-aggregate control

2010-10-07 14:50:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix debug statement

2010-12-03 15:38:00 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Parse more variants of numerical IDIT tag

2010-05-07 17:30:30 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/libpng/gstpngenc.c:
	  pngenc: Use proper framerate range in caps

2010-12-03 15:04:26 +0100  Edward Hervey <bilboed@bilboed.com>

	* tests/check/pipelines/wavenc.c:
	  tests: Fix previously unbuildable/untested wavenc test

2010-10-24 15:21:08 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Refactor tag pushing logic
	  The logic of when to push was wrong also (resulting in some tags never
	  being pushed).

2010-10-24 15:20:27 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/Makefile.am:
	* gst/flv/gstflvdemux.c:
	  flvdemux: Use pbutils for codec descriptions

2010-04-13 11:29:30 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/udpsink.c:
	  check: Use fail_unless_equals_int instead of fail_if
	  Makes the error message more interesting

2010-11-30 19:22:11 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Also extract IDIT tags present too early
	  https://bugzilla.gnome.org/show_bug.cgi?id=636143

2010-11-30 19:21:23 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Also emit DateTime tag
	  https://bugzilla.gnome.org/show_bug.cgi?id=636143

2010-12-03 00:22:48 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: detect DTS advertised as PCM correctly in some more cases
	  The DTS typefinder may return a lower probability for frames that start
	  at non-zero offsets and where there's no second frame sync in the first
	  buffer. It's fairly unlikely that we'll acidentally identify PCM data
	  as DTS, so we don't do additional checks for now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=636234

2010-11-08 17:11:42 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/Makefile.am:
	  tests: makefile cleanup
	  Fix indentation. Use $(GST_MAJORMINOR) instead of hardcoded 0.10.

2010-11-08 17:02:56 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/Makefile.am:
	* tests/check/pipelines/.gitignore:
	* tests/check/pipelines/wavenc.c:
	  tests: add a test for wav muxing

2010-11-08 16:57:17 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/elements/interleave.c:
	* tests/check/pipelines/wavpack.c:
	  tests: remove newlines between variable decls (old gst-indent failure)

2010-11-08 14:47:04 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/libpng/gstpngdec.c:
	  pngdec: use png_error() as recommended by libpng docs to signal an error
	  Without that the element loops endlessly on broekn pngs. Fixes #634314

2010-11-16 17:48:16 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Parse and use creation time tag from mvhd
	  Expose creation time from mvhd as a datetime tag
	  Fixes #634928

2010-10-27 19:15:20 +0200  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/icydemux/gsticydemux.c:
	  icydemux: Add 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag

2010-10-23 19:34:00 -0400  Tom Janiszewski <Tom.Janiszewski@alcatel-lucent.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix for nellymoser codecid setting
	  Fixes bug #632897.

2010-10-21 16:15:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Add support for E-AC3

2010-10-21 16:14:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Add support for DTS

2010-10-31 18:08:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't send seeks behind the end of file to the server
	  Also improve debug output, re-initialize the content size and let the
	  seek handler error out on invalid seek segments.
	  Fixes bug #632977.

2010-12-02 17:53:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kpay.c:
	  j2kpay: use SOP markers to split bitstream
	  When parsing the bitstream, look for SOP markers because we are allowed to split
	  packets on those marker boundaries.
	  Rework the parsing code a little so that we can pack multiple Packetization
	  units in one RTP packet.

2010-11-18 12:49:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpj2kpay.h:
	  rtpj2kpay: use buffer lists
	  Use buffer lists for doing zerocopy payloading.
	  Add property to disable buffer lists.

2010-11-16 16:54:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  h264pay: small cleanups
	  Allocate adapter only once.
	  Make some guint8 * const.

2010-11-16 15:39:24 +0100  Tambet Ingo <tambet at gmail.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: implement full bytestream scan mode.
	  Implement the full bytestream scan mode.
	  Fixes #634910

2010-11-15 10:52:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/client-H263p-AMR.sh:
	* tests/examples/rtp/client-H263p-PCMA.sh:
	* tests/examples/rtp/client-H263p.sh:
	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-H264.sh:
	* tests/examples/rtp/client-PCMA.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  examples: improve RTP examples
	  Make the examples use autovideosink and ffmpegcolorspace for better
	  compàtibility.
	  Make some more variables for the sink and the decoders.
	  Set zerolatency tuning on x264enc for better realtime results.

2010-11-10 11:04:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: select multicast transports in a smarter way
	  When we see a multicast address in the SDP connection, only try to negotiate a
	  multicast transport with the server.
	  Fixes #634093

2010-12-02 18:14:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  Bump GLib requirement to implicit requirement
	  ie. >= 2.20 while we depend on core/base 0.10.31

2010-12-02 18:13:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development

=== release 0.10.26 ===

2010-12-01 21:15:09 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.26

2010-11-30 15:28:50 -0800  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: analyse RFF fields in correct order
	  Code was repeating the second field, not the first.
	  Fixes: #636179.

2010-11-29 15:32:40 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle stale digest authentication session data
	  In particular, handle Unauthorized server response when trying to convey
	  keep-alive.
	  Fixes #635532.

2010-11-26 15:00:29 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: fix segfault on empty payload
	  https://bugzilla.gnome.org/show_bug.cgi?id=635843

2010-11-25 19:24:56 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/audiofx/gststereo.c:
	  stereo: Remove dead assignments

2010-11-25 19:06:27 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  dtmf: Remove dead assignments

2010-11-18 00:45:29 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.25.5 pre-release

2010-11-18 00:44:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/bg.po:
	* po/fi.po:
	* po/hu.po:
	* po/sk.po:
	* po/tr.po:
	  po: update translations

2010-11-14 00:18:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix reference leak

2010-11-12 23:59:06 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Flush QoS and history before applying segment
	  When handling newsegment, flush out the buffer history in the
	  existing segment, not the new one. Fixes playback in some DVD
	  cases.
	  Partially fixes #633294

2010-11-12 12:20:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: improve event logging

2010-11-05 17:00:15 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Implement field history flushing
	  In a number of cases it is necessary to flush the field history by
	  performing 'degraded' deinterlacing - that is, using the user-chosen
	  method for as many fields as possible, then using vfir for as long as
	  there are >= 2 fields remaining in the history, then using linear for
	  the last field.
	  This should avoid losing fields being kept for history for example at
	  EOS.
	  This may address part of #633294

2010-11-05 15:44:35 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Refactor chain function
	  This is needed to be able to output a frame from outside the chain
	  function, i.e. in the following commit that adds flushing of the field
	  history.

2010-11-05 17:17:56 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: we still require Gtk+ >= 2.14.0 when compiling against 2.0
	  The check for the minor version was dropped in the previous commit.

2010-11-05 16:24:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: add --with-gtk option and default to Gtk+ 2.0 while the 3.0 API is still in flux
	  https://bugzilla.gnome.org/show_bug.cgi?id=634014

2010-11-04 16:42:07 +1000  Jonathan Matthew <jonathan@d14n.org>

	* gst/icydemux/gsticydemux.c:
	  icydemux: fix use-after-free of taglist
	  Broken by commit 4c2f5333 (bug #630205).
	  https://bugzilla.gnome.org/show_bug.cgi?id=633970

2010-11-01 17:29:01 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.25.4 pre-release

2010-11-01 17:28:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/es.po:
	* po/fr.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/pl.po:
	* po/sl.po:
	* po/sv.po:
	  po: update translations

2010-11-01 16:04:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: fix --disable-external

2010-11-01 14:56:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: only set delta unit on all-non-key units
	  Only set the delta flag when all of the units in the packet are delta units.
	  Based on patch from Olivier Crête <olivier.crete@collabora.co.uk>
	  Fixes #632945

2010-10-26 15:44:37 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: Return not-negotiated when bps is unknown
	  If caps weren't negotiated, goom should return not-negotiated
	  from its chain functions instead of using bps unitialized, which
	  leads to a division by 0
	  https://bugzilla.gnome.org/show_bug.cgi?id=633212

2010-10-27 13:16:54 +0100  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Automatic update of common submodule
	  From 7bbd708 to 011bcc8

2010-10-26 16:54:11 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Forward src pad events upstream.
	  Fix passing navigation and other events upstream by actually sending them.
	  Fixes: #633205

2010-10-24 18:50:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix deadlock in error code path
	  GST_ELEMENT_ERROR must not be called with the object lock held,
	  since it will call gst_object_get_parent() internally, which
	  takes the object lock as well.

2010-10-20 10:21:48 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Remove useless clearing of send_xiph_headers for Dirac
	  This looks like a mistake when copy-pasting the Theora code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=632682

2010-10-20 13:28:28 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: don't crash if vorbis/theora codec data is missing
	  Error out properly in this case instead of crashing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=632682

2010-10-22 18:11:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.25.3 pre-release

2010-10-19 16:45:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix duration reporting
	  Init segment prior to storing duration info in it.
	  Fixes #632548.

2010-10-19 14:21:53 +0100  Bastien Nocera <hadess@hadess.net>

	* gconf/Makefile.am:
	  gconf: Don't install schemas when GConf is disabled
	  https://bugzilla.gnome.org/show_bug.cgi?id=632553

2010-10-19 13:43:14 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/audiofx/gststereo.c:
	  various (gst): add missing G_PARAM_STATIC_STRINGS flags
	  Canonicalize property names as needed.

2010-10-19 13:43:14 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  various (gst): add missing G_PARAM_STATIC_STRINGS flags
	  Canonicalize property names as needed.

2010-10-19 13:44:25 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/dtmf/gstdtmfsrc.c:
	  dtmfsrc: remove DEBUG_FUNCPTR from gobject vmethods

2010-10-19 12:20:40 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/lame/gstlame.c:
	  various: canonicalize property names

2010-10-19 10:06:33 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  various (ext): add a missing G_PARAM_STATIC_STRINGS flags

2010-10-16 15:43:53 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  win32: set GST_PACKAGE_RELEASE_DATETIME also in win32 config.h

2010-10-16 01:33:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.25.2 pre-release

2010-10-16 01:26:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/el.po:
	* po/vi.po:
	  po: update translations

2010-10-15 13:22:03 -0700  David Schleef <ds@schleef.org>

	* tests/check/Makefile.am:
	  tests: Don't dist generated orc files

2010-10-15 14:02:19 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	  Update generated orc code

2010-10-15 18:00:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: bump Orc requirement to 0.4.11

2010-10-14 17:41:30 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Use the right constant to define the "use-pipeline-clock" property
	  The wrong #define was being used, now use the correct one.

2010-10-14 12:31:48 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From 5a668bf to 7bbd708

2010-10-14 17:26:14 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/qtdemux/qtdemux.c:
	  ac3: demuxers provide framed output

2010-10-14 00:11:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	  matroskamux: reduce newsegment event spam and set discont flag where needed
	  Only send newsegment events with new positions downstream when actually
	  needed, instead of sending multiple newsegment events with new seek
	  positions in a row. Also set the discont flag on buffers after a
	  discontinuity.

2010-10-13 23:46:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	  matroskamux: set correct buffer offsets after seeks
	  Re-use the existing 'pos' field maintained by ebml writer to set
	  buffer offsets. This also makes sure that we set the right offsets
	  on buffers after a seek (e.g. when writing an index at the end).

2010-10-14 00:22:03 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: don't forward tag events downstream
	  Don't forward stream-specific tag events downstream (esp. not
	  before any newsegment event).x

2010-10-13 17:15:25 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: handle another mp4v variation
	  ... including the glbl atom containing codec-data.

2010-10-13 17:21:23 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavimux.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/efence.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/negotiation.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/id3demux/gstid3demux.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska-mux.c:
	* gst/median/gstmedian.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/smpte/gstsmpte.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstvideotemplate.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	  various (gst): add a missing G_PARAM_STATIC_STRINGS flags

2010-10-13 17:13:04 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/oss/gstossmixerelement.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-mixer.c:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/ximage/gstximagesrc.c:
	  various (sys): add a missing G_PARAM_STATIC_STRINGS flags

2010-10-13 16:25:15 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/aalib/gstaasink.c:
	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/annodex/gstcmmltag.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/esd/esdmon.c:
	* ext/esd/esdsink.c:
	* ext/flac/gstflacenc.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/hal/gsthalaudiosink.c:
	* ext/hal/gsthalaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libpng/gstpngenc.c:
	* ext/mikmod/gstmikmod.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/wavpack/gstwavpackenc.c:
	  various (ext): add a missing G_PARAM_STATIC_STRINGS flags

2010-10-13 16:34:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/aalib/gstaasink.c:
	* ext/esd/esdmon.c:
	* gst/median/gstmedian.c:
	  various: wrap property registration and add a single fixme for long desc.

2010-10-13 11:46:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: always mark the codec_data as keyframe
	  We need to mark the codec_data as a keyframe or else downstream decoders might
	  decide to skip it, waiting for a keyframe.
	  Fixes #631996

2010-10-13 07:16:47 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/ebml-write.c:
	  matroskamux: make buffer offsets a byte count rather than a buffer count

2010-10-07 21:12:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/dv/gstdvdec.c:
	* ext/esd/esdmon.c:
	* ext/flac/gstflacenc.c:
	* ext/mikmod/gstmikmod.c:
	* ext/raw1394/gstdv1394src.c:
	* gst/debugutils/efence.c:
	* gst/rtpmanager/gstrtpbin.c:
	  ext, gst: canonicalise property names where this wasn't the case
	  ie. "foo_bar" -> "foo-bar"

2010-10-12 15:02:42 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtp/gstrtpmpvpay.c:
	  rtpmpvpay: fix timestamping of rtp buffers
	  Incomming buffer is only pushed on the adapter at the end of the
	  handle_buffer function. But duration/timestamp of this buffer is already
	  taken into account for the current data in the adapter. This leads to
	  wrong rtp timestamps and extra latency.

2010-10-12 11:37:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/equalizer/demo.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  examples: Fix build with GTK+ 3.0

2010-10-11 15:12:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: mark as a source
	  Mark the rtspsrc element as a source.
	  Requires 0.10.31.1 now

2010-10-11 14:24:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosrc.c:
	  autodetect: Set GST_ELEMENT_IS_SOURCE flag on sources

2010-10-11 14:21:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstswitchsrc.c:
	  switchsrc: Set the GST_ELEMENT_IS_SOURCE flag

2010-10-11 14:17:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Require core 0.10.30.1

2010-10-10 14:43:58 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	  matroskamux: set offsets on outgoing buffers

2010-10-09 14:14:27 +0200  IOhannes m zmölnig <zmoelnig@iem.at>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Only get/set overlay params if needed
	  it's perfectly ok for a video output device to not have overlay capabilities.
	  this patch removes the need to get/set the overlay parameters if the user
	  does not explicitely request one of the overlay properties

2010-09-30 15:28:23 +0200  IOhannes m zmölnig <zmoelnig@iem.at>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Protect against NULL-pointer access
	  gst_v4l2sink_change_state() would free the pool without checking whether there
	  was a valid pool...

2010-10-08 12:43:51 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From c4a8adc to 5a668bf

2010-10-08 12:53:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 5e3c9bf to c4a8adc

2010-10-06 11:29:55 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix required fields logic
	  Both history_count and fields_required count from 1. As per the while loop
	  condition that follows this code, to perform the deinterlacing method, we need
	  history_count >= fields_required fields in the history. Therefore if we have
	  history_count < fields_required (not fields_required + 1), we need more fields.

2010-09-20 19:43:45 +0200  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: resend onMetada tag when tags changes in streamable mode

2010-10-05 19:40:50 +0100  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: AAC codec_data can be > 2 bytes long
	  This fixes the assumption that DecoderSpecificInfo must be 2 bytes long
	  for AAC files. The specification allows HE-AAC to be explicitly
	  signalled in a backward compatible way. This is done by means of an
	  additional information after the regular AAC header. It is expected that
	  decoders that can play AAC but not HE-AAC will parse the header normally
	  and ignore extended bits, much as they do for the HE-AAC specific payload
	  in the actual stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=612313

2010-10-05 16:01:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: only unref buffer when no longer needed for cluster scanning
	  Fixes #629047.

2010-10-05 16:00:45 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: avoid infinite cluster scanning

2010-10-05 12:20:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	  goom: take duration into account when doing QoS
	  Take the duration of the frames into account so that we don't drop frames that
	  are only partially past the QoS deadline.

2010-10-05 10:40:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom/gstgoom.c:
	* gst/goom/gstgoom.h:
	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	  goom: use adapter for timestamping
	  Use the adapter timestamp code to get more accurate timestamps.
	  Fix latency calculation, we add our own latency in the worst case.

2010-10-04 22:31:32 +0200  Edward Hervey <bilboed@bilboed.com>

	* configure.ac:
	* ext/raw1394/Makefile.am:
	* ext/raw1394/gst1394.c:
	  raw1394: Don't compile hdv1394src if libiec61883 isn't available
	  Fixes #629896

2010-09-20 19:44:09 +0200  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/icydemux/gsticydemux.c:
	  icydemux: forward tag events
	  https://bugzilla.gnome.org/show_bug.cgi?id=630205

2010-10-04 19:00:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom2k1/gstgoom.c:
	  goom2k1: report our latency correctly
	  Fixes #631303

2010-10-04 18:56:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom2k1/gstgoom.c:
	  goom2k1: add defines for default width/height/fps
	  Add some defines for the default width/height/fps instead of using different
	  values in different places.

2010-10-04 18:52:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: add latency compensation code.
	  Implement a latency query and report how much latency we will add to the
	  stream.
	  Alse make some defaults for the default width/height/framerate
	  Fixes #631303

2010-10-04 17:56:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/server-alsasrc-PCMA.py:
	  test: add python version of the audio sender
	  Add a python version of the audio sender pipeline.
	  Ported by Sp4rc on IRC.

2010-10-04 17:52:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/client-PCMA.py:
	  tests: Add python RTP client example
	  Add a python version of the PCMA client app.
	  Ported by Sp4rc on IRC.

2010-10-04 09:39:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpmp4gpay.c:
	  rtp: Fix unitialized compiler warnings on OS X build bot
	  These warnings are wrong though, the variables are only used in
	  the cases where they *are* initialized by the bit reader.

2010-10-03 23:49:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpg722pay.c:
	  rtpg722pay: Fix uninitialized variable compiler warning
	  The clock rate is always 8000 Hz according to the RFC and
	  the sampling rate must always be 16000 Hz.

2010-10-01 13:59:10 +0400  Vladimir Eremeev <eremeev@atlantis.ru>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: improve article reference in comment block
	  https://bugzilla.gnome.org/show_bug.cgi?id=631082

2010-04-30 21:00:31 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/quicktime.c:
	  qtdemux: Use pbutils for H.264 profile/level extraction
	  The functions used to extract this data have been moved to gstpbutils to
	  facilitate reuse.
	  https://bugzilla.gnome.org/show_bug.cgi?id=617318

2010-04-30 21:00:31 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/matroska/Makefile.am:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska.c:
	  matroskademux: Use pbutils for H.264 profile/level extraction
	  The functions used to extract this data have been moved to gstpbutils to
	  facilitate reuse.
	  https://bugzilla.gnome.org/show_bug.cgi?id=617318

2010-04-22 19:39:47 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Export MPEG-4 video profile and level in stream caps
	  This uses gstpbutils to extract the profile and level from the video
	  object sequence and adds this to stream caps. This can be used as
	  metadata and for fine-grained decoder selection.
	  https://bugzilla.gnome.org/show_bug.cgi?id=616521

2010-09-30 12:44:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix aac channel override based on codec data for 7.1 case

2010-04-30 14:06:27 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/qtdemux/Makefile.am:
	* gst/qtdemux/qtdemux.c:
	  qtdemux: Export AAC profile and level in caps
	  This exports the AAC profile and level in caps for use as metadata and
	  (eventually) for more fine-grained selection of decoders at
	  caps-negotiation time. (Doesn't work for HE-AAC yet though.)
	  https://bugzilla.gnome.org/show_bug.cgi?id=612313

2010-09-30 18:34:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722depay.h:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg722pay.h:
	  rtp: add G722 pay and depayloader

2010-09-30 12:08:49 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: update link to documentation

2010-09-30 11:34:56 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* tests/examples/rtp/client-H264.sh:
	  examples: fix indentation on rtp client example

2010-09-30 11:33:24 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-H264.sh:
	  examples: fix typo in port of rtp examples

2010-09-29 13:20:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	  wavenc: miniscule code clean-up
	  GST_CLOCK_TIME_NONE is not something that should be used in connection with
	  GST_FORMAT_BYTES.

2010-09-29 10:34:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: reverse playback; prevent overlap of subsequent fragments

2010-09-28 16:21:48 +0300  René Stadler <rene.stadler@nokia.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix missing null-terminator in protocols array
	  Fixes random crash regression from commit ae84ae.

2010-09-24 16:26:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't add /UDP in the transport, it's the default
	  don't add the default UDP lower-transport, some servers don't seem to like it.
	  Fixes #630500

2010-06-25 17:08:03 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpmanager: packet lost should not be a warning. It happens all the time...

2010-09-24 15:33:40 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpbin: Make cleaning up sources in rtp_session_on_timeout MT safe
	  Using _foreach_remove on the hashtable, while releasing the lock protecting
	  that table inside the callback is not a good idea. The hashtable might
	  then change (a source removed or added) while signals like on_timeout
	  are being sent.
	  This solution makes a copy of the table, performs the _foreach without
	  actually removing any sources, but marks them for removal on a second
	  iteration with the real list, but this time not letting go of the lock.
	  Fixes #630452

2010-09-24 15:19:15 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/id3demux/id3tags.c:
	  id3demux: Sanitize id3 frame names
	  This is similar to what is done in qtdemux. Avoids providing invalid
	  structure/tags names

2010-09-24 14:59:45 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/apetag/gstapedemux.c:
	  apedemux: Skip empty tags
	  Avoid creating bogus string tags. Also added logging of the string
	  values of the tag name and value.

2010-09-24 08:56:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  soup: init debug category before using it

2010-04-12 09:49:14 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Handle rysnc of iterator when looking for free pad name
	  If a new pad was added while iterating then a pad could be
	  returned that was already in use.
	  Fixes #630451

2010-09-24 14:09:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: fix compilation

2010-04-07 15:31:52 +0200  Trond Andersen <trond.andersen@tandberg.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Unlock before adding pad in new_payload_found
	  Holding internal locks while potentially calling out is a source
	  of deadlocks, and in this case the application might subscribe to the
	  pad-added signal.
	  Fixes #630449

2009-08-31 18:37:40 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: relax third-party collision detection
	  If the source has been inactive for some time, we assume that it has
	  simply changed its transport source address. Hence, there is no true
	  third-party collision - only a simulated one.
	  Fixes #630447

2010-09-24 13:50:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: whitespace fixes

2010-09-24 13:48:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: simplify the rate estimation some more

2009-08-31 18:34:08 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpmanager: provide additional statistics

2010-09-24 00:01:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: set plugin release datetime

2010-09-23 21:21:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/equalizer/gstiirequalizer10bands.h:
	* gst/equalizer/gstiirequalizer3bands.h:
	* gst/equalizer/gstiirequalizernbands.h:
	  equalizer: fix class definitions
	  Class structures must be based on the parent class struct, not on
	  the parent instance struct.

2010-09-15 20:36:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: pre-register pad class properly with g_type_class_ref
	  Fix code to match the comment. Also, there's no need to register the
	  background enum type again, this is already done via install_property.

2010-09-23 21:57:18 +0200  David Hoyt <dhoyt@llnl.gov>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	  speex: Fix crashes with MSVC
	  Using the symbols for the different Speex modes results
	  in crashes when using MSVC. Use the library functions to
	  get the modes instead.
	  Fixes bug #630378.

2010-08-24 13:25:02 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/level/gstlevel.c:
	  level: avoid division by zero on silence
	  Fixes bug #630458.

2010-09-23 16:46:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: parse and use cts
	  For H264, there is an extra header containing the CTS, which is a timestamp
	  offset that should be applied to the PTS. Parse this value and use it to adjust
	  the pts.
	  Fixes #630088

2010-09-23 16:45:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: improve pts debugging

2010-09-22 19:01:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* tests/examples/Makefile.am:
	* tests/examples/pulse/.gitignore:
	* tests/examples/pulse/Makefile.am:
	* tests/examples/pulse/pulse.c:
	  pulse: add test app for pulse device probe

2010-09-22 18:50:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulse: fix device_description in READY
	  Make the is_dead check more clear and add an option to check for the status of
	  the stream in addition to the context.
	  We don't need a stream to get the device_description string.
	  Fixes #630317

2010-09-22 12:56:00 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Don't post tags if there are none
	  And make all code go through _post_global_tags.

2010-09-22 12:37:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: refactor and simplify AU merging
	  Move the processing of the NALU to a separate method.
	  Simplify the merging of NALU into AU and use common code when possible.

2010-09-21 23:23:07 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/shapewipe/shapewipe-example.c:
	  shapewipe: add optional border parameter and slowdown animation
	  Allow to play with the border property (sharp/soft edges).

2010-09-21 19:14:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: Force format to AYUV in the example pipeline for the same reason

2010-09-21 19:13:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/shapewipe/shapewipe-example.c:
	  shapewipe: Force the input to AYUV to prevent negotiation failures in videomixer
	  The second videotestsrc chain might produce YUY2 because everything is
	  accepted downstream before the first shapewipe chain gets negotiated.

2010-09-21 19:12:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: Improve debugging and immediately return empty caps from the getcaps functions

2010-09-21 18:33:55 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From aa0d1d0 to 5e3c9bf

2010-09-21 12:49:31 +0200  Philippe Normand <pnormand@igalia.com>

	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/gstv4l2xoverlay.h:
	  v4l2: use the xoverlay APIs

2010-09-21 12:48:34 +0200  Philippe Normand <pnormand@igalia.com>

	* configure.ac:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: use the new xoverlay APIs
	  Also bumped -base requirements.

2010-09-21 12:31:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Use -DGST_DISABLE_DEPRECATED again for GIT versions

2010-09-21 11:52:22 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Fix debug statement

2010-09-20 23:17:35 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Parse uuid atoms in push mode
	  Parses uuid atoms in push mode when they are found, they might
	  contain xmp tags.
	  Also does a minor refactoring to put the global tags posting
	  into a single function instead of repeating it in 3 different
	  places.
	  Fixes #629839

2010-09-16 08:04:02 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Delay tags posting a little
	  Delay tags posting until we've parsed all the headers so
	  that the native and xmp tags get merged before posting
	  https://bugzilla.gnome.org/show_bug.cgi?id=629839

2010-09-15 22:13:43 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: Parse xmp packet in uuid atom
	  xmp packet is placed into a top-level uuid atom for
	  isom/mp4 variants.
	  This patch makes qtdemux parse all top-level atoms
	  in pull-mode before starting to push data, making
	  it able to find those tags.
	  https://bugzilla.gnome.org/show_bug.cgi?id=629839

2010-09-17 11:07:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpstats.c:
	  rtpstats: printf format fixes

2010-09-17 11:07:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpg729pay.c:
	  rtppay: some printf format fixes

2010-09-15 18:21:11 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix logic when pushing EOS.
	  Don't check for return values when pushing EOS. Still post an error if EOS is
	  reached and no streams have been found.

2010-09-15 17:02:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	  docs: add gtk-doc chunks with Since: markers for new v4l2src properties

2010-09-15 18:43:50 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/v4l2/camctrl.c:
	  camctrl: add license header to demo

2010-09-14 17:41:28 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: don't send EOS twice on the same pad.

2010-09-14 10:07:58 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: move the shared mainloop from class to static var
	  Just have one static var for the shared mainloop instead of one class variable
	  and copies in the instance.

2010-09-13 17:31:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: cleanups for DRI markers
	  Protect against invalid DRI markers.
	  do some cleanups

2010-09-10 11:35:53 -0400  American Dynamics <GStreamer-Bugs@tycosp.com>

	* gst/rtp/gstrtpjpegpay.c:
	  gstrtpjpegpay: Added Define Restart Interval (DRI) Marker
	  Added ability to detect and respond to a JPEG-defined DRI marker

2010-06-19 19:20:18 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: Split getting the caps into its own function

2010-09-13 16:03:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: small cleanup.

2010-09-13 16:24:26 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: rework context sharing
	  We also need to share the main-loop threads as this owns the context. Thus have
	  a class wide main-loop thread. From this we create a context per client-name.
	  Instead of always looking up the context, we keep this with the instance. The
	  reverse mapping is only needed in pulse singal handlers. This saves a lot of
	  locking. Also one signal handler becomes simpler as ther eis only one mainloop
	  to notify.
	  Now valgind happy - no leaks, no bad reads/writes.
	  This reverts major parts of commit 69a397c32f4baf07a7b2937c610f9e8f383e9ae9.
	  Fixes #628996

2010-09-13 15:44:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpstats.c:
	  rtpsession: Small cleanups
	  Make the property description prettier.
	  Actually multiple the bandwidth with the fraction.

2010-06-01 21:35:40 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpsession: Calculate RTCP bandwidth as a fraction of the RTP bandwidth
	  Calculate the RTCP bandwidth to be a fraction of the RTP bandwidth if it is
	  specified as a value between 0 and 1.

2010-09-13 15:29:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: improve bandwidth recalculation
	  Also recalculate bandwidth when one of the source bandwidths changed.
	  Use the newly calculated bandwidth.

2010-06-01 21:17:26 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Add the option to auto-discover the RTP bandwidth

2010-09-13 14:38:11 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: set use-pipeline-clock on correct GObject

2010-06-02 17:51:12 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Initialise the average scaled by 16

2010-09-13 12:41:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: add running_time argument docs

2010-06-23 16:13:01 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpstats.h:
	  rtpstats: Rectify description of current_time in RTPArrivalStats
	  It is the current time, it is unrelated to when the packet was actually received.

2010-09-13 12:31:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: compute the average correctly scaled

2010-06-01 20:31:18 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Count sent RTCP packets after they have been finished
	  If they are counted before calling gst_rtcp_buffer_end(), then the
	  size is way too big.

2010-06-01 19:51:34 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: Don't unref  pads in finalize
	  The gstrtpsession object is not holding any reference to them directly

2010-09-12 00:09:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/POTFILES.in:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update translations for new souphttpsrc messages

2010-09-12 00:08:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  soup: hook up i18n bits for plugin
	  Call bindtextdomain() etc.

2010-09-12 00:04:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  soup: fix error messages
	  Error messages should be translated. URIs and filenames should not
	  be part of the error message string that's shown to the user.
	  soup_message->reason_phrase is not translated and not suitable as
	  error message for users (see libsoup documentation). Also fix up
	  error codes a bit, as far as possible with the existing codes.

2010-09-10 09:43:24 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: don't post an error message if buffer alloc fails with NOT_LINKED flow
	  This is not fatal, let upstream handle it.

2010-09-10 18:06:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't clear sdp when set as uri
	  when we set the SDP with an uri, don't clear it when we go to READY.

2010-09-10 18:01:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: use sdp uri parse method
	  Use the sdp parse method that does proper uri escaping.

2010-09-10 16:59:10 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/v4l2/.gitignore:
	* tests/examples/v4l2/Makefile.am:
	* tests/examples/v4l2/camctrl.c:
	  example: add v4l2 example, demonstrating the use of gst controller

2010-09-10 16:55:25 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: don't skip calculating the duration

2010-06-22 15:48:04 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2src: add controlable colorbalance parameters
	  Expose colorbalance controls as object properties (like we do on xvimagesink).
	  Make them controlable.

2010-09-10 13:25:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: fix some mis-implementation
	  Also add some debug.

2010-09-10 13:24:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: properly insert dummy buffers

2010-09-10 11:55:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add rtsp-sdp protocol support
	  Allow setting an SDP with the rtsp-sdp:// url.
	  Based on patch from Marco Ballesio.
	  See #628214

2010-09-10 11:35:58 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: make passthrough work.

2010-09-09 21:43:40 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtp/gstrtpmp4adepay.c:
	  mp4adepay: small logging cleanup and addition to debug config parsing

2010-09-09 21:42:46 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/aalib/gstaasink.c:
	  aasink: fix context initialisation and freeing to not leak

2010-09-09 21:40:51 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/check/Makefile.am:
	* tests/check/generic/states.c:
	  tests: allow running state tests for all elements
	  Now one can use GST_NO_STATE_IGNORE_ELEMENTS=1 make generic/states.check
	  to try elements that would normaly be skipped.

2010-09-09 18:47:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtp-payloading.c:
	  tests: fix rtpjpegpay test
	  Make the data we send to the jpeg payloader be a valid jpeg file because the
	  payloader now expects this.

2010-09-09 18:47:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: improve debugging

2010-09-09 16:31:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: use valid bitrate for dummy frame

2010-09-08 17:07:53 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/taglib/gstid3v2mux.cc:
	  id3v2mux: Adds mapping for album artist
	  Maps GST_TAG_ALBUM_ARTIST to TPE2 in id3v2mux

2010-09-08 18:35:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Require orc 0.4.8
	  The deinterlace plugin apparently fails to compile with older versions.

2010-09-08 17:50:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: QoS handling logic only applies to forward playback
	  Fixes #628894.

2010-09-08 17:43:47 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: remove unused code

2010-09-08 14:36:48 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: fixup last commit
	  We need to prevent the eventual leak better.

2010-09-08 14:16:58 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: code cleanups
	  Use g_slist_prepend as we don't care about the order. Check for list == NULL
	  instead of iterating the list to see if it is empty. Move ctx allocation down
	  to prevent leak in case of failure.

2010-09-08 07:13:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Fix uninitialized variable compiler warning
	  Fixes bug #629018.

2010-09-07 19:02:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: simplify clock provide code
	  Don't leak the pulsesink element by having the clock keep a ref to the sink.
	  Create the clock only once in the constructor and use the baseaudiosink clock
	  cleanup code.

2010-09-07 17:49:05 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: move the context table init to _get_type phase
	  This seems to fix the invalid reads on context shutdown better, altough
	  I can't really explain.

2010-09-07 17:06:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use older g_array_free
	  g_array_unref() is only since 2.22

2010-09-07 16:49:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: avoid invalid adapter flush on QoS
	  First store the available data in the adapter in the rem_img_len instance field
	  before trying to flush the adapter with that value on QoS.

2010-09-07 16:40:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: do some more sanitity checks
	  Protect some more against invalid input.

2010-09-07 15:20:12 +0200  American Dynamics <GStreamer-Bugs at tycosp.com>

	* gst/rtp/gstrtpjpegpay.c:
	  jpegpay: handle corrupted jpeg better
	  Protect against corrupted jpeg input.

2010-09-07 13:55:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  rvawdepay: cleanup unused fields

2010-09-07 13:51:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  vrawdepay: handle invalid payload better
	  Make sure we don't read more data than available in the input buffer.
	  Clip the input data into the output buffer.

2010-08-16 15:35:51 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulse: allow setting stream properties
	  Add a "properties" property to the elements to allow setting extra stream
	  properties.
	  Fixes #537544

2010-09-07 12:08:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf3.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: remove introspection info for gdkpixbuf3 plugin and update version for others
	  The versions got accidentally reverted to a pre-release version, fix that.

2010-09-07 11:42:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From c2e10bf to aa0d1d0

2010-09-07 09:20:03 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	  cmmldec: fix flow return handling
	  Fix buggy GST_FLOW_IS_FATAL substitution, and 'make check':
	  -  if (!GST_FLOW_IS_FATAL (dec->flow_return) && !dec->sent_root) {
	  +  if (dec->flow_return != GST_FLOW_OK && !dec->sent_root) {

2010-09-07 00:27:07 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: don't free the context multiple times
	  Apparently the close function of the ring-buffer can be called multiple times.

2010-08-12 12:33:06 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtp/gstrtpmp4adepay.c:
	  rtpmp4adepay: grab the sampling arte and put into caps
	  This is needed to be able to mux the received audio into mp4 (in the case of
	  aac). Fixes #625825.

2010-09-06 14:40:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	  rtp: mark constant tables as const

2010-08-18 14:40:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpamrpay.h:
	  rtpamrpay: properly support perfect-rtptime

2010-08-18 11:42:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpamrpay.c:
	  rtpamrpay: proper duration for multiple frame payload

2010-08-18 11:42:07 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	  rtpamr(de)pay: support AMR-WB SID frame

2010-08-18 11:39:06 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpg729pay.h:
	  rtpg729pay: properly support perfect-rtptime

2010-08-16 16:08:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: improve framerate determining
	  Collect a limited number of starting sample durations and use the median of
	  those to determine caps framerate.

2010-08-17 12:08:10 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: attempt more resync upon (cluster) parse error
	  That is, if parse error occurs in state requiring to move to next cluster,
	  and doing so to the expected next position of cluster fails, then scan for a
	  next cluster from present position and resume from there.
	  Fixes #620790.

2010-08-16 16:05:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: not so fatal error handling
	  If some bits out of place in block(group) parsing, forego and move to next.
	  Also skip large blocks in pull mode, but need to give up in push mode.
	  Fixes #626463.
	  Improves #620790.

2010-07-26 15:51:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: additional parse recovery
	  In particular, upon parse failure in one cluster, we may forego remaining
	  content and try resuming from next cluster onwards.
	  Fixes #620790.

2010-08-26 02:54:55 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  dtmfsrc: Make the dtmfsrc accept events sent with gst_element_send_event
	  The doc says to use gst_element_send_event on the pipeline, but if
	  we are to call it on the element itself, it's a noop. This should make it
	  handle the event properly before delegating it to basesrc.

2010-09-06 12:22:11 +0200  American Dynamics <GStreamer-Bugs at tycosp.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add property to configure udpsrc buffer size
	  Add a new udp-buffer-size property to configure the buffer-size on the udpsrc
	  elements.
	  Fixes #628058

2010-08-27 17:58:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: add ntp-sync property
	  Add an ntp-sync property that will sync the received streams to the server
	  NTP time. This requires synchronized NTP times between the sender and receivers,
	  like with ntpd.
	  Based on patch from Thijs Vermeir.
	  Fixes #627796

2010-08-27 12:14:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rename a variable to avoid confusion

2010-08-27 11:07:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: rename some variables for less confusion

2010-08-27 10:41:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: move comment where it belongs

2010-08-26 16:00:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  session: minor cleanups
	  Make clock snapshots more accurate by only sampling the same clock once.

2010-08-26 10:58:26 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin: add use-pipeline-clock property
	  With this property RTCP SR NTP times can be based
	  on the system clock (maybe synced with ntpd) or the
	  current pipeline clock.
	  https://bugzilla.gnome.org/show_bug.cgi?id=627796

2010-08-25 09:58:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspext.c:
	  rtspext: stop configuration on first failure
	  Stop the configuration of a stream as soon as some of the extensions return
	  FALSE.
	  Fixes #581294

2010-08-20 15:35:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	  multifdsink: use refcount to count host/port duplicates
	  Instead of adding multiple client structures for the same host/port pair, use a
	  refcount.
	  Add a send-duplicates feature that allows you to disable sending multiple copies
	  of the same packet to the same host when it was added multiple times. The
	  send-duplicates property is by default set to TRUE for backwards compatibility
	  although it is very likely that this is not desired behaviour.

2010-08-19 17:06:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: implement custom event handler
	  Extend the _push_event() function so that it can also send events to the udp
	  sources when asked.
	  Implement a custum send_event function that correctly dispatches the downstream
	  events in TCP mode. This fixes sending EOS to rtspsrc and have it push the EOS
	  downstream.

2010-08-19 11:37:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: use _get_caps_reffed() when we can
	  Use _get_caps_reffed()
	  Add some more debug when opening the server connection.

2010-08-16 11:29:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegdepay.h:
	  jpegdepay: handle DISCONT and reset state
	  Put a DISCONT event on the next output buffer when the input buffer had a
	  DISCONT.
	  Make sure we clear our adapter and reset our state before going to PAUSED.
	  Free the qtables.
	  Fixes #626869

2010-08-16 11:27:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.h:
	  g729pay: extend from right parent

2010-09-06 09:57:10 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: add since docs for new property.

2010-08-30 16:45:48 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use GST_BOILERPLATE macro

2010-08-16 17:23:58 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/videomixer/videomixer.c:
	  videmixer: add a example showing how to use the child properties
	  Show how to position and set the alpho of the videos on gst-launch.

2010-08-16 15:19:38 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: move the property-setter to the getter.

2010-08-11 15:48:18 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum only aggregate magnitude/phase if user asks for it

2010-08-11 15:45:56 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: improve performance with local vars
	  Use 'input' instead of 'spectrum->input' which was intende already (variable
	  exists, but not used everywhere). Also use a local version of
	  'spectrum->input_pos'.

2010-08-11 15:44:03 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: code cleanup
	  More comments and logging. Extract one complex condition to a variable. Reorder
	  some code for readability.

2010-08-11 15:40:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: improve property setter
	  consistently only update if the property actualy changed the value. Do it
	  without reading the gvalue twice. No need to reset the spectrum analyzer for
	  threshold changes.

2010-08-11 15:38:24 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	  spectrum: add helper to only flush ringbuffer data without resetting the fft
	  Reduces some duplicated code as well.

2010-08-11 12:45:53 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum: more comments

2010-09-05 22:22:42 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Document methods with bad quality

2010-09-05 22:19:56 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: initialize all deinterlace class members
	  This fixes UYVY deinterlacing.

2010-09-05 18:58:13 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From d3d9acf to c2e10bf

2010-09-05 18:45:21 -0700  David Schleef <ds@schleef.org>

	* gst/videomixer/blend.c:
	  videomixer: orc_init() doesn't need to be called
	  There's no need to call orc_init() unless you're using the Orc
	  API directly.  All code created by orcc is guaranteed to work
	  without calling orc_init().

2010-09-05 18:40:48 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime.orc:
	* gst/deinterlace/tvtime/greedy.c:
	  deinterlace: Fix greedyl Orc implementation
	  To agree with the previous C/asm code.

2010-09-05 22:31:34 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Fail when caps are incompatible
	  Do not forget to return false when caps are incompatible.

2010-09-05 20:56:52 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/videomixer/blend.c:
	  videomixer: Only init orc if it is available
	  Put some ifdef around orc_init to prevent build errors

2010-09-05 12:17:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From ec60217 to d3d9acf

2010-09-04 12:46:31 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	  deinterlace: Update disted Orc files

2009-06-29 11:43:07 -0700  David Schleef <ds@schleef.org>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2src: add decimate property

2010-06-04 12:09:23 -0700  David Schleef <ds@schleef.org>

	* ext/dv/Makefile.am:
	* ext/dv/gstdvdemux.c:
	* ext/dv/gstsmptetimecode.h:
	  dvdemux: Parse SMPTE time codes

2010-08-23 02:50:36 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	  deinterlace: remove assembly code in favor of orc

2010-06-08 14:54:49 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/tvtime.orc:
	* gst/deinterlace/tvtime/greedy.c:
	  deinterlace: implement greedy in Orc

2010-09-04 11:43:21 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	  update disted Orc files

2010-09-02 14:34:50 +0200  Thibault Saunier <tsaunier@gnome.org>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: Fix classification
	  This is no effect but a converter. Fixes bug #628608.

2010-09-02 11:19:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.types:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf3.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst/videomixer/Makefile.am:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	* gst/videomixer/videomixer2pad.h:
	  videomixer2: Add documentation and add to the docs

2010-07-26 16:07:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/gstcollectpads2.c:
	* gst/videomixer/gstcollectpads2.h:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer2: Add videomixer2 element
	  This is based on collectpads2 and is synchronizing
	  all streams based on the running time.
	  New features compared to old videomixer:
	  * Synchronizing frames on the running time
	  * Improved and simplified negotiation
	  * Full QoS support
	  * Variable framerate support
	  Fixes bug #626048, #624905.

2010-09-01 11:11:34 +0200  Pavel Kostyuchenko <shprotx@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Relax parsing of date tags
	  Before we required a complete date in matroskademux but in
	  id3demux for example only the year or year and month was possible too.
	  Fixes bug #628454.

2010-08-30 19:03:52 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Use GstBaseSrc::block-size as fallback size

2010-08-30 18:36:54 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Fix using mpegts via the mmap interface
	  MPEG doesn't have a static size per frame, so don't pretend it has one
	  and fail when capturing because it doesn't match. Instead mark the size
	  as unknown and let the read frame grabbing method use a reasonable fallback
	  value (assuming that's only for actual streaming formats)
	  Fixes bug #628349.

2010-08-27 18:15:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/wavpack/gstwavpackparse.c:
	  wavpackparse: Don't use GST_FLOW_IS_FATAL()

2010-08-27 18:13:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	  pngdec: Don't use GST_FLOW_IS_FATAL()
	  And don't post an error message if downstream returns UNEXPECTED.

2010-08-27 18:09:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Don't use GST_FLOW_IS_FATAL()

2010-08-27 18:05:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Don't use GST_FLOW_IS_FATAL()
	  And don't post an error message if buffer allocation failed because
	  of UNEXPECTED, which only means that downstream wants us to EOS now.

2010-08-27 18:02:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	  flacenc/dec: Don't use GST_FLOW_IS_FATAL()
	  And properly handle UNEXPECTED and WRONG_STATE.

2010-08-27 17:52:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	  cmmldec/enc: Don't use GST_FLOW_IS_FATAL()
	  And as a result, don't ignore WRONG_STATE and NOT_LINKED.
	  Both mean that it's a good idea to pass them upstream instead
	  of pretending that everything is good.

2010-08-27 17:47:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Don't use GST_FLOW_IS_FATAL()

2010-08-27 17:45:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't use GST_FLOW_IS_FATAL() and GST_FLOW_IS_SUCCESS()

2010-08-27 17:39:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Don't use GST_FLOW_IS_FATAL()

2010-08-27 17:37:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't use GST_FLOW_IS_FATAL()

2010-08-27 17:35:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/debugutils/rndbuffersize.c:
	  rndbuffersize: Don't use GST_FLOW_IS_FATAL()

2010-08-27 17:35:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't use GST_FLOW_IS_FATAL()

2010-08-27 17:32:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Don't use GST_FLOW_IS_FATAL()
	  And document why wrong-state doesn't need an error message.

2010-08-26 13:44:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Fail gracefully if no threaded PA mainloop can be created
	  Fixes bug #628020.

2010-08-24 15:11:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	  videomixer: Update disted ORC files

2010-08-23 15:44:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend.c:
	* gst/videomixer/blend_mmx.h:
	* gst/videomixer/blendorc.orc:
	* gst/videomixer/videomixer.c:
	  videomixer: Optimize ARGB blending and implement BGRA blending with orc
	  This now means, that we have absolutely no handwritten assembly anymore
	  in videomixer and it's also faster now when using SSE.

2010-08-22 01:58:05 -0700  David Schleef <ds@schleef.org>

	* gst/videomixer/blend.c:
	* gst/videomixer/blendorc.orc:
	  videomixer: Add orc implementation for blending
	  videomixer: Add orc implementation for blending

2010-08-22 01:54:16 -0700  David Schleef <ds@schleef.org>

	* gst/videomixer/videomixer.c:
	  videomixer: Fix example pipelines
	  videomixer: Fix example pipelines

2010-08-20 11:41:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/imagefreeze.c:
	  imagefreeze: Add test for checking if imagefreeze correctly returns UNEXPECTED after the first buffer

2010-08-20 11:38:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/imagefreeze.c:
	  imagefreeze: Add test for bufferalloc passthrough

2010-08-20 10:35:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/imagefreeze.c:
	  imagefreeze: Fix race conditions in the unit test
	  If setting the pipeline to PLAYING before issuing the seek, buffers
	  are already arriving at the sink before the seek is handled and
	  will have the wrong timestamps and everything.
	  Fixes bug #625547.

2010-08-20 10:34:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: Fix another subtle race condition related to starting the srcpad task
	  Due to a seek the srcpad task could be started in rare circumstances although
	  it shouldn't be started anymore because no upstream buffer is available.

2010-08-20 10:24:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: Protect the flushing-seek variable by the srcpad's stream lock
	  This fixes a subtle race condition, that caused bufferalloc to fail
	  with wrong-state due to a seek but caused it to be not retried as
	  it should.

2010-08-20 09:14:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Always generate a perfectly timestamped stream
	  Before there could be rounding errors when calculating the duration,
	  resulting in timestamp + duration being smaller than the next buffer's
	  timestamp.

2010-08-19 18:38:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Only include the server name in the context name if it's not NULL

2010-08-18 16:37:41 +0200  Philippe Normand <pnormand@igalia.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Add "client" property to set the PA client name
	  Allows the application to modify the client name used to connect when
	  connecting to the PulseAudio daemon. Note however that updating the
	  property after the element reached the READY state will have no
	  effect until the next NULL->READY transition.
	  Fixes bug #627174.

2010-08-19 17:59:09 +0200  David Hoyt <dhoyt@llnl.gov>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Improve error messages
	  Before they contained the URL before the actual failure. The other
	  way around makes more sense and we do the same in other elements
	  like filesrc.
	  Fixes bug #627289.

2010-08-19 12:46:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Free the clock on state change failures too

2010-08-17 16:26:41 +0200  Philippe Normand <pnormand@igalia.com>

	* configure.ac:
	* ext/pulse/pulseutil.c:
	* win32/common/config.h:
	  pulseutil: include pid value in gst_pulse_client_name() fallback return value
	  Fixes bug #627162

2010-08-19 12:32:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Free the GstPulseContext after usage

2010-08-16 09:12:04 +0200  Philippe Normand <pnormand@igalia.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: share the PA context between all clients with the same name
	  Avoid to create a new PA context for each new client by using a hash
	  table containing the list of ring-buffers and the shared PA context
	  for each client. Doing this will improve application memory usage in
	  the cases where multiple pipelines involving multiple pulsesink
	  elements are used.
	  Fixes bug #624338.

2010-08-17 13:41:49 +0200  Philippe Normand <phil@base-art.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: clear the PA mainloop if baseaudiosink failed to open the ring_buffer
	  If the application requests a state-change and pulsesink fails to open
	  the ring_buffer device the mainloop attribute of the sink should be
	  cleaned up to avoid future state-change (NULL->READY) failures.

2010-08-19 12:23:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Post an error message if EOS happens before valid input is found
	  Fixes bug #627341.

2010-08-12 11:49:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Send close newsegment event from the streaming thread

2010-08-11 11:36:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: Retry bufferalloc if it was aborted with WRONG_STATE because of a flushing seek

2010-08-11 08:46:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Return GST_FLOW_UNEXPECTED when getting a second buffer
	  This prevents upstream from pushing many useless buffers and makes
	  it go into EOS state.

2010-08-10 20:11:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Passthrough buffer allocations

2010-09-04 13:10:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development
	  Temporarily disable -DGST_DISABLE_DEPRECATED for git builds until
	  the code is updated for the GST_FLOW_IS_* macro deprecations.

=== release 0.10.25 ===

2010-09-02 23:44:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* gst-plugins-good.doap:
	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	* win32/common/config.h:
	  Release 0.10.25

2010-09-02 23:12:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update docs for release

2010-09-02 23:07:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/LINGUAS:
	* po/es.po:
	* po/gl.po:
	* po/lt.po:
	* po/nl.po:
	* po/ro.po:
	* po/sv.po:
	  po: update translations

2010-08-25 19:01:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  0.10.24.5 pre-release

2010-08-22 21:15:07 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: use separate buffer metadata for fields
	  Call gst_buffer_make_metadata_writable() on buffers that are
	  duplicated into fields.  Fixes #627689.

2010-08-21 21:41:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  0.10.24.4 pre-release

2010-08-19 18:30:05 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Prevent crash when reading image with problems
	  Check if we have data on the adapter and fail if not.
	  Fixes #627413

2010-08-13 17:24:01 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 3e8db1d to ec60217

2010-08-11 22:20:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Send close segments when seeking only for non-flushing seeks and if we already sent a newsegment event
	  Fixes bug #626619.

2010-08-11 16:50:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	* win32/common/gstrtpbin-marshal.c:
	* win32/common/gstudp-enumtypes.c:
	* win32/common/gstudp-enumtypes.h:
	* win32/common/gstudp-marshal.c:
	  0.10.24.3 pre-release

2010-08-11 11:17:18 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: prevent reading past avc1 atom when parsing
	  ... when one of the subatoms has a large/invalid size.
	  Fixes #626609.

2010-08-10 23:37:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  0.10.24.2 pre-release

2010-08-10 10:57:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From bd2054b to 3e8db1d

2010-08-09 00:36:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulse: fix printf format in some debugging messages

2010-08-08 23:31:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* pkgconfig/gstreamer-plugins-good-uninstalled.pc.in:
	  pkgconfig: set pluginsdir to top-level builddir without the pkgconfig/.. bits
	  Removes clutter in plugin dir paths. This is only used to find the -good
	  plugins for unit tests of ugly/bad/ffmpeg/etc. in an uninstalled setup.

2010-08-06 20:04:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: also log pixel formats in sorted order

2010-08-06 18:07:46 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: sort formats in the right order so that non-emulated formats are prefered
	  The format list should be sorted from high ranks to low ranks. In the GSList
	  sorting function this means the compare needs to return a positive value if
	  format a has a lower rank than format b.
	  Among other things this fixes v4l2src to prefer non-emulated formats
	  to emulated formats when built against libv4l.

2010-08-06 19:24:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Fix pipeline in the documentation
	  Make sure that we have the same color format on all streams, i.e. AYUV
	  Fixes bug #625452.

2010-08-05 13:56:44 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From a519571 to bd2054b

2010-06-14 19:58:11 +1000  Jonathan Matthew <jonathan@d14n.org>

	* ext/taglib/gstid3v2mux.cc:
	* tests/check/elements/id3v2mux.c:
	  id3v2mux: write beats-per-minute tag using TBPM frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=621520

2010-07-25 11:47:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Move debug categories into the source files and add debug category for the blend functions

2010-08-04 19:25:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Check if the compiler supports ISO C89 or C99 and which parameters are required
	  This first checks what is required for ISO C99 support and sets the relevant
	  compiler parameters and if no C99 compiler is found, it checks for a
	  C89 compiler. This enables us to check for and use C89/C99 functions
	  that gcc hides from us without the correct compiler parameters.

2010-07-15 10:10:31 +0200  Philippe Normand <pnormand@igalia.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: use G_TYPE_DEFINE to define ring buffer type
	  The existing get_type() implementation is racy, and the
	  g_type_class_ref() workaround didn't actually work because
	  it was in the wrong function. Since class creation in GObject
	  is thread-safe these days (since 2.16), the class_ref workaround
	  is no longer needed and it is sufficient to ensure the _get_type()
	  function is thread-safe, which G_TYPE_DEFINE does.
	  https://bugzilla.gnome.org/show_bug.cgi?id=624338

2010-08-04 15:20:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Post CLOCK-LOST/CLOCK-PROVIDE when going to/from READY
	  Otherwise the clocks are redistributed every time the pipeline
	  goes to PAUSED, which is quite expensive.

2010-07-12 12:35:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4gpay.h:
	  rtpmp4gpay: implement perfect timestamps
	  Use bitreader for parsing the config string
	  Reset state variables when going to READY
	  Parse frame length and use it to keep track of the rtptimestamps

2010-07-09 14:07:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263pdepay.c:
	  rtph263pdepay: allow more clock-rates as input
	  Although the spec says that the clock-rate should always be 90000, some rtsp
	  servers send different clock-rates so we must accept then in order to handle
	  those streams too.

2010-07-06 19:02:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpL16depay.c:
	  L16depay: default to 1 channel
	  When we can't find any channel or encoding-params on the caps for dynamic
	  payload types, set the default number of channels to 1, as the spec says we
	  should.
	  See #623209

2010-07-06 18:22:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't reuse udp sockets
	  Don't reuse sockets but make the udpsrc element fail the state change when the
	  socket is already in use. If we don't prevent reuse, we might end up using the same
	  port for different streams in some cases.
	  Fixes #622017

2010-07-06 18:11:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: add property to enable port reuse

2010-07-05 10:23:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpL16depay.c:
	  L16depay: use encoding-params for the channels
	  When parsing the number of channels, use the encoding-params property from the
	  RTP caps because that is where we can find the channels according to the spec.
	  Fall back to the channels property in the caps when needed.
	  Fixes #623209

2010-06-29 10:46:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: improve error and warning message
	  Improve error and warning message.
	  Fixes #622577

2010-08-02 23:15:56 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  examples: no need to set the color for each frq-band

2010-08-02 12:56:29 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpg729pay.h:
	  rtpg729pay: avoid basertppayload perfect-rtptime mode
	  G729 packets may only occur intermittently (e.g. cn packets), and as such
	  do not allow for perfect-rtptime calculating rtp times based on frame or byte
	  count.  In particular, do not use rtp audio base payloader as base class, but
	  rather base payloader directly.

2010-08-02 12:48:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: fix element leak

2010-08-02 12:46:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4vdepay.c:
	  rtpmp4vdepay: fix buffer leak

2010-08-02 12:46:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp payloading: fix pad leak

2010-07-29 17:18:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: push mode; use proper movi offset for movi based index
	  Fixes #623357.

2010-07-29 10:00:15 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: Correctly parse mvhd atoms
	  Parse mvhd data according to its version to avoid failing
	  on valid files.

2010-07-28 12:21:41 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix the max/avg in btrt atom reading
	  According to ISO media base format, the max bitrate is the
	  first one, and the avg comes next.

2010-07-27 15:58:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: proper handling of streaming upstream without duration
	  Fixes #625371.

2010-07-26 18:33:09 +0200  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: initialize some variables to fix compiler warnings on OSX build bot

2010-07-26 18:15:25 +0200  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: correctly check what version of gst-plugins-base we're compiling against
	  We need to check the gst-plugins-base version, not the core version
	  (even if both should be the same in any sane setup).

2010-07-26 17:45:42 +0200  Arnaud Vrac <rawoul at gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add port-range property to rtspsrc
	  To support setups with firewall/ipsec, it is useful for an rtsp client to be
	  able to set the range of ports that can be used for rtp/rtcp reception.
	  Allows this by adding a "port-range" property to the rtspsrc element.
	  Fixes #625153

2010-07-26 13:38:31 +0200  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: set the pixel-aspect-ratio field also for par=1/1
	  https://bugzilla.gnome.org/show_bug.cgi?id=625302

2010-07-26 15:31:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix memory leak in server request reply
	  The RTSP server rtspsrc is communicating with, sends a GET_PARAMETER request
	  periodically as a ping.  The code in gst_rtspsrc_handle_request forms an OK
	  response and sends, but doesn't call gst_rtsp_message_unset to free the memory
	  after sending the response.  This results in a constant slow memory leak.
	  Fixes #624770

2010-07-24 22:39:54 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/debugutils/cpureport.c:
	  cpureport: remove bogus docs

2010-07-24 22:37:11 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/debugutils/Makefile.am:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/cpureport.h:
	* gst/debugutils/gstdebug.c:
	  debugutils: new element cpureport
	  cpureport posts bus messages after every buffer received of cpu used, system
	  clock time, buffer time

2010-07-24 10:29:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/equalizer/demo.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  examples: Destroy the cairo context after usage

2010-07-24 10:21:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/Makefile.am:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	  Revert "gdkpixbuf: Add a gdkpixbuf3 plugin that uses gdkpixbuf3"
	  This reverts commit b6788153161b4e07fbf3d42a2d8921ea049305d0.
	  There's no gdk-pixbuf3 anymore. gdk-pixbuf was separated from GTK+
	  and will stay at version 2.0 for GTK+ 3.0.

2010-07-24 10:19:37 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/equalizer/demo.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  examples: Use cairo instead of to-be-deprecated GDK API
	  Fixes bug #625002.

2010-07-22 16:24:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix event leak

2010-07-22 12:05:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: pull mode non-cue seeking
	  That is, in files that have no index (Cue), perform seek by scanning for
	  nearest cluster with timecode before requested position.  Scanning is done
	  as a combination of interpolation and sequential scan.
	  Fixes #617368.

2010-07-16 12:46:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: streamable files need no _finish
	  Fixes #624455.

2010-07-22 11:46:35 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: push mode; handle 0-size data chunks
	  Fixes #618535.

2010-07-21 08:11:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Only reset QoS information and send a NEWSEGMENT event downstream for NEWSEGMENT events on the master pad

2010-07-14 20:31:44 -0700  David Schleef <ds@schleef.org>

	* gst/debugutils/Makefile.am:
	* gst/debugutils/gstcapsdebug.c:
	* gst/debugutils/gstcapsdebug.h:
	* gst/debugutils/gstdebug.c:
	  capsdebug: Add new element

2010-07-20 16:11:25 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: demote WARNING message to LOG level
	  It's not a warning.

2010-07-19 14:47:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Fix regression on markers parsing
	  Fixes a regression introduced when fixing bug #583047 in
	  commit a391bf52cc3c580c7a0a2316ca52eb66da3b85c1
	  Skip the data when libjpeg asks it to be skipped on
	  one of its callbacks.

2010-07-16 18:04:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: add missing argument in debug message

2010-07-16 17:53:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsemixerctrl.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulsesink: Only use gst_audio_clock_new() when compiling against newer base

2010-07-09 17:33:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/raw1394/gstdv1394src.c:
	  dv1394src: Post clock-provide and clock-lost messages when going from/to PLAYING
	  In PAUSED and below the clock is not working.

2010-07-04 16:57:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstswitchsink.c:
	* ext/gconf/gstswitchsink.h:
	* ext/gconf/gstswitchsrc.c:
	* ext/gconf/gstswitchsrc.h:
	  gconf: Fix ref handling of new child elements and minor cleanup

2010-07-04 09:45:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstgconfvideosrc.c:
	  gconfvideosrc: Use correct GConf key

2010-07-03 14:16:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstgconfaudiosrc.c:
	* ext/gconf/gstgconfaudiosrc.h:
	  gconf: Port gconfaudiosrc to GstSwitchSrc

2010-07-03 14:12:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstgconfvideosrc.c:
	* ext/gconf/gstgconfvideosrc.h:
	  gconf: Port gconfvideosrc to GstSwitchSrc

2010-07-03 14:11:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/Makefile.am:
	* ext/gconf/gstswitchsrc.c:
	* ext/gconf/gstswitchsrc.h:
	  gconf: Add GstSwitchSrc base class

2010-07-03 13:56:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstswitchsink.c:
	  gconf: Create the ghostpad of the switchsink from the template

2010-07-07 10:10:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Post clock-provide/clock-lost when going to/from PAUSED
	  Also use gst_audio_clock_new_full() to prevent crashes when the
	  clock is used after the element was destroyed.

2010-07-15 11:49:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: remove bogus UNLOCK

2010-07-13 12:34:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: also calculate PAR using track width and height for QT files
	  (... as opposed to only for ISO style files).
	  Fixes #624173.

2010-07-12 17:29:12 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: handle bogus files storing ADTS AAC data

2010-07-09 16:57:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: do not error out on a block with unknown tracknumber

2010-07-08 18:57:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: do not align reverse playback reference stream twice
	  Timestamp rounding issues could lead to going backwards 2 keyframe periods
	  (rather than only 1).  While this is not necessarily a problem, it might
	  potentially place additional (buffering) load on downstream and could be
	  avoided (because We Can).
	  Fixes #623629.

2010-07-08 16:07:16 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: convert some more mov format timestamp to gst time

2010-07-07 14:16:59 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: additional verification heuristics for VBR audio stream
	  Check for and override some header field(s) for reasonable values, according
	  to later expected use in calculations.

2010-07-14 15:21:21 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Fix wrong lock order that could lead to a deadlock. Fixes #624331.

2010-07-16 11:31:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development

=== release 0.10.24 ===

2010-07-15 01:49:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.24

2010-07-15 01:35:06 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/cs.po:
	* po/lv.po:
	  po: update translations

2010-07-07 00:42:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  0.10.23.4 pre-release

2010-07-07 00:31:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/LINGUAS:
	* po/da.po:
	* po/el.po:
	* po/es.po:
	* po/fr.po:
	* po/id.po:
	* po/pt_BR.po:
	* po/sl.po:
	* po/tr.po:
	* po/zh_CN.po:
	  po: update translations

2010-06-23 11:47:43 +0200  Michael Grzeschik <m.grzeschik@pengutronix.de>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: destroy buffer pool when changing state to NULL
	  In the case we change the State from READY_TO_NULL the buffers in the pool
	  still hold an open dup file descriptor to the device, therefore the device
	  release function will not be called and the device will probably answer with
	  -EBUSY when we reopen it in the next NULL_TO_READY transition.
	  Signed-off-by: Michael Grzeschik <m.grzeschik@pengutronix.de>
	  See bug #622500 and #612244.

2010-07-06 13:21:19 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix order of bitrates in 'btrt' atom
	  There seems to be a bug in libmp4v2 that generates a MPEG4BitRateBox as
	  (bufferSizeDB, avgBitrate, maxBitrate) instead of (bufferSizeDB,
	  maxBitrate, avgBitrate), according to the spec. I used the mp4file
	  output while writing this code, so the order is wrong. This patches
	  fixes that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=623654

2010-07-05 12:05:57 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix skipping extra 0xff markers
	  Fixes #623585.

2010-06-29 23:18:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: fix memory leak
	  Don't leak result of gst_adapter_take(). There are most likely
	  smarter things we can do, but let's keep things simple for the
	  release.
	  Fixes #623172.

2010-07-02 12:31:31 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: strip out bogus tags from XMP atom
	  https://bugzilla.gnome.org/show_bug.cgi?id=623366

2010-07-02 14:25:22 +0200  Andrzej K. Haczewski <ahaczewski@gmail.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Write duration at the correct position

2010-06-30 11:12:08 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: fix memleak on custom downstream events
	  by not sending custom downstream event twice and fix memleak when
	  not handling the event
	  https://bugzilla.gnome.org/show_bug.cgi?id=623196

2010-06-29 20:18:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  0.10.23.3 pre-release

2010-06-29 20:14:53 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: fix unportable printf format specifiers in commented out code
	  To avoid false positives when grepping for unportable specifiers.

2010-06-29 19:12:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: fix --disable-external

2010-06-28 15:44:06 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* autogen.sh:
	* configure.ac:
	  Bump automake requirement to 1.10 and autoconf to 2.60
	  For maintainability reasons and $(builddir).
	  See #622944.

2010-06-28 09:07:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/goom/plugin_info.c:
	  goom: don't allocate 260kB struct on the stack
	  PluginInfo is quite a sizeable struct, let's not allocate it on the
	  stack, especially not if we're copying it over into another dynamically
	  allocated copy anyway.
	  Fixes #570761.

2010-06-27 10:31:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Require GTK+ >= 2.14 for the examples

2010-06-26 20:12:25 +0200  Guido Günther <agx@sigxcpu.org>

	* tests/examples/equalizer/demo.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  examples: Make demos -DSEAL safe to fix build with GTK+ 3.0

2010-06-26 21:39:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/jpeg/Makefile.am:
	  jpeg: Explicitely link with libgstbase

2010-06-26 18:42:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.23.2 pre-release

2010-06-26 18:41:49 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videomixer/blendorc-dist.c:
	  gst: update orc files

2010-06-26 18:41:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update translations

2010-06-25 19:40:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Fix leaking of the streamheader buffers
	  gst_value_set_buffer() increases the refcount and doesn't
	  take ownership of the buffer.

2010-06-24 16:32:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstvideoflip.c:
	  matroska, videobox, videofilter: fix compiler warnings when debugging is disabled in gstreamer
	  Fixes unused variable warnings when GStreamer's debugging system has been disabled.

2010-06-24 15:17:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	  tests: add plugin loading whitelist to test environment
	  Only want to load core/base/good plugins here.
	  Fixes #619717.

2010-06-24 15:09:16 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 73ff93a to a519571

2010-06-24 13:02:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	  gdkpixbufdec: bump rank to SECONDARY
	  Bump gdkpixbufdec's rank to SECONDARY to give it an edge over misc.
	  image decoders in gst-ffmpeg that also have a MARGINAL rank.
	  Fixes #620162.

2010-06-23 12:15:13 +0200  Michael Grzeschik <m.grzeschik@pengutronix.de>

	* gst/avi/gstavidemux.c:
	  reset the have_index flag at transition PAUSED_TO_READY
	  If we restart the Stream in the case of doing a transition from
	  PAUSED_TO_READY and back with READY_TO_PAUSED aso. the duration of the video
	  will get calculated even if we have a avi header with that information.
	  Signed-off-by: Michael Grzeschik <m.grzeschik@pengutronix.de>

2010-06-23 20:29:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix negotiation for I420/YV12
	  We don't support conversion into *all* YUV
	  formats for them, only into I420/YV12/AYUV.
	  Fixes bug #622501.

2010-06-22 15:22:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: proper closing segment construction
	  Fixes #618982.

2010-06-22 15:46:51 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2: precalculate duration
	  Have frame duration in the instance struct and calculate it after changing the caps.

2010-06-21 12:17:39 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: use glib defines in property declarations for readability

2010-06-21 12:15:14 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: use G_PARAM_STATIC_STRINGS to save a few bytes and strdups

2010-06-18 20:02:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix locking after moving things around

2010-06-18 14:13:58 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/taglib/gstapev2mux.cc:
	  taglib: Use newly added gst_tag_list_peek_string_index
	  Replace calls to gst_tag_list_get_string_index with
	  gst_tag_list_peek_string_index to avoid a string copy

2010-06-18 16:56:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: make some errors as warnings
	  Avoid spamming the testsuite with these error debug lines.

2010-06-18 16:49:08 +0200  Keith Nicholson <keith.nicholson at ultra-ccs.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix multicast support on windows builds
	  On windows builds, sets source address for bind to INADDR_ANY, while
	  maintaining the original multicast group address for subsequent join.
	  Fixes #595978

2010-06-18 16:16:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpnetutils.c:
	  udp: make url parsing compatible with VLC syntax
	  Skip everything before the @ sign in the url location. VLC uses that as the
	  remote address to connect to (but we ignore it for now). This makes our udp urls
	  compatible with the ones used by VLC.
	  Fixes #597695

2010-06-18 15:08:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: factor out the connections
	  Keep a global connection for aggregate control but also keep stream connections
	  for non-aggregate control.
	  Add some helper methods to connect/close/flush the connections.

2010-06-17 13:06:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add non-aggregate control
	  Add non-aggregate control.
	  Separate retrieving thr SDP from parsing and setting up the streaming from the
	  SDP.

2010-06-17 22:10:03 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* common:
	  common: update common back to what it was

2010-06-17 17:24:22 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* common:
	* gst/flv/gstflvmux.c:
	  flvmux: add documentation for streamable property

2010-06-17 16:43:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  docs: update introspected plugin docs for gstdoc-scangobj and other changes
	  Update common for latest gstdoc-scangobj, and inspect xml files for
	  escaping and pad template order changes.

2010-06-17 16:41:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/.gitignore:
	  tests: ignore sub-directory with orc tests

2010-06-17 10:44:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix an uninitialized variable compiler warning

2010-06-16 21:02:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	  ebml-read: Zero-sized ints/uints/floats have a value of 0 according to the EBML spec

2010-06-16 20:02:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix possible NULL pointer dereference and assertion that could be caused by invalid files

2010-06-16 19:50:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Clean up/fix some minor error handling bugs

2010-06-16 19:30:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Fix NULL pointer dereference when allocation of the ximage fails

2010-06-16 19:28:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflactag.c:
	  flactag: Fix possible NULL pointer dereference

2010-06-16 19:24:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioiirfilter.c:
	  audioiirfilter: Fix possible NULL pointer dereference

2010-06-16 19:20:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	  warptv: Don't use floats as loop counters

2010-06-16 11:21:35 -0400  Havoc Pennington <hp@pobox.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: do not try to change device format if it's already correct
	  This allows set_caps to succeed if caps change in a way that
	  would not modify the format we're getting from the hardware.
	  Otherwise if not in NULL state, setting caps would fail
	  with EBUSY.
	  With this change, in some cases it's OK to go PLAYING->READY->PLAYING
	  rather than PLAYING->NULL->PLAYING to avoid a time-consuming close
	  and reopen of the device.
	  Fixes #621723

2010-06-16 11:09:17 -0400  Havoc Pennington <hp@pobox.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: in negotiate, check for error return from set_caps
	  Fixes #621723  (partially)
	  set_caps can fail if the video device is running, in that case
	  setting its format leads to EBUSY.
	  If set_caps fails then we will not have set up the buffer pool
	  (it will be NULL) which leads to a crash when we try to pull
	  buffers. If we fail the negotiate on set_caps failure, then we
	  won't go to playing state and won't crash.
	  This is a small improvement. Of course, a nicer fix would
	  be to make set_caps work in the case where the format is
	  unchanged. If the format has changed, failing is
	  probably correct because we need to close the device
	  (go to NULL state) in order to set caps.

2010-06-16 15:40:34 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: improve audio vbr detection
	  Subsequent entry time calculations use blockalign value to determine
	  number of frames per chunk, and blockalign == 1 is then most unlikely to result
	  in reasonable values (which also aligns with "spec").

2010-06-16 15:52:57 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: tweak DELTA_UNIT labeling
	  Consider SPS, PPS and IDR as keyframe, all others as DELTA_UNIT.
	  See #620154.

2010-06-15 20:06:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: Initialize uninitialized variable and don't unref it if it's NULL

2010-06-15 20:04:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Assign variables before printing them

2010-06-15 20:00:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Initialize uninitialized variable

2010-06-15 19:47:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Initialize variable

2010-06-15 19:45:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: Fix NEWSEGMENT parsing logic and don't use uninitialized variables

2010-06-15 17:20:20 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/ebml-read.c:
	  matroska: Fix unitialized variable

2010-06-15 16:49:49 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From 9339ccc to 35617c2

2010-06-15 16:54:04 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 5adb1ca to 9339ccc

2010-06-15 16:35:18 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 57c89b7 to 5adb1ca

2010-06-15 14:08:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* .gitignore:
	  .gitignore: ignore generated tvtime.h file

2010-06-15 15:36:33 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From c804988 to 57c89b7

2010-05-17 13:54:03 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* ext/raw1394/gst1394clock.c:
	* ext/raw1394/gst1394clock.h:
	  raw1394: remove useless last_time
	  It seems to me this code is useless: removing it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=618871

2010-06-14 19:21:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: respect aggregate control attributes
	  when the SDP specifies an aggregate control url, use that for playback
	  control.
	  Fixes #619531

2010-06-14 15:36:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: Call orc_init() before trying to get target flags

2010-06-14 15:35:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Call orc_init() before trying to get target flags

2010-06-14 14:26:22 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-mux.c:
	* tests/check/elements/matroskamux.c:
	  matroskamux: revert change that set a reserved flag on the Block.
	  So matroska's Block structure has no keyframe flag, only the SimpleBlock has it.
	  To detect keyframes in Blocks, it is just the BlockGroup container that needs
	  to have a ReferenceBlock attached if it is a delta frame in video.

2010-05-31 12:45:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: use libjpeg scatter-gather operation to avoid data copying
	  Fixes #583047 (more).

2010-05-27 15:45:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: optimize buffer handling when parsing
	  Use an adapter to collect incoming data, and use adapter API to scan and peek.
	  Fixes #583047.

2010-06-14 13:48:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/oss4/oss4-mixer.c:
	  oss4: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp()

2010-06-14 13:27:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Use GLIB_EXTRA_CFLAGS

2010-06-14 13:03:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 7a0fdf5 to c804988

2010-06-14 11:46:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: also consider AU and SEI NALUs as DELTA_UNIT
	  Fixes #620154.

2010-06-14 11:32:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 6da3bab to 7a0fdf5

2010-06-12 21:26:16 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtp/gstrtpmparobustdepay.c:
	  build: include stdio.h for sscanf

2010-06-12 14:12:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/Makefile.am:
	  tests: Add clean rule for the orc tests

2010-06-12 14:12:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/Makefile.am:
	  tests: Add autogenerated orc tests

2010-06-12 08:27:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 733fca9 to 6da3bab

2010-06-11 16:23:29 -0700  David Schleef <ds@schleef.org>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Fix element description

2010-06-11 21:13:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpmparobustdepay.c:
	  rtpmparobustdepay: don't try to unref NULL buffers
	  Fixes generic/states unit test.

2010-06-11 20:50:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: use typefind functions to check if PCM data contains dts stream
	  Use new dts audio typefinder from -base to check if the PCM data
	  contains a dts stream. This way we recognise more varieties more
	  reliably and also detect the dts stream if there isn't a frame
	  sync right at the start of the data.
	  Fixes #413942.

2010-06-11 20:47:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: set buffer offsets before using the buffer for the first time
	  gst_type_find_helper_for_buffer() will need the correct offset
	  set on the buffer (ie. 0) and not the byte offset we started
	  pulling the data from.

2010-06-10 16:14:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmparobustdepay.h:
	  rtp: add mpa-robust depayloader
	  Fixes #589997.

2010-06-11 10:57:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: fix avi header bytewriting
	  ... by using proper offsets for tag list writing.
	  Also use _reset rather than _free and consistently use bytewriter position.
	  See #619293.

2010-06-10 22:58:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* .gitignore:
	  Update .gitignore
	  Add the generated orc source files

2010-06-10 22:55:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/matroskamux.c:
	  matroskamux: Fix unit test for changed key-frame behaviour
	  All audio frames are marked as keyframe now instead of marking
	  them all as delta unit...

2010-06-10 22:45:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend.c:
	* gst/videomixer/blend_mmx.h:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	* gst/videomixer/blendorc.orc:
	  videomixer: Port most blending related functions to orc
	  Only remaining MMX implementation is the ARGB/BGRA/AYUV blending
	  for which we first need the orc compositing opcodes.

2010-06-10 20:17:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_mmx.h:
	  videomixer: Replace some tabs by spaces

2010-06-10 11:04:38 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* ext/raw1394/gst1394clock.c:
	  dv1394: Fix the internal clock even more
	  The cycleCount register is 13 bits long and the cycleOffset one
	  is 12 bits long. To read the cycleCount register we need to shift
	  12 bits and not 13. Fixes #615461

2010-06-09 18:37:29 -0700  David Schleef <ds@schleef.org>

	* configure.ac:
	  configure: use m4 macro to check for Orc

2010-06-09 22:40:23 +0200  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-mux.c:
	  matroskamux: some non-delta buffers were not marked as keyframes

2010-06-09 22:00:16 +0200  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: change 2 second limit per cluster
	  Start cluster at every keyframe or when we would overflow the previous
	  cluster's relative timestamp field. This would avoid as much as possible
	  starting clusters at non-keyframes.

2010-06-09 12:40:09 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From fad145b to 733fca9

2010-06-09 12:34:01 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From 47683c1 to fad145b

2010-06-09 20:53:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Don't request more shared memory than needed

2010-06-09 20:45:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstswitchsink.c:
	  switchsink: Set the GST_ELEMENT_IS_SINK flag on the sink

2010-06-09 20:43:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstgconfvideosink.c:
	* ext/gconf/gstgconfvideosink.h:
	  gconfvideosink: Use GstSwitchSink as base class

2010-06-09 20:30:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstgconfaudiosink.c:
	  gconfaudiosink: Use G_PARAM_STATIC_STRINGS

2010-06-09 20:29:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/gstgconfaudiosink.c:
	* ext/gconf/gstgconfaudiosink.h:
	  gconfaudiosink: Rename instance variable to be more descriptive

2010-06-09 20:22:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautovideosink.c:
	  auto{audio,video}sink: Don't lose the GST_ELEMENT_IS_SINK flag after removing the child

2010-06-09 20:07:09 +0200  Julien Moutte <julien@fluendo.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Plug some memleak and support 22050Hz mono sound.
	  Segment size needs to be a multiple of the sample size in bytes.

2010-06-09 16:22:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Flush shm buffer immediately if it's full

2010-06-09 16:21:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Fix writing of buffers larger than segsize
	  Fixes bug #620540.

2010-06-09 15:42:37 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Fix playback if PA doesn't give us a large enough shared memory buffer

2010-06-09 15:42:19 +0200  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: change indexed property to streamable
	  The property streamable has reverse semantics to indexed.

2010-06-09 09:13:09 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Rename unreleased property 'indexed' to 'streamable'
	  Rename 'indexed' to 'streamable' for a better name while it
	  hasn't been released

2010-06-08 15:23:51 -0700  David Schleef <ds@schleef.org>

	* REQUIREMENTS:
	* configure.ac:
	  configure: remove liboil check

2010-06-08 14:44:19 -0700  David Schleef <ds@schleef.org>

	* gst/level/gstlevel.c:
	  level: remove unused liboil include

2010-06-04 18:22:42 -0700  David Schleef <ds@schleef.org>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend.c:
	  videomixer: liboil to orc conversion

2010-06-04 18:21:21 -0700  David Schleef <ds@schleef.org>

	* gst/videobox/Makefile.am:
	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videobox/gstvideoboxorc.orc:
	  videobox: liboil to orc conversion

2010-06-04 18:16:25 -0700  David Schleef <ds@schleef.org>

	* gst/goom/Makefile.am:
	* gst/goom/README:
	* gst/goom/gstgoom.c:
	* gst/goom/plugin_info.c:
	  goom: liboil to orc conversion

2010-06-08 16:04:23 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/Makefile.am:
	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/deinterlace/tvtime.orc:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/vfir.c:
	  deinterlace: orcify some deinterlacing methods

2010-06-08 16:03:36 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/Makefile.am:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: convert from liboil to orc

2010-06-08 15:23:28 -0700  David Schleef <ds@schleef.org>

	* REQUIREMENTS:
	* configure.ac:
	  configure: Add orc check

2010-06-08 14:09:00 +0200  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Add indexed property to replace disabled is-live.
	  Add indexed property to be the negation of what the disabled is-live property
	  was. Fixes bug #613066.

2010-06-08 09:22:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  raw1394: Require libraw1394 >= 2.0.0 for raw1394_read_cycle_timer
	  Fixes bug #620929.

2010-06-08 07:35:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/annodex/gstcmmlenc.c:
	  cmmlenc: Remove hack to let oggmux start a new page for every CMML buffer
	  oggmux does this for CMML by its own now

2010-06-07 18:32:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't handle non-TIME seeks
	  Don't send them upstream because for upstream a BYTES seek
	  might make sense but is completely wrong because upstream
	  can't seek to a byte position of the audio or video stream.
	  Also don't build the index in push mode for non-TIME seeks,
	  things will go wrong here otherwise.

2010-06-07 11:15:26 -0400  Olivier Crête <tester@tester.ca>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfdetect.h:
	  dtmfdetect: Only works with rate=8000, fix in caps

2010-06-02 19:16:20 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  Cope with short startcodes in the h264 bytestream

2010-06-06 17:25:16 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulse: log message printf format fixes

2010-06-06 18:00:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* ext/pulse/pulsemixer.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/speex/gstspeexenc.c:
	* ext/taglib/gsttaglibmux.c:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackparse.c:
	  ext: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs

2010-06-06 17:57:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstossdmabuffer.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxringbuffer.c:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/waveform/gstwaveformsink.c:
	  sys: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs

2010-06-06 17:52:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	* gst/flv/gstflvdemux.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videomixer/videomixer.c:
	  gst: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs

2010-06-06 15:12:16 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: refactor delta unit handling
	  This allows us to skip delta units earlier and is a bit clearer in my
	  opinion. It also makes only video buffers ever be delta units, not
	  just for SimpleBlock as before.

2010-06-06 15:17:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Clear adapter on discontinuities

2010-06-06 14:03:53 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Ignore keyframe flag for non-video streams
	  When the keyframe bit of SimpleBlock Flags wasn't set, the buffer was being
	  marked with GST_BUFFER_FLAG_DELTA_UNIT, causing all buffers to be skipped
	  after a seek. This may be a problem with the Sorenson Squish encoder, but
	  arguably the keyframe bit should only be applied to video.
	  Fixes bug #620358.

2010-06-06 14:56:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: First try upstream when handling seek events/queries

2010-06-04 14:54:59 -0400  Tristan Matthews <tristan@sat.qc.ca>

	* gst/rtp/gstrtpceltpay.c:
	  gstrtpceltpay: don't always fixate sink caps to 1 channel
	  The getcaps function should not fixate the channels field until we
	  get the encoding-params field from our srcpad's caps. Fixes #620591

2010-06-04 13:57:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: try all ranges from the sdp
	  Try all ranges in the SDP before giving up.

2010-06-04 13:56:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: make parse_range return result
	  Make the parse_range function return if the parsing succeeded or failed.

2010-06-04 11:44:09 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/videomixer.c:
	  videomixer: if we're not linked downstream, we can do any format
	  Stupid me, assuming _get_allowed_caps() would actually return the
	  pad templates if there was no peer.

2010-05-31 16:26:19 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  Keep announcing the delivery-method in the capabilities
	  Even though we don't use delivery-method in our payloader, older versions of
	  the theora payloader in gstreamer required it. As such we need to keep this
	  around in the caps for backwards-compatibility.
	  This reverts part of 49463a37cbaa952e1401291f0a2623de6cab3880
	  Fixes #618940

2010-06-03 17:52:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	* sys/oss4/oss4-mixer.c:
	  oss4: add some comments for translators to clarify meaning of "Low"
	  "Low" etc. are quality settings here (e.g. for the internal resampler).
	  Some day when we use GLib's i18n functions we might want to use
	  NC_() and g_dpgettext2() here instead of the comments.
	  Fixes #555967.

2010-06-03 19:23:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gdepay.h:
	  mp4gdepay: calculate the frame duration correctly
	  When we calculate the frame duration, we need to use the amount of
	  frames in the _previous_ packet, not the current packet. The frame duration is
	  needed to correctly de-interleave interleaved streams. This fixes the case where
	  there are a variable number of frames in a packet.
	  Fixes #620494

2010-06-03 18:58:42 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/videomixer.c:
	  videomixer: Don't return caps in get_caps() that will be rejected
	  This commit basically puts _get_caps() in sync with accept_caps().
	  If we don't have a master pad OR the master pad caps aren't negotiated
	  then we just return the downstream allowed caps.
	  If we have a master pad with negotiated caps, we return those caps
	  with a free range of width/height/framerate

2010-06-03 13:45:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  Revert "pulsesink: Add comments to remove the provide-clock message posting once we depend on base 0.10.30"
	  This reverts commit 8f3708f38aa3839a6a625ca7d1c166101c9fbb7f.
	  The baseaudiosink commit was reverted

2010-06-03 10:27:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Add comments to remove the provide-clock message posting once we depend on base 0.10.30
	  baseaudiosink does all this for us now.

2010-05-07 18:42:06 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: Remove rtpdtmfmux stream-lock code

2010-06-02 16:36:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: delayed seek handling also deserves TRUE event response

2010-06-02 15:30:47 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix compiler warning
	  unused variable ‘estimated’

2010-06-02 15:04:00 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* common:
	  common: revert the change i did in my previous commit

2010-06-02 13:39:10 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* common:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: stop buffering and emit EOS at the end of a stream
	  When using RTP_JITTER_BUFFER_MODE_BUFFER, make sure that the ringbuffer doesn't
	  get stuck buffering forever when there isn't enough data left to fill the
	  buffer.

2010-06-01 21:52:59 +0200  Benjamin Otte <otte@redhat.com>

	* gst/debugutils/testplugin.c:
	  debugutils: Don't consume preroll buffer twice

2010-06-01 21:32:11 +0200  Benjamin Otte <otte@redhat.com>

	* ext/pulse/pulseutil.c:
	  pulse: Style fix: use g_strdup() instead of printf()ing a simple string

2010-05-27 16:07:31 +0200  Benjamin Otte <otte@redhat.com>

	* gst/debugutils/tests.c:
	  debugutils: Replace md5 implementation with glib's
	  https://bugzilla.gnome.org/show_bug.cgi?id=619824

2010-05-22 11:55:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: clean up code for avi header using a bytewriter
	  https://bugzilla.gnome.org/show_bug.cgi?id=619293

2010-06-01 18:54:41 -0500  Pierre-Louis Bossart <pierre-louis.bossart@intel.com>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	  pulsesink: optimize communication with PulseAudio using pa_stream_begin_write

2010-06-02 10:52:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Post provide-clock message on the bus if the clock appears/disappears
	  Fixes bug #620277.

2010-06-01 23:49:17 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From 17f89e5 to 47683c1

2010-06-01 22:54:49 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From cdff0fb to 17f89e5

2010-06-01 20:45:29 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/videomixer.c:
	  videomixer: filter caps returned from downstream with our pad template.

2010-06-01 16:56:32 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Remove more unneeded warnings

2010-06-01 16:54:03 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/ebml-write.c:
	  matroskamux: remove unneeded warning

2010-06-01 16:49:14 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/ebml-write.c:
	  matroskamux: remove unneeded debug statement

2010-06-01 16:24:53 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: change is-live property to indexed

2010-05-23 13:56:16 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: use the uint64 scaling functions
	  In demuxer and muxer use the gst_util_uint64 scaling functions rather than
	  standard integer division. Add warnings (to be changed to debug) for debugging
	  the timestamp and duration.

2010-05-21 14:35:34 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: set delta unit on all buffers except cluster start ones

2010-05-21 13:38:11 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: store caps and set on buffers rather than using pad caps

2010-05-21 13:25:24 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-mux.c:
	  matroskamux: make sure pads caps are set before any buffers pushed.

2010-05-21 13:14:04 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: add streamheaders

2010-05-21 12:23:08 +0100  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* gst/matroska/matroska-mux.c:
	  matroskamux: no need to set cache twice

2010-05-21 01:59:53 +0200  Xavier Queralt <xqueralt@gmail.com>

	* gst/matroska/matroska-mux.c:
	  Do not create a SeekHeader, Cues, .. when doing live

2010-05-20 23:39:59 +0200  Xavier Queralt <xqueralt@gmail.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  Add is-live property

2010-06-01 13:22:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix variable init

2010-05-28 16:37:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-ids.h:
	  matroskademux: improve reverse playback
	  Slightly modify approach to also handle cases where cue entries do not reliably
	  lead to initial keyframes.
	  Fixes #619817.

2010-05-24 16:02:58 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: avoid gtk-doc confusing comments

2010-05-21 11:21:58 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/matroskamux.c:
	  matroskamux: adjust unit test to modified behaviour

2010-05-20 14:33:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: use write caching also when writing buffer data
	  Specifically, this reduces pushing several small buffers for each
	  data buffer and also avoids a seek for each buffer altogether
	  (though a seek is still needed for each cluster).
	  Fixes #619273.

2010-05-20 14:23:07 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: fix ebml write caching with bytewriter implementation
	  Also cache a bit more during header writing.
	  Fixes #619273.

2010-05-20 14:08:42 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroskamux: use consistent debug category name for ebmlwrite

2010-05-18 14:44:15 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-read.h:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: use bytereader based GstEbmlRead as a helper
	  ... rather than basing on it by inheritance.
	  Also use more common code for push and pull mode.
	  Fixes #619198.
	  Fixes #611117.

2010-06-01 15:47:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: _get_pad_template result needs no unref

2010-05-18 19:42:58 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/libpng/gstpngenc.c:
	  pngenc: Support 8 bit grayscale
	  Adds support to 8 bit grayscale input

2010-05-18 14:46:54 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Adds 8bit grayscale support
	  Adds decoding support for jpeg images in 8 bit grayscale format.

2010-05-18 01:57:14 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Accept grayscale as input
	  Adds video/x-raw-grayscale (8 bit) support to jpegenc

2010-05-31 13:30:05 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/videomixer.c:
	  videomixer: Implement sinkpad GetCapsFunction.
	  This allows returning only the formats, width, height, framerate
	  and pixel-aspect-ratio that downstream can support.
	  https://bugzilla.gnome.org/show_bug.cgi?id=620148

2010-05-20 11:28:47 -0400  Tristan Matthews <tristan@sat.qc.ca>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: implement latency query
	  The encoder's latency is deduced from the framesize. Fixes #618896.

2010-05-31 07:49:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't compare running times with stream times when doing QoS

2010-05-27 21:06:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Don't reconfigure the caps when changing properties
	  Fixes bug #619848.

2010-05-26 13:13:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: Add property to allow passthrough mode
	  This passthrough mode is used if the alpha method is "set"
	  and the alpha value is 1.0.
	  Fixes bug #617512.

2010-05-25 15:16:06 +1000  Alexander Kojevnikov <alexander@kojevnikov.com>

	* gst/spectrum/gstspectrum.c:
	  spectrum: support 24-bit width
	  Fixes #619045

2010-05-24 21:50:58 +1000  Alexander Kojevnikov <alexander@kojevnikov.com>

	* gst/spectrum/gstspectrum.c:
	  spectrum: support arbitrary bit depth
	  Partially fixes #619045

2010-05-25 05:36:46 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix deadlock introduced by video keyframe QoS

2010-05-23 09:32:08 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: skip buffers before a late keyframe (QoS)
	  Before, vp8dec had no option but to decode all frames even if some/all
	  of them would be late. With this change, performance when keyframes are
	  frequent is helped a great deal. On my Thinkpad X60s, decoding a 20 s
	  1080p sunflower encode with keyframes every 10 frames went from taking
	  42 s with 5 frames shown to 21 s with 15 frames shown (still slow
	  enough to count by hand). When keyframes are more sparse, you will
	  still be able to catch up eventually, but the results won't be as
	  noticable.

2010-05-14 17:57:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	* gst/videomixer/videomixerpad.h:
	  videomixer: Don't mix input with different pixel aspect ratios
	  Fixes bug #618530.

2010-05-17 19:54:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/greedyh.asm:
	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: Add MMX/3DNow implementations of greedyh for UYVY

2010-05-17 19:16:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: Fix UYVY implementation of greedyh to be actually used

2010-05-11 11:43:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/Makefile.am:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	  gdkpixbuf: Add a gdkpixbuf3 plugin that uses gdkpixbuf3

2010-06-01 10:06:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	* common:
	* win32/common/gstrtpbin-marshal.c:
	* win32/common/gstrtpbin-marshal.h:
	* win32/common/gstudp-enumtypes.c:
	* win32/common/gstudp-marshal.c:
	* win32/common/gstudp-marshal.h:
	  win32: add more generated marshal and enumtype files to win32-update

2010-06-01 09:27:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska.c:
	  Revert "matroska: add temporary webm typefinder"
	  This reverts commit d148ec0ad2053abb0c38fc681a8953292985388f.
	  We depend on -base git now, which has a webm typefinder in the usual
	  place.

2010-06-01 09:26:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/flv/gstflvmux.c:
	* gst/matroska/matroska-mux.c:
	  Revert "avimux, flvmux, matroskamux: don't crash if tags arrive on multiple input pads at the same time"
	  This reverts commit 6a9983cd20c48b96396229b3f94d0254a05ddf48.
	  Rely on locking done in GstTagSetter in core git.

2010-06-01 09:23:18 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: require core/base git
	  For WebM typefinding and GstTagsetter fixes.

2010-06-01 09:17:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development

=== release 0.10.23 ===

2010-05-30 14:03:53 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.23

2010-05-30 14:02:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2010-05-29 10:23:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Fix position query

2010-05-28 15:14:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/webm-mux.c:
	  docs: remove unnecessary videorate element from webmmux example pipeline

2010-05-28 10:43:36 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Keep variables in sane state after _reset
	  When reseting, keep 'row' variables at a sane state after
	  freeing to avoid it being freed again on _resync realloc
	  when the element is reused.
	  Fixes #619943

2010-05-27 18:08:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix floating point to integer conversion for the alpha values
	  Fixes bug #619835.

2010-05-26 08:54:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.22.3 pre-release

2010-05-26 00:33:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update translations

2010-05-25 15:34:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: handle truncated input data at EOS in pull mode
	  Fixes #617733.

2010-05-26 11:55:13 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 357b0db to fd7ca04

2010-05-25 21:14:05 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Round timestamp up when scaling to mov format
	  Fix timestamp rounding to allow the correct index to be located.
	  The issue was that scaling from GStreamer time format to mov time format was
	  rounding down causing the timestamp of the newsegment event received after a
	  flushing keyframe seek to find the sample index before the one it should
	  causing further backward seeking to the keyframe prior until no rounding error
	  occurred.
	  Rounding up when scaling to mov format has the desired effect, and it is
	  not clear whether just the _round () variant would be sufficient.
	  Fixes bug #619105

2010-05-24 17:26:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/flv/gstflvmux.c:
	* gst/matroska/matroska-mux.c:
	  avimux, flvmux, matroskamux: don't crash if tags arrive on multiple input pads at the same time
	  This is a temporary fix for the release only.
	  Fixes #619533.

2010-05-25 17:05:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	  rtptheora: remove delivery-method from caps
	  We can accept all delivery methods so don't advertise anything on the caps or
	  parse anything, we will handle whatever we receive.
	  Fixes #618940

2010-05-25 15:40:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska.c:
	  matroska: add temporary webm typefinder
	  Add webm typefinder just for the release, so webm works for
	  people whose distros don't patch gst-plugins-base as well.
	  We'll remove this again after the release.

2010-05-23 11:17:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/webm-mux.c:
	  docs: add some pipeline examples to webmmux docs

2010-05-21 12:27:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: add webmmux to docs

2010-05-21 13:01:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/inspect/plugin-matroska.xml:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska.c:
	* gst/matroska/webm-mux.c:
	  matroska: fix up plugin and element descriptions a bit

2010-05-21 12:47:03 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/Makefile.am:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	* gst/matroska/matroska.c:
	* gst/matroska/webm-mux.c:
	* gst/matroska/webm-mux.h:
	  matroska: move webmmux into own source files
	  Makes things easier for gtk-doc.

2010-05-21 12:26:05 +0500  Christian Schaller <christian.schaller@collabora.co.uk>

	* gst-plugins-good.spec.in:
	  Update spec file with latest changes

2010-05-20 20:01:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroska: Remove the doctype enum, it's not needed anymore

2010-05-20 19:57:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  webmmux: Add new webmmux element that only supports muxing of WebM
	  ...and remove the doctype property from matroskamux again.

2010-05-20 17:31:59 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/matroskamux.c:
	  matroskamux: unit test checks version 1 files

2010-05-18 15:27:06 -0400  Tristan Matthews <tristan@sat.qc.ca>

	* ext/speex/gstspeexenc.c:
	  speex: fix latency query
	  Speex should report 30 ms latency for narrowband mode, 34 otherwise.
	  Fixes #619018

2010-05-18 21:04:32 +0800  Philip <philipj@opera.com>

	* gst/matroska/ebml-read.c:
	  ebmlread: rm floatcast.h include (not used)

2010-05-17 05:36:00 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: bump default doctype version to 2
	  In this day and age this should be safe. There's otherwise a risk people
	  will be creating unneccessarily big WebM files as they can't use
	  SimpleBlock in v1.

2010-05-17 05:27:44 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: handle matroska and webm doctype versions equally
	  The original plan was to let WebM v1 be the same as Matroska v2 (with
	  extra constraints), but for simplicity it was decided to handle the
	  versions equally, such that e.g. SimpleBlock is only allowed in WebM v2.

2010-05-13 12:10:54 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Verify lace size in _parse_blockgroup_or_simpleblock
	  Failure to do this for corrupt input can cause a subbuffer bigger
	  than the actual buffer to be created, quickly leading to segfault.
	  Test case:
	  bug_s222005751_r0.001____memcpy.webm

2010-05-13 10:23:10 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  ebml: crude hack to avoid crashing on unexpected metadata
	  The comment says this cannot happen, but it did and I don't know
	  why. This is not the correct fix, needs investigation. Test case:
	  bug_s555010094_r0.0005:0.008____IA__g_assertion_message_expr.webm

2010-05-13 09:18:56 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/ebml-read.c:
	  ebml: don't modify out str if returning an error in _read_ascii
	  This is a regression from ASCII validation changes. Test case:
	  bug_s66876390_r0.001____malloc_printerr.webm

2010-05-12 13:16:28 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/ebml-read.c:
	  ebml: Validate 7-bit ASCII in gst_ebml_read_ascii
	  This was triggering an UTF-8 assertion in gst_caps_set_simple for
	  corrupt files with garbage as codec id. Test case:
	  gstreamer_error_trying_to_set_invalid_utf8_as_codec_id.webm
	  Old gst_ebml_read_ascii renamed to gst_ebml_read_string, also used by
	  gst_ebml_read_utf8. Unlike for UTF-8, failure to validate is an error,
	  as gst_ebml_read_ascii is used for reading doctype and codec id and we
	  might just as well give up early in those cases.

2010-05-12 14:30:18 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Ignore unexpected CodecState
	  Because GstMatroskaTrackContext *stream is set up in the first
	  SimpleBlock or Block, a rogue CodecState otherwise causes a segfault on
	  derefencing the NULL pointer. Test case:
	  bug_s5506167_r0.001____gst_matroska_demux_parse_blockgroup_or_simpleblock.webm

2010-05-10 06:00:49 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Add video/webm sink caps

2010-05-09 19:46:51 +0200  Philip Jägenstedt <philip@foolip.org>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Use SimpleBlock for WebM when possible

2010-05-09 19:28:59 +0200  Philip Jägenstedt <philip@foolip.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Support "webm" DocType

2010-05-09 12:35:10 +0200  Philip Jägenstedt <philip@foolip.org>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: rename matroska_version to doctype_version

2010-05-09 12:09:57 +0200  Philip Jägenstedt <philip@foolip.org>

	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: Support "webm" DocType

2010-05-12 18:38:48 -0700  David Schleef <ds@schleef.org>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Add VP8

2010-04-27 15:26:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: Add support for On2 VP8
	  ...matroskademux automatically supports it through libgstriff.

2010-04-27 15:25:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: Add support for On2 VP8
	  ...avidemux automatically supports it through libgstriff.

2010-05-17 17:17:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulse: Don't lock the mainloop in NULL

2010-05-15 21:15:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Use = instead of == in shell scripts for equality checks

2010-05-14 18:33:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.22.2 pre-release

2010-05-14 18:24:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 4d67bd6 to 357b0db

2010-05-14 18:16:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/souphttpsrc.c:
	  tests: fix leak in souphttpsrc unit test
	  Unref server objects when done. Fixes check-valgrind.

2010-05-14 17:30:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: fix two leaks
	  Don't leak othercaps or jpegenc ref.

2010-05-13 13:01:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix docs
	  Documentation error spotted by tony <caicai0119 at gmail.com>
	  Fixes #618419

2010-05-11 13:18:42 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: make delivery-method parameter optional
	  It probably will not be in the final RFC as it is not in RFC 5215 for Vorbis.
	  If there is a configuration specified, assume it is in-line and if nothing is
	  specified, assume it is in-band.
	  https://bugzilla.gnome.org/show_bug.cgi?id=618386

2010-05-13 12:16:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: increase acceptable output sizes
	  We can perfectly decode 1x1 images so lower the min width and height to 1.
	  Fixes #618392

2010-05-13 11:30:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpceltpay.c:
	  celtpay: fix queue duration calculations
	  Don't blindly add the durations of incomming buffers to the total queued
	  duration because it might be invalid. Mark the total queued duration invalid
	  when we receive an invalid incomming timestamp because that's when we lose track
	  of the total queued duration.
	  Fixes #618324

2010-05-10 11:14:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: extract SPS and PPS from property provided parameter set
	  ... so it can also be regularly inserted into the stream if so configured.
	  Fixes #617164.

2010-05-11 22:28:08 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: allow switching views at runtime.

2010-05-11 20:26:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/Makefile.am:
	  rtp: dist missing header file to fix make distcheck

2010-05-11 19:05:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/oss4/oss4-sink.c:
	  oss4: minor cleanup
	  Remove fixed FIXME, change finalise to finalize for consistency.

2010-05-11 19:01:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-oss4.xml:
	  docs: add oss4 elements to docs

2010-05-11 16:09:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/ky.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: move oss4 strings from -bad to -good

2010-05-11 16:08:21 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* gst-plugins-good.spec.in:
	* po/POTFILES.in:
	* sys/Makefile.am:
	* tests/icles/.gitignore:
	* tests/icles/Makefile.am:
	  Move oss4 plugin from -bad to -good
	  Hook up build infrastructure, docs and tests.
	  Fixes #614305.

2010-04-29 13:18:58 +0100  Brian Cameron <brian.cameron@oracle.com>

	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-sink.h:
	  oss4sink: implement GstStreamVolume interface and add mute and volume properties
	  OSS4 supports per-stream volume control, so expose this using the right
	  API, so that playbin2 and applications like totem can make use of it
	  (instead of using a volume element for volume control).
	  Fixes #614305.

2010-04-08 10:45:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/oss4/oss4-audio.c:
	  oss4: 8-bit PCM audio caps don't need an endianness field

2010-04-08 10:40:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/oss4/oss4-audio.c:
	  oss4: don't iterate the formats table twice for each entry
	  When iterating the formats table, we can just pass the whole
	  entry to our helper function, which avoids iterating the table
	  again to find the entry structure from the passed format id.

2010-03-30 11:43:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/oss4/oss4-audio.c:
	  oss4: also accept formats not natively supported
	  Also accept formats that are not natively supported by the
	  hardware, OSS4 can convert them internally. List the native
	  formats first in the caps though, to express our preference
	  for the native formats. We need this in order to support the
	  case properly where the audio hardware supports only e.g.
	  little endian PCM, but the host is big endian, since many
	  audio elements only support native endianness and make the
	  reasonable assumption that any audiosink will be able to
	  handle audio in native endianness.
	  Based on patch by Jerry Tan <jerry.tan@sun.com>
	  Fixes #614317.

2010-03-30 01:14:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/oss4/oss4-mixer.c:
	  oss4: add comment for translators
	  Not that that will make these strings much better. Also remove i18n
	  marker where it doesn't make sense.

2010-03-22 16:13:12 +0100  Benjamin Otte <otte@redhat.com>

	* sys/oss4/oss4-mixer.c:
	  oss4: Refactor code to make it look more modern
	  A side effect is that it passes -Wformat-nonliteral and doesn't read
	  invalid memory in some cases, like when the mixer track contains
	  a % sign or there is a number but not a known mixer name.

2010-03-22 14:09:24 +0100  Benjamin Otte <otte@redhat.com>

	* sys/oss4/oss4-mixer.c:
	  oss4: Avoid g_quark_to_string (g_quark_from_string ()) madness
	  We to the strdup inside gst_oss4_mixer_control_get_translated_name()
	  instead of in the only caller.

2010-03-21 21:39:18 +0100  Benjamin Otte <otte@redhat.com>

	* sys/oss4/oss4-mixer.c:
	  Add -Wmissing-declarations -Wmissing-prototypes to configure flags
	  And fix all warnings

2010-01-20 13:29:52 +0100  Benjamin Otte <otte@redhat.com>

	* sys/oss4/oss4-mixer.c:
	  Fix compiler warning about unused return value

2009-08-21 01:17:18 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/icles/test-oss4.c:
	  tests: fix test-oss4 to treat an empty device name the same as a NULL name

2009-07-16 13:55:14 +0100  Jan Schmidt <thaytan@noraisin.net>

	* sys/oss4/oss4-mixer.c:
	  oss4: Attempt to fix a compiler warning
	  Don't store a const gchar * in a non-const gchar * local var.
	  Also, make the translation string function static since it's only
	  used in the one file.

2009-06-10 19:21:21 +0100  Garrett D'Amore <garrett.damore@sun.com>

	* sys/oss4/oss4-audio.c:
	* sys/oss4/oss4-mixer-slider.c:
	* sys/oss4/oss4-mixer-switch.c:
	* sys/oss4/oss4-mixer.c:
	  oss4: Enhancements to the mixer and audio output
	  Code cleanups, general improvements, support for the
	  new mixer flags in latest gst-plugins-base.
	  Fixes: #584252
	  Patch By: Brian Cameron <brian.cameron@sun.com>
	  Patch By: Garrett D'Amore <garrett.damore@sun.com>

2009-06-19 16:21:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/oss4/oss4-mixer.c:
	  Make build without warnings with debugging disabled

2008-11-04 12:42:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Don't install static libs for plugins. Fixes #550851 for -bad.
	  Original commit message from CVS:
	  * ext/alsaspdif/Makefile.am:
	  * ext/amrwb/Makefile.am:
	  * ext/apexsink/Makefile.am:
	  * ext/arts/Makefile.am:
	  * ext/artsd/Makefile.am:
	  * ext/audiofile/Makefile.am:
	  * ext/audioresample/Makefile.am:
	  * ext/bz2/Makefile.am:
	  * ext/cdaudio/Makefile.am:
	  * ext/celt/Makefile.am:
	  * ext/dc1394/Makefile.am:
	  * ext/dirac/Makefile.am:
	  * ext/directfb/Makefile.am:
	  * ext/divx/Makefile.am:
	  * ext/dts/Makefile.am:
	  * ext/faac/Makefile.am:
	  * ext/faad/Makefile.am:
	  * ext/gsm/Makefile.am:
	  * ext/hermes/Makefile.am:
	  * ext/ivorbis/Makefile.am:
	  * ext/jack/Makefile.am:
	  * ext/jp2k/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/lcs/Makefile.am:
	  * ext/libfame/Makefile.am:
	  * ext/libmms/Makefile.am:
	  * ext/metadata/Makefile.am:
	  * ext/mpeg2enc/Makefile.am:
	  * ext/mplex/Makefile.am:
	  * ext/musepack/Makefile.am:
	  * ext/musicbrainz/Makefile.am:
	  * ext/mythtv/Makefile.am:
	  * ext/nas/Makefile.am:
	  * ext/neon/Makefile.am:
	  * ext/ofa/Makefile.am:
	  * ext/polyp/Makefile.am:
	  * ext/resindvd/Makefile.am:
	  * ext/sdl/Makefile.am:
	  * ext/shout/Makefile.am:
	  * ext/snapshot/Makefile.am:
	  * ext/sndfile/Makefile.am:
	  * ext/soundtouch/Makefile.am:
	  * ext/spc/Makefile.am:
	  * ext/swfdec/Makefile.am:
	  * ext/tarkin/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/timidity/Makefile.am:
	  * ext/twolame/Makefile.am:
	  * ext/x264/Makefile.am:
	  * ext/xine/Makefile.am:
	  * ext/xvid/Makefile.am:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/dshow/Makefile.am:
	  * gst/aiffparse/Makefile.am:
	  * gst/app/Makefile.am:
	  * gst/audiobuffer/Makefile.am:
	  * gst/bayer/Makefile.am:
	  * gst/cdxaparse/Makefile.am:
	  * gst/chart/Makefile.am:
	  * gst/colorspace/Makefile.am:
	  * gst/dccp/Makefile.am:
	  * gst/deinterlace/Makefile.am:
	  * gst/deinterlace2/Makefile.am:
	  * gst/dvdspu/Makefile.am:
	  * gst/festival/Makefile.am:
	  * gst/filter/Makefile.am:
	  * gst/flacparse/Makefile.am:
	  * gst/flv/Makefile.am:
	  * gst/games/Makefile.am:
	  * gst/h264parse/Makefile.am:
	  * gst/librfb/Makefile.am:
	  * gst/mixmatrix/Makefile.am:
	  * gst/modplug/Makefile.am:
	  * gst/mpeg1sys/Makefile.am:
	  * gst/mpeg4videoparse/Makefile.am:
	  * gst/mpegdemux/Makefile.am:
	  * gst/mpegtsmux/Makefile.am:
	  * gst/mpegvideoparse/Makefile.am:
	  * gst/mve/Makefile.am:
	  * gst/nsf/Makefile.am:
	  * gst/nuvdemux/Makefile.am:
	  * gst/overlay/Makefile.am:
	  * gst/passthrough/Makefile.am:
	  * gst/pcapparse/Makefile.am:
	  * gst/playondemand/Makefile.am:
	  * gst/rawparse/Makefile.am:
	  * gst/real/Makefile.am:
	  * gst/rtjpeg/Makefile.am:
	  * gst/rtpmanager/Makefile.am:
	  * gst/scaletempo/Makefile.am:
	  * gst/sdp/Makefile.am:
	  * gst/selector/Makefile.am:
	  * gst/smooth/Makefile.am:
	  * gst/smoothwave/Makefile.am:
	  * gst/speed/Makefile.am:
	  * gst/speexresample/Makefile.am:
	  * gst/stereo/Makefile.am:
	  * gst/subenc/Makefile.am:
	  * gst/tta/Makefile.am:
	  * gst/vbidec/Makefile.am:
	  * gst/videodrop/Makefile.am:
	  * gst/videosignal/Makefile.am:
	  * gst/virtualdub/Makefile.am:
	  * gst/vmnc/Makefile.am:
	  * gst/y4m/Makefile.am:
	  * sys/acmenc/Makefile.am:
	  * sys/cdrom/Makefile.am:
	  * sys/dshowdecwrapper/Makefile.am:
	  * sys/dshowsrcwrapper/Makefile.am:
	  * sys/dvb/Makefile.am:
	  * sys/dxr3/Makefile.am:
	  * sys/fbdev/Makefile.am:
	  * sys/oss4/Makefile.am:
	  * sys/qcam/Makefile.am:
	  * sys/qtwrapper/Makefile.am:
	  * sys/vcd/Makefile.am:
	  * sys/wininet/Makefile.am:
	  * win32/common/config.h:
	  Don't install static libs for plugins. Fixes #550851 for -bad.

2008-10-12 21:52:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/oss4/: Add some spaces in translateable strings.
	  Original commit message from CVS:
	  * sys/oss4/oss4-mixer.c:
	  * sys/oss4/oss4-sink.c:
	  * sys/oss4/oss4-source.c:
	  Add some spaces in translateable strings.
	  Fixes: #555969 #555968 #555965

2008-08-07 16:20:30 +0000  Frederic Crozat <fcrozat@mandriva.org>

	  Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
	  Original commit message from CVS:
	  Patch by: Frederic Crozat <fcrozat@mandriva.org>
	  * ext/sndfile/gstsf.c: (plugin_init):
	  * sys/dvb/gstdvbsrc.c: (gst_dvbsrc_plugin_init):
	  * sys/oss4/oss4-audio.c: (plugin_init):
	  Make sure gettext returns translations in UTF-8 encoding rather
	  than in the current locale encoding (#546822).

2008-06-16 07:30:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Final round of doc updates.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/speed/gstspeed.c:
	  * gst/speexresample/gstspeexresample.c:
	  * gst/videosignal/gstvideoanalyse.c:
	  * gst/videosignal/gstvideodetect.c:
	  * gst/videosignal/gstvideomark.c:
	  * sys/dvb/gstdvbsrc.c:
	  * sys/oss4/oss4-mixer.c:
	  * sys/oss4/oss4-sink.c:
	  * sys/oss4/oss4-source.c:
	  * sys/wininet/gstwininetsrc.c:
	  Final round of doc updates.

2008-06-12 14:49:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Do not use short_description in section docs for elements. We extract them from element details and there will be war...
	  Original commit message from CVS:
	  * ext/dc1394/gstdc1394.c:
	  * ext/ivorbis/vorbisdec.c:
	  * ext/jack/gstjackaudiosink.c:
	  * ext/metadata/gstmetadatademux.c:
	  * ext/mythtv/gstmythtvsrc.c:
	  * ext/theora/theoradec.c:
	  * gst-libs/gst/app/gstappsink.c:
	  * gst/bayer/gstbayer2rgb.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/rawparse/gstaudioparse.c:
	  * gst/rawparse/gstvideoparse.c:
	  * gst/rtpmanager/gstrtpbin.c:
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/rtpmanager/gstrtpptdemux.c:
	  * gst/rtpmanager/gstrtpsession.c:
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  * gst/selector/gstinputselector.c:
	  * gst/selector/gstoutputselector.c:
	  * gst/videosignal/gstvideoanalyse.c:
	  * gst/videosignal/gstvideodetect.c:
	  * gst/videosignal/gstvideomark.c:
	  * sys/oss4/oss4-mixer.c:
	  * sys/oss4/oss4-sink.c:
	  * sys/oss4/oss4-source.c:
	  Do not use short_description in section docs for elements. We extract
	  them from element details and there will be warnings if they differ.
	  Also fixing up the ChangeLog order.

2008-06-12 13:06:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/icles/test-oss4.c: Include stdlib.h.
	  Original commit message from CVS:
	  * tests/icles/test-oss4.c:
	  Include stdlib.h.

2008-05-22 16:33:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/: Small oss4 test that probes for available devices and retrieves their caps and mixer tracks and all tha...
	  Original commit message from CVS:
	  * tests/icles/.cvsignore:
	  * tests/icles/Makefile.am:
	  * tests/icles/test-oss4.c: (opt_show_mixer_messages), (WAIT_TIME),
	  (show_mixer_messages), (probe_mixer_tracks), (probe_pad),
	  (probe_details), (probe_element), (main):
	  Small oss4 test that probes for available devices and retrieves
	  their caps and mixer tracks and all that. Also allows testing of
	  mixer change messages on the bus.

2008-05-22 15:14:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss4/: Make device-name probing in NULL state work better (e.g. for the gnome-control-center sound capplet).
	  Original commit message from CVS:
	  * sys/oss4/oss4-mixer.c: (gst_oss4_mixer_open):
	  * sys/oss4/oss4-property-probe.c:
	  (gst_oss4_property_probe_find_device_name),
	  (gst_oss4_property_probe_find_device_name_nofd):
	  * sys/oss4/oss4-property-probe.h:
	  * sys/oss4/oss4-sink.c: (gst_oss4_sink_get_property):
	  * sys/oss4/oss4-source.c: (gst_oss4_source_get_property):
	  Make device-name probing in NULL state work better (e.g. for the
	  gnome-control-center sound capplet).

2008-05-08 19:16:17 +0000  Clive Wright <clive_wright@ntlworld.com>

	  sys/oss4/oss4-mixer-slider.c: Apparently mono sliders have the mono value repeated in the upper bits, so mask those o...
	  Original commit message from CVS:
	  Based on patch by: Clive Wright <clive_wright ntlworld com>
	  * sys/oss4/oss4-mixer-slider.c: (gst_oss4_mixer_slider_unpack_volume):
	  Apparently mono sliders have the mono value repeated in the upper bits,
	  so mask those out when reading them. Probably makes the mixer applet
	  work properly in some more cases.

2008-04-11 08:13:22 +0000  Julien Moutte <julien@moutte.net>

	  sys/oss4/: Fix arguments format in debug statements.
	  Original commit message from CVS:
	  2008-04-11  Julien Moutte  <julien@fluendo.com>
	  * sys/oss4/oss4-mixer-enum.c:
	  (gst_oss4_mixer_enum_get_values_locked):
	  * sys/oss4/oss4-source.c: (gst_oss4_source_delay): Fix arguments
	  format in debug statements.

2008-04-02 20:18:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add initial support for OSSv4. Mixer still needs a bit more love, but even magic has its limits.
	  Original commit message from CVS:
	  * configure.ac:
	  * sys/Makefile.am:
	  * sys/oss4/Makefile.am:
	  * sys/oss4/oss4-audio.c:
	  * sys/oss4/oss4-audio.h:
	  * sys/oss4/oss4-mixer-enum.c:
	  * sys/oss4/oss4-mixer-enum.h:
	  * sys/oss4/oss4-mixer-slider.c:
	  * sys/oss4/oss4-mixer-slider.h:
	  * sys/oss4/oss4-mixer-switch.c:
	  * sys/oss4/oss4-mixer-switch.h:
	  * sys/oss4/oss4-mixer.c:
	  * sys/oss4/oss4-mixer.h:
	  * sys/oss4/oss4-property-probe.c:
	  * sys/oss4/oss4-property-probe.h:
	  * sys/oss4/oss4-sink.c:
	  * sys/oss4/oss4-sink.h:
	  * sys/oss4/oss4-soundcard.h:
	  * sys/oss4/oss4-source.c:
	  * sys/oss4/oss4-source.h:
	  Add initial support for OSSv4. Mixer still needs a bit more love,
	  but even magic has its limits.

2010-05-11 10:52:58 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: implement the xoverlay interface. Fixes #618349.

2010-05-11 18:42:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix push based seeking
	  ... where it comes down to transforming incoming BYTE segment
	  to a corresponding TIME segment.
	  Also fixes #609405.

2010-05-11 14:23:47 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	  Move imagefreeze plugin from -bad to -good
	  Hook up build infrastructure, docs and unit test for new plugin.
	  Fixes #613786.

2010-05-05 12:23:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Set fixed caps on the correct pad
	  This makes the sink getcaps function actually used instead of using
	  the fixed caps function for it.

2010-03-21 21:39:18 +0100  Benjamin Otte <otte@redhat.com>

	* tests/check/elements/imagefreeze.c:
	  Add -Wmissing-declarations -Wmissing-prototypes to configure flags
	  And fix all warnings

2010-03-15 11:54:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Only start the task after a seek if a buffer was received already

2010-02-28 16:08:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/imagefreeze.c:
	  imagefreeze: Add some unit tests

2010-02-28 16:04:31 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Set undefined framerate in sink getcaps function

2010-02-28 15:02:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: Implement reverse playback and set buffer offsets

2010-02-27 17:33:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/imagefreeze/Makefile.am:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/imagefreeze/gstimagefreeze.h:
	  imagefreeze: Add still frame stream generator element

2010-05-11 13:07:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-debug.xml:
	* gst/debugutils/Makefile.am:
	* gst/debugutils/gstdebug.c:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	  Move capsfilter element from -bad to -good
	  Hook up moved files to the build infrastructure and docs.
	  Fixes #617739.

2010-05-06 13:12:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstcapssetter.h:
	  capssetter: Some minor cleanup

2010-03-22 16:56:03 +0100  Benjamin Otte <otte@redhat.com>

	* tests/check/elements/capssetter.c:
	  Add -Wold-style-definition
	  and fix the warnings

2010-03-18 17:30:26 +0100  Benjamin Otte <otte@redhat.com>

	* gst/debugutils/gstcapssetter.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple

2009-10-08 19:51:31 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/capssetter.c:
	  capssetter: add unit test

2009-06-25 16:41:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstcapssetter.h:
	  capssetter: import element into -bad

2010-05-11 12:06:10 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: check that pads have been negotiated
	  Also set fcc_handler field in audio stream header.
	  Fixes #618351.

2010-05-10 18:33:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix partial parsing of ctts table
	  Fixes #616516.

2010-05-10 18:32:15 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: cleanup a comment and add some debug and conditional compilation

2010-05-11 10:01:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Check for GTK+ 3.0 and if it's not available for GTK+ 2.0

2010-05-10 22:11:10 +0200  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: only store the last buffer timestamp if it's valid
	  Fixes bug #618305

2010-01-08 22:13:59 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Re-send SPS/PPS when requested
	  https://bugzilla.gnome.org/show_bug.cgi?id=606689

2010-05-07 17:09:16 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: fix typo in debug message

2010-05-07 15:42:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtptheorapay.h:
	  rtptheorapay: add config-interval parameter to re-insert config in stream
	  Add a new config-interval property to instruct the payloader to insert
	  configuration headers at periodic intervals in the stream
	  (when a keyframe is countered).

2010-05-07 15:31:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: fix in-band configuration parsing
	  Also make configuration header parsing a bit more relaxed with respect
	  to length field interpretation.

2010-05-07 15:30:30 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbisdepay: fix in-line configuration parsing
	  Also make configuration header parsing a bit more relaxed with respect
	  to length field interpretation.

2010-05-04 16:57:35 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: do not discard downstream flow return

2010-05-04 16:57:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: refactor buffer payloading

2010-05-07 20:41:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Add support for UYVY

2010-05-07 19:06:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: fix return value

2010-05-07 19:02:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't leak the session

2010-05-07 18:59:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: configure bandwidth properties in the session

2010-05-07 18:58:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: add properties to configure the bandwidth
	  Add properties to proxy the bandwidth configuration to the session object.

2010-05-07 18:57:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: add properties to configure bandwidths
	  Add properties to configure the sender and receiver bandwidths.
	  Configure the bandwidths before calculating the RTCP timeout when we need to.

2010-05-07 18:56:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpstats.c:
	  rtpstats: add some debug info

2010-05-07 18:55:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: small cleanups

2010-05-07 16:55:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpstats: make bandwidths more configurable
	  Add a method to configure the various bandwidths in the session.

2010-05-07 13:32:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: handle NONE RTCP intervals
	  Prepare for handling RTCP reporting intervals of GST_CLOCK_TIME_NONE, which
	  means don't send RTCP at all.

2010-05-07 12:51:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: fall back to SDP ports instead of server_port
	  In multicast, fall back to the ports in the SDP instead of the server_port
	  attribute as this is more in line with the RFC.

2010-05-07 12:24:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: refactor collecting the transport info
	  Make a method to collect the ports and destination address.

2010-05-07 11:28:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle servers that send broken Transports
	  Handle servers that send their port pairs with the wrong name.
	  Fixes #617537

2010-05-06 16:52:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: use the SDP connection info in multicast
	  Parse the connection info from the SDP.
	  When we need to configure the multicast destination, fall back to the SDP
	  connection info when the transport did not specify a destination and ttl.
	  Fixes #617537

2010-05-06 15:42:38 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/monoscope/gstmonoscope.c:
	  goom,monoscope: truncate own caps, instead of copying and using the first only
	  We got the caps from an intersect, it is our own, hence we can truncate it.

2010-05-06 15:40:33 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: reflow to truncate caps just once
	  We get writable cpas from the intersection (unless it failed). As we truncate
	  those anyway, we don't need to manyaly copy the first structure.

2010-05-06 15:39:31 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	  gdkpixbuf: don't leak template caps

2010-05-06 15:38:35 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	  auto{audio,video}{src,sink}: use can_intersect to avoid a caps copy

2010-04-27 13:36:35 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: tell what we can do
	  Any-caps are bad. If apps scan the registry, they'd like to know what we can
	  output.

2010-04-27 13:43:29 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: also lift the arbitrary restrictions for width and height
	  This was already done for jpegdec.

2010-05-06 14:03:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Allocate/free PA mainloop during state changes
	  ...also destroy the stream and context during state changes.

2010-05-06 13:57:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Allocate and free the custom clock in NULL<->READY

2010-05-06 13:51:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Create and free the PA mainloop in NULL->READY/READY->NULL
	  This fixes a race condition, when stopping the mainloop during finalization
	  is done from a mainloop callback.
	  Fixes bugs #614765 and #590662.

2010-05-05 19:35:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Make selection of a sinkpad number threadsafe

2010-05-05 17:39:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Add support for all common RGB formats

2010-05-05 16:06:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.asm:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Add support for AYUV

2010-05-04 16:34:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: make setup url in a smarter way
	  Make sure we always separate the base and control url parts with a / when
	  creating the setup url.

2010-05-04 16:04:39 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle SEEKING queries.

2010-05-04 11:13:45 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	  rtpmp4vpay: add config-interval parameter to re-insert config in stream
	  Add a new config-interval property to instruct the payloader to insert
	  config (VOSH, VOS, etc) at periodic intervals in the stream
	  (when a GOP or VOP-I is encountered).
	  Based on patch by <marc.leeman at gmail.com>
	  Fixes #607452.

2010-05-03 13:26:32 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: move some initialization code from change_state to _init.
	  Set ->active to TRUE in _init so it can be set to FALSE after creating the
	  jitterbuffer and it won't be mistakenly reset to TRUE in the change_state
	  function.
	  This is needed to start the jitterbuffer as inactive when rtpbin is buffering.

2010-05-03 11:56:58 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix a bug handling BUFFERING messages.
	  If a session exists but has no streams, set the min buffering percent to 0
	  since it means that we haven't received anything for that session yet.

2010-05-03 11:51:37 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: when a stream is created, pause the jitterbuffer if rtpbin is buffering.

2010-05-03 11:23:59 +0200  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix a bug calculating stream offsets.

2010-05-01 14:20:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: Write previous cluster's size
	  This is useful for backwards playback, which should be implemented
	  in matroskademux at some point.

2010-05-01 14:15:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Set interlaced flag in the caps if the flag is set in the Matroska file

2010-05-01 14:12:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Write interlaced flag if the input video content is interlaced
	  Unfortunately Matroska has no way to specify TFF and friends...

2010-05-01 11:25:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	  rtp: fix printf format of some debug messages

2010-05-01 11:06:53 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska: init variable to avoid compiler warning on OSX
	  Fixes (bogus) "'offset' may be used uninitialized in this function"
	  warning on build bot (also spotted by philn).

2010-04-30 17:19:44 -0700  David Schleef <ds@schleef.org>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: UYVY is 4:2:2, not 4:2:0

2010-04-30 22:22:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulseutil.c:
	  pulse: Don't compare values of two different enum types

2010-04-30 22:13:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Make automatic detection of interlacing the default
	  Previously "force deinterlacing" was the default, which is a not very
	  sensible default for the normal use case where deinterlace should act
	  in passthrough mode unless interlaced content is present.

2010-04-29 16:26:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: optimise buffer scanning
	  Specifically, when needing more data, do not rescan from start next time
	  around, but resume from last position.
	  See also #583047.

2010-04-29 15:38:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: disregard superfluous lines when indirect decoding

2010-04-27 15:44:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: add support for RGB and grayscale color space
	  Also refactor src caps negotiation and setting.

2010-04-27 12:19:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/Makefile.am:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpegenc: support more colour spaces and some cleanups

2010-04-30 12:47:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: more generic sink getcaps

2010-04-30 12:42:42 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: more sanity checks on input
	  Specifically, verify input components / colour space is as code
	  subsequently expects, thereby avoiding crashes or otherwise bogus output.
	  Presently, that means 3 components YCbCr colour space, and somewhat
	  limited sampling factors.
	  Fixes #600553.

2010-04-22 12:28:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: also accept in-band configuration
	  Fixes #574416 (theora).

2010-04-22 12:27:35 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbisdepay: also accept in-line configuration
	  Fixes #574416 (vorbis).

2010-04-07 17:21:55 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: Ignore packets without a known codebook
	  Don't produce an error if a packet is received without a valid codebook,
	  it's possible that the codebook will just be coming later.
	  See #574416.

2010-04-20 12:17:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/y4menc.c:
	  y4menc: adjust unit test to element behaviour

2010-02-23 22:16:39 -0500  Benjamin M. Schwartz <bens@alum.mit.edu>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  y4menc: add 4:2:2, 4:1:1, and 4:4:4 output support
	  Fixes #610902.

2010-04-15 12:21:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: DELTA_UNIT marking of output buffers
	  ... which evidently makes (most) sense if output buffers are
	  actually frames.
	  Partially based on a patch by
	  Miguel Angel Cabrera <mad_aluche at hotmail.com>
	  Fixes #609658.

2010-04-16 17:21:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263depay.h:
	  rtph263depay: extra keyframe info from PTYPE header
	  ... as opposed to taking it from h263 payload header, which need not
	  be so reliable.
	  Fixes #610172.

2010-04-16 17:08:47 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: also use Picture Start Code to detect packet loss
	  This ensures a whole frame is dropped if a (start) packet is lost,
	  rather than relying only on the DISCONT flag.

2010-04-16 17:06:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: detect frame start using Picture Start Code
	  So we stop dropping fragments as soon as there is a picture start (code).
	  In particular, this prevents dropping the first frame following
	  initial DISCONT.

2010-04-16 16:34:06 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: handle a few FIXMEs

2010-04-16 16:27:25 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: slightly refactor payload dropping

2010-04-16 11:53:17 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	  rtph263pay: use found GOBs to apply Mode A payloading
	  ... rather than falling back to sending the whole frame in one packet
	  if number of GOB startcodes < maximum.
	  One might take this further and still perform Mode B/C payloading,
	  but at least this should cater for decent fragments in typical cases.
	  Fixes #599585.

2010-04-14 11:53:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: implement push mode seeking

2010-04-29 20:08:43 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* gst/smpte/gstsmptealpha.c:
	  docs: update for videofilter plugin merge and add gtk-doc blurb for new property

2010-04-26 18:12:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Improve segment handling a bit

2010-04-26 18:05:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Order caps by amount of contained information

2010-04-26 17:25:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Properly set interlaced field in getcaps

2010-04-24 16:28:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Add planar YUV support to all other simple methods

2010-04-24 16:10:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/greedyh.asm:
	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: Add planar YUV support to greedyh method

2010-04-24 15:42:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/greedy.c:
	  deinterlace: Add support for planar YUV formats in greedyl method

2010-04-24 13:58:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/vfir.c:
	  deinterlace: Add support for Y444, Y42B, I420, YV12 and Y41B
	  The vfir method supports them and will be used until something else
	  supports it.

2010-04-24 09:16:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlacemethod.c:
	  deinterlace: Define deinterlace method base classes as abstract types

2010-04-23 17:40:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/Makefile.am:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Move deinterlacing methods to their own file

2010-04-23 17:25:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Simplify passthrough mode detection

2010-04-23 14:35:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Fix unit test that checks caps handling
	  deinterlace now always adds the interlaced field to the output caps,
	  if it wasn't present in the input caps the output caps will still
	  contain interlaced=false.

2010-04-21 17:00:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/Makefile.am:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.asm:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Refactor deinterlacing as preparation for supporting more color formats

2010-04-22 19:05:37 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Add support for Y444, Y42B and Y41B

2010-04-22 15:54:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Add support for YVYU and reorder template caps

2010-04-18 21:11:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Translate navigation events to make sense again upstream

2010-04-18 20:58:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Properly handle ranges/lists of width or height when transforming caps
	  Code partly taken from the videocrop element.

2010-04-22 15:45:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Fix planar YUV->RGB processing

2010-04-22 15:42:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Correctly clamp after YUV->RGB conversion

2010-04-22 15:20:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Add support for YUY2, YVYU and UYVY

2010-04-18 15:02:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Sync properties to the controller in before_transform

2010-04-16 17:00:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Add support for YUY2 and UYUV

2010-04-21 17:41:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Refactor processing and add support for other planar YUV formats
	  This reduces the generated code size by a factor of 2.5.

2010-04-21 17:15:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Add support for YV12 input

2010-04-22 13:56:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/videomixer.c:
	  videomixer: Add support for YUY2, YVYU, UYVY

2010-04-20 12:18:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/videomixer.c:
	  videomixer: Add support for Y444, Y42B, Y41B and YV12

2010-04-21 17:07:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	  videofilter: Order color formats by their contained amount of information

2010-04-20 18:22:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Drop Y41B/Y42B support
	  Rotating 90°/270° with subsampled YUV where horizontal
	  and vertical subsampling are different doesn't really work.

2010-04-19 14:37:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Also flip the pixel-aspect-ratio if width/height are exchanged

2010-04-18 23:08:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/videofilter.c:
	  videofilter: Extend the unit test to test different color formats

2010-04-18 22:55:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/videofilter.c:
	  videofilter: Add some more tests
	  These check different property combinations

2010-04-18 22:54:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Change the default method to identity

2010-04-18 22:50:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideobalance.h:
	  videobalance: Reduce number of allocations per instance

2010-04-18 22:45:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	  videofilter: Update last-reviewed comments

2010-04-18 22:40:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Add support for all RGB formats

2010-04-18 22:28:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Add support for YUY2, UYVY, AYUV and YVYU

2010-04-18 22:23:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Add debug category

2010-04-18 22:19:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Make property access threadsafe

2010-04-18 22:18:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Add support for Y41B, Y42B and Y444

2010-04-18 22:17:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideobalance.h:
	  videobalance: Use libgstvideo for format specific things

2010-04-18 22:09:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Make properties controllable

2010-04-18 22:06:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Emit "value-changed" signal of color balance interface when values change

2010-04-18 21:58:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideobalance.h:
	  videobalance: Some random cleanup

2010-04-18 21:37:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideobalance.c:
	  videobalance: Stop using liboil
	  The used liboil function is deprecated and has no optimized
	  implementation anyway.

2010-04-18 21:14:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Make property access threadsafe

2010-04-18 15:00:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	  gamma: Sync properties to the controller in before_transform

2010-04-18 14:46:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Add support for all RGB formats and AYUV

2010-04-18 14:31:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Add support for Y41B, Y42B and Y444

2010-04-18 14:29:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	  videoflip: Make processing more general and use libgstvideo for all format specific things

2010-04-18 13:12:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Make method property controllable and improve debug output

2010-04-18 13:03:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	  videoflip: Some random cleanup

2010-04-18 10:17:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* Makefile.am:
	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/plugin.c:
	  videofilter: Move all elements into a single plugin
	  Having all these small elements in a separate plugin
	  is not very memory effective...

2010-04-18 10:07:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstgamma.h:
	  gamma: Improve docs a bit

2010-04-18 09:59:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	  gamma: Add support for all RGB formats

2010-04-18 09:46:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	  gamma: Add support for many packed YUV formats
	  That is YUY2, UYVY, AYUV and YVYU.

2010-04-18 09:38:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	  gamma: Add support for all other planar YUV formats
	  That is Y41B, Y42B, Y444, NV12 and NV21.

2010-04-18 09:33:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstgamma.c:
	  gamma: Stop using liboil
	  The used liboil function is deprecated, only has a reference implementation
	  and is more complex than what's needed here.

2010-04-17 18:13:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstgamma.h:
	  gamma: Use libgstvideo for format specific values and make gamma processing more generic
	  Allows us to easily add support for new color formats later.

2010-04-17 18:01:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstgamma.c:
	  gamma: Make gamma property controllable
	  ...and properly use liboil.

2010-04-17 17:55:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videofilter/gstgamma.c:
	  gamma: Some random cleanup

2010-04-19 14:45:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/smpte/gstsmptealpha.c:
	  smptealpha: Sync properties to the controller in before_transform

2010-04-17 17:47:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/smpte/gstsmptealpha.c:
	  smptealpha: Add support for YV12 (converted to AYUV)

2010-04-17 17:43:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/smpte/gstsmptealpha.c:
	  smptealpha: Add support for all 4 ARGB formats
	  ...without format conversion.

2010-04-16 17:27:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/smpte/gstsmptealpha.c:
	* gst/smpte/gstsmptealpha.h:
	  smptealpha: Make color format support more generic
	  This allows easier addition of new formats later.

2010-04-16 17:18:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/smpte/gstsmptealpha.c:
	* gst/smpte/gstsmptealpha.h:
	  smptealpha: Some random cleanup

2010-04-15 22:28:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/smpte/gstmask.c:
	* gst/smpte/gstmask.h:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmpte.h:
	* gst/smpte/gstsmptealpha.c:
	* gst/smpte/gstsmptealpha.h:
	  smpte: Add property for inverting the transition mask
	  This converts a left-to-right transition to right-to-left or
	  clock-wise to counter-clock-wise.

2010-04-15 22:27:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/smpte/gstsmptealpha.c:
	  smptealpha: Correctly detect property changes and update properties

2010-04-16 19:35:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqcelpdepay.h:
	  qcelpdepay: add first version of a QCELP depayloader

2010-04-29 15:18:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development.

=== release 0.10.22 ===

2010-04-28 02:58:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.22

2010-04-28 02:57:21 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2010-04-25 23:36:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.21.3 pre-release

2010-04-25 21:19:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: hide is-live property for release
	  At the very least it needs a better/less wrong name.
	  See #613066.

2010-04-25 15:12:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: don't crash if jpeg image contains more than three components
	  Our code currently only handles a maximum of 3 components, so error
	  out for now if the image has more components than that.
	  Fixes #604106.

2010-04-20 17:21:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-plugins-good.doap:
	  doap: update repository info from cvs->git and maintainers

2010-04-23 14:40:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From fc85867 to 4d67bd6

2010-04-22 13:30:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend.c:
	  videomixer: Fix byte order for MMX ARGB/AYUV color filling
	  Fixes bug #616409.

2010-04-21 17:53:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend.c:
	  videomixer: Fix AYUV checker/color filling

2010-04-19 16:43:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_mmx.h:
	  videomixer: Add i387 floating point registers to the clobbered registers list
	  They are the same as the mm0-mm7 MMX registers and will be overwritten
	  by the assembly code if gcc doesn't know about the MMX registers.
	  Note: They're all added to the list of clobbered registers in all cases
	  and not only when __MMX__ is not defined just to make sure that no other
	  bugs happen with this code just because some compiler version gets things
	  wrong.
	  Fixes bug #614466.

2010-04-19 14:09:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Use libgstvideo to get the order of RGB

2010-04-17 10:06:41 +0100  Brian Cameron <brian.cameron@oracle.com>

	* gst/goom/xmmx.c:
	  goom: add edx to clobber list in inline assembly code
	  mull modifies %edx, so should be mentioned in clobber list.
	  Fixes crash on Solaris (#615998).

2010-04-15 13:39:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/icles/Makefile.am:
	  tests: don't use GST_PLUGIN_LDFLAGS when building test binaries

2010-04-16 15:27:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix I420->I420 copying
	  Fixes bug #615143.

2010-04-13 18:15:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix AYUV->I420 copying

2010-04-16 12:14:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: profile-level-id is an optional parameter
	  So, if needed, extract the corresponding info from
	  sprop-parameter-sets.
	  Based on patch provided by <dxssx at gmail.com>
	  Fixes #612657.

2010-04-15 07:13:46 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* configure.ac:
	  configure: Drop -Wcast-align
	  Commit message copied from core's commit from Benjamin Otte:
	  246f5dba96a5b50bb74621af67b30942cca72af5
	  Apparently gcc warns that GstMiniObject is not castable to
	  GstEvent/Message/Buffer due to them containing 64bit variables, even
	  though ARM hackers claim that those only need 4byte alignment. And as
	  long as gcc behaves that way, this warning is not very useful.
	  So we'll remove the warning until this problem is fixed.
	  Fixes #615698

2010-04-14 23:46:06 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflactag.c:
	  flactag: fix adapter assertion when used directly after flacenc
	  Unlike filesrc, flacenc outputs the flac blocks neatly aligned one in
	  each buffer. This means that when we switch from metadata mode to
	  audio data passthrough mode, there's no data left in the adapter to
	  push out at this point, so check if there's data in the adapter
	  before requesting buffers from it (also needed in case we get input
	  buffers of 0 size).
	  Fixes #615793.

2010-04-14 23:18:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.21.2 pre-release

2010-04-14 20:31:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update

2010-04-14 20:06:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/equalizer/Makefile.am:
	* tests/examples/shapewipe/Makefile.am:
	* tests/examples/spectrum/Makefile.am:
	* tests/examples/v4l2/Makefile.am:
	* tests/icles/Makefile.am:
	  tests: use LDADD for libs to link to instead of LDFLAGS
	  Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to.
	  This should make sure arguments are passed to the linker in the right
	  order, and makes LDFLAGS usable again.
	  Based on patch by Brian Cameron <brian.cameron@oracle.com>
	  Fixes #615697.

2010-04-14 18:13:56 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videobox/gstvideobox.c:
	  videobox: transform_caps : We can only convert AYUV to xRGB
	  We were previously stating that we could convert AYUV/I420/YV12 to xRGB.

2010-04-13 00:14:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: also remove -Waggregate-return from warning flags
	  It causes problems with Objective-C code like in osxvideosink.
	  Fixes #613663.

2010-04-12 18:22:39 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/Makefile.am:
	  check: Ignore osx audio/video src/sinks in state change tests
	  And make the line readable for those mere mortals that don't own a 30" screen

2010-04-12 18:03:20 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/cmmldec.c:
	* tests/check/elements/cmmlenc.c:
	* tests/check/elements/level.c:
	* tests/check/elements/matroskamux.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rglimiter.c:
	* tests/check/elements/rgvolume.c:
	* tests/check/elements/spectrum.c:
	* tests/check/elements/videofilter.c:
	  check: Don't re-declare 'GList *buffers' in the tests
	  It's an external which lives in gstcheck.c. Redeclaring it makes some
	  compilers/architectures think the 'buffers' in the individual tests are
	  a different symbol... and therefore we end up comparing holodecks with
	  oranges.

2010-04-12 14:50:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/qtdemux/qtdemux.c:
	  matroskademux, qtdemux: minor code cleanup in avc_level_idc_to_string()
	  Do the same with slightly fewer LOC.

2010-04-12 12:40:11 +0200  Edward Hervey <bilboed@bilboed.com>

	* configure.ac:
	  configure: Remove -Wundef flag
	  Fixes #615161

2010-04-12 11:43:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix I420->AYUV copying

2010-04-12 11:25:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Correctly clamp frame/background alphas to [0,255] before writing them

2010-04-12 11:16:56 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/.gitignore:
	  check: Ignore jpegenc test

2010-04-11 13:14:30 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Only check interlaced flag in sink caps
	  Fixes #615460.

2010-04-09 11:21:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From ba33d1f to fc85867

2010-04-08 18:05:46 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/rtpmanager/gstrtpbin.c:
	  docs: do proper escaping for "%"

2010-04-08 17:50:49 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtsp/gstrtspgoogle.c:
	* gst/rtsp/gstrtspgoogle.h:
	  rtsp: remove obsolete google extension
	  This was not build for a while and can be removed.

2010-04-08 17:42:52 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: move two symbols to private section

2010-04-08 17:36:30 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: add flxdec docs

2010-04-08 17:17:06 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegpay.c:
	  docs: enable the 2 of 65 rtp elements in the docs

2010-04-08 11:54:19 +0200  Benjamin Otte <otte@redhat.com>

	* ext/shout2/gstshout2.c:
	  shout2: Don't wait if we're late
	  In fact, due to signedness issues, a negative delay would be changed to
	  an almost infinite wait causing shout2send to "lock up".
	  Reported by Christopher Montgomery.

2010-04-08 16:56:37 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/udp/gstmultiudpsink.c:
	  docs: upd -> udp and voila it shows up in the docs

2010-04-08 16:51:27 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/alpha/gstalpha.h:
	  docs: fix doc blob syntax

2010-04-08 16:51:05 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: add (sparse) docs for auparse element

2010-04-08 14:40:43 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: add videobox symbols

2010-04-08 14:40:19 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	  docs: remove dynudpsink until someone documents it

2010-04-08 14:34:59 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: make debug category static

2010-04-08 14:29:19 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flxdemux: rename GstFLVDemux for GstFlvDemux

2010-04-08 14:23:19 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/flv/Makefile.am:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvparse.c:
	* gst/flv/gstflvparse.h:
	  flvdemux: merge flvparse into the demuxer and make function static
	  No need to hide certain function in the docs. Allows to do more cleanups.

2010-04-08 13:13:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: Add documentation

2010-04-08 14:00:08 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: v4l2buffer pool is now a separate object, remove them from v4l2src docs

2010-04-08 13:58:11 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: remove non existing flags and add two internal methods
	  If someone cares flvparse could be merged into flvdemux.

2010-04-08 13:57:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpsession.h:
	  rtpsession: remove prototype for non existing function
	  There is no function by that name anywhere.

2010-04-08 12:56:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	  docs: Update inspected plugin information

2010-04-08 12:56:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: Improve docs a bit

2010-04-08 13:47:42 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: add effecttv defines and reorder list

2010-04-08 13:41:47 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: remove three entries that are not exported from the headers anymore

2010-04-08 13:40:36 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: move macro to c source
	  One less semi public symbol without namespace prefix in the headers.

2010-04-08 13:40:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/speex/gstspeexenc.h:
	  speexenc: remove unused defines

2010-04-08 13:23:38 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: fix last commit
	  Use a local define for WAVEFORMAT_EX based on the size of the struct + 2 bytes
	  for the extension size.

2010-04-08 13:16:53 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/speex/gstspeexdec.h:
	  speex: remove unused define

2010-04-08 13:03:43 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/wavenc/Makefile.am:
	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/riff.h:
	  wavenc: remove internal copy of riff.h and use riff-library instead.
	  We don't use any function yet, just the structures and defines.

2010-04-08 12:56:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: use riff lib more
	  Remove BITMAPINFOHEADER and use the one from riff-lib. Also remove the
	  WAVEFORMATEX_SIZE define and use a sizeof together with the respective struct.
	  Besides better code reuse this lessens the ununsed symbols in the docs.

2010-04-08 12:14:07 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  docs: trim sections file more
	  Rename some defines and move some itesm to *.c files. Add more items to internal
	  subsection.

2010-04-08 11:19:43 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docsw: trim the section file

2010-04-08 10:26:25 +0300  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: add v4l2sink to docs

2010-04-08 10:15:08 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/audiofx/audioamplify.c:
	* gst/multifile/gstmultifilesink.c:
	  docs: fix xml
	  The title tag belongs into the refsect2.

2010-04-07 17:43:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Add support for YV12, including conversion support for I420/AYUV

2010-04-07 17:27:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Add support for grayscale input/output
	  This doesn't do any conversion and is the next step to
	  replacing videocrop by supporting all remaining formats
	  in passthrough mode.

2010-04-07 16:24:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: Add support for filling the background with red, yellow and white

2010-04-07 16:11:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Add support for direct RGB<->AYUV conversion

2010-04-07 16:11:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix RGB24 filling

2010-04-07 16:06:54 +0300  Marco Ballesio <marco.ballesio@nokia.com>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: handle properly STAPs
	  in rtph264depay.c, lines 577-576, NALU-type 24 (Single-Time Aggregation
	  Packet) is handled in fall-through as NALU-type 26 (unhandled).
	  This leads high quality h264 streams such as:
	  rtsp://stream.yle.mobi/yle/areena/MEDIA_E0342657_p3.mp4
	  to fail with "NAL unit type 24 not supported yet" (but it's actually
	  supported), and thus to close any stream which contains STAPs.
	  The proposed one-liner patch fixes the issue.
	  Fixes #615051.

2010-04-07 13:47:02 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst-libs/gst/gst-i18n-plugin.h:
	* gst/avi/gstavi.c:
	  build: fix compiler warnings
	  fix warnings for all plugins that use: setlocale (LC_ALL...

2010-04-07 13:31:13 +0200  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/avi/gstavi.c:
	  avi: fix compiler warning

2010-03-31 17:54:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: restrict resyncing to subtitle tracks
	  This should prevent skipping audio or video in not so well interleaved
	  cases.
	  Fixes #614460.

2010-04-06 13:21:51 +0530  Arun Raghavan <ford_prefect@gentoo.org>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: Post avg./max. bitrate tags for H.264
	  This reads the average and maximum bitrates from the 'btrt' atom if
	  available, and pushes these as tags,
	  https://bugzilla.gnome.org/show_bug.cgi?id=614927

2010-04-03 23:39:20 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix racy shutdown
	  Keep a ref of pulsesink for deferred mainloop invocation. Fixes #614765

2010-04-05 15:48:17 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/jpegenc.c:
	  tests: jpegenc: Adds some getcaps test
	  Adds tests for the jpegenc getcaps function, to avoid
	  having it returning non-subset caps

2010-04-05 14:51:58 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Fix getcaps function
	  When creating the caps allowed to upstream using downstream
	  restrictions, use gst_pad_get_allowed_caps as that has the
	  usable formats and puts into it the width, height and framerate
	  fields. This avoids getting errors about getcaps returning
	  non subset caps of its pad template.
	  This error showed up on the metadata plugin unit test in -bad.

2010-04-05 17:31:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix conversion from 3 byte RGB to ARGB

2010-04-05 17:08:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Add support for 3 byte RGB formats and refactor RGB code a bit

2010-04-05 15:51:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: Add support for all 32 bit RGB formats
	  ...including conversion between them.

2010-04-05 15:26:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add property to control the buffering method
	  Add a property to control how the jitterbuffer performs timestamping and
	  buffering.

2010-04-04 19:02:41 -0300  André Dieb Martins <andre.dieb@gmail.com>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: Removing unused variable
	  Fixes bug #614843.

2010-04-04 20:31:38 -0300  André Dieb Martins <andre.dieb@gmail.com>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: should not return caps ANY based on downstream
	  When downstream has a sink pad with ANY caps, jpegenc should
	  treat it the same as NULL and return its template caps.
	  Fixes #614842

2010-04-04 22:28:33 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/oss/gstosshelper.c:
	  oss: add fixme comment

2010-04-04 22:26:59 +0300  Stefan Kost <ensonic@users.sf.net>

	* gconf/Makefile.am:
	  build: use $(builddir) for installing generated files

2010-04-04 22:07:33 +0300  Stefan Kost <ensonic@users.sf.net>

	* configure.ac:
	  Revert "configure: fix out of source dir builds"
	  This reverts commit ca0bd3a8cea31f9ea0df798a83d3007e696958ba.

2010-04-04 21:36:35 +0300  Stefan Kost <ensonic@users.sf.net>

	* configure.ac:
	  configure: fix out of source dir builds
	  Remove non-existing gst-libs from include and library-paths'.
	  Fixes #614354 even more.

2010-04-01 10:19:00 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: Read replaygain peak/gain tags
	  Make qtdemux read tags replaygain tags that are within '----' atoms.
	  Fixes #614471

2010-04-01 18:48:43 +0530  Arun Raghavan <ford_prefect@gentoo.org>

	* gst/matroska/matroska-demux.c:
	* gst/qtdemux/qtdemux.c:
	  matroska: Export h.264 profile and level in caps
	  This replicates the code in qtdemux to export the h.264 profile and
	  level in the stream caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=614651

2010-04-02 18:50:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix off-by-one introduced in last commit

2010-04-01 18:38:38 +0530  Arun Raghavan <ford_prefect@gentoo.org>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Minor refactor of the code
	  This will make it easier to clump together common code when copying to
	  mastroskademux.
	  https://bugzilla.gnome.org/show_bug.cgi?id=614651

2010-04-01 18:17:09 +0530  Arun Raghavan <ford_prefect@gentoo.org>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Export h.264 level in caps
	  This exports the h.264 level in the stream caps (as a string) which can
	  be used to match a decoder, or as metadata.
	  https://bugzilla.gnome.org/show_bug.cgi?id=614651

2010-04-01 16:58:32 +0530  Arun Raghavan <ford_prefect@gentoo.org>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Export h.264 profile in caps
	  This adds the h.264 profile for a given stream into caps. This can
	  (eventually) be used to select an appropriate decoder and as metadata
	  for certain applications.
	  https://bugzilla.gnome.org/show_bug.cgi?id=614651

2010-03-31 14:43:14 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: remove obsolete reverse playback code path

2010-03-31 14:40:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvparse.c:
	  flvdemux: support (pull mode) negative seek rate

2010-03-29 15:27:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: also check for segment stop for non-segment-seek

2010-03-30 16:50:10 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: push correctly sized flac header buffers
	  Fixes #614353.

2010-03-30 07:34:07 -0500  Rob Clark <rob@ti.com>

	* configure.ac:
	  build: fix compiler warning when srcdir != builddir
	  Fixes '../../gst-libs: No such file or directory' warning/error when
	  the build directory is not the same as the source directory.
	  Fixes #614354.

2010-03-30 01:50:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3v2frames.c:
	  id3demux: fix parsing of unsynced frames with data length indicator
	  Fixes bug #614158.

2010-03-29 11:00:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	* ext/Makefile.am:
	* gst/Makefile.am:
	* sys/Makefile.am:
	* tests/examples/Makefile.am:
	  build: build plugins and examples in parallel where possible

2010-03-18 18:49:24 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: fix redundant function redeclaration compiler warnings
	  Re-apply this again as well, as it was undone by the previous commit..

2010-03-18 14:31:35 +0100  Benjamin Otte <otte@redhat.com>

	* sys/directsound/gstdirectsoundsink.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple
	  Apply this again, as it was overwritten by the previous commit. Merging
	  is hard, apparently.

2010-03-26 23:20:10 +0100  Julien Moutte <julien@fluendo.com>

	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: Implement SPDIF support for AC3.
	  Detect if the sound card supports SPDIF passthru of AC3 and add
	  necessary code to support that like alsasink.

2010-03-26 17:06:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	  build: add cruft alert for common/shave*

2010-03-26 16:50:22 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/Makefile.am:
	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_lang.c:
	* gst/qtdemux/qtdemux_lang.h:
	  qtdemux: extract stream language in more cases
	  The 16-bit language code can be either a packed ISO-639-2T code
	  or a 'Macintosh language code'. Handle the latter type of language
	  codes as well, and map to the matching ISO code. Lastly, fix
	  language code posting for language #0, which is valid and stands
	  for 'English'.
	  Fixes #614001.

2010-03-26 14:55:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: Improve debugging and add some FIXMEs

2010-03-26 14:42:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: Sample rate markers 0x01, 0x02 and 0x03 are valid
	  They are for 88.2kHz, 176.4kHz and 192kHz.

2010-03-26 14:16:39 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: Take samplerate, width and number of channels from the STREAMINFO
	  ...and update it from the frame headers if it should change for some reason.
	  This allows playback of files with odd sample rates.

2010-03-26 13:45:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix AYUV->I420 frame copying

2010-03-26 13:34:17 +0100  Raimo Järvi <raimo.jarvi@gmail.com>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Set correct getcaps/setcaps functions on srcpads and simplify them
	  This fixes downstream negotiation, upstream negotiation isn't really
	  supported by jpegenc yet.
	  Fixes bug #613789.

2010-03-26 10:31:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: Always fill the complete frame if borders should be added
	  This makes sure that we don't get any gaps between rectangles because
	  of chroma subsampling for example.

2010-03-18 22:12:40 +0000  Damien Lespiau <damien.lespiau@intel.com>

	* autogen.sh:
	  autogen.sh: Don't call configure with --enable-plugin-docs
	  configure gives a nice warning:
	  configure: WARNING: unrecognized options: --enable-plugin-docs
	  and indeed, I could not find anything in the configure.ac or the m4
	  macros that would allow enabling that option. Remove it then.

2010-03-22 16:58:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: Refactor boxing to reduce code duplication

2010-03-22 13:13:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Simplify caps transformation

2010-03-21 20:14:19 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Add const qualifier to the source frame data

2010-03-23 17:47:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: only seek when in proper state
	  ... and data structures can be thread-safely accessed.
	  See #601617.

2010-03-23 17:34:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-ids.h:
	  matroskademux: support (pull mode) negative seek rate

2010-03-18 15:29:00 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: track clip duration in segment

2010-03-18 13:39:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: prefer index of video track to perform seeking

2010-03-25 22:58:47 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/dtmf/gstdtmfdetect.c:
	  dtmfdetect: if we tell that we handle gap flags, then do so

2010-03-25 22:55:32 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/dtmf/gstdtmfdetect.c:
	  dtmfdetect: use glib types

2010-03-25 22:54:49 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/dtmf/gstdtmfdetect.c:
	  dtmfdetect: fix classification

2010-03-25 22:53:20 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/dtmf/gstdtmfdetect.c:
	  dtmfdetect: reformat message docs
	  Use a list like in other element docs as an untweaked docbook table look ugly.

2010-03-24 16:19:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix typo in header validation check

2010-03-24 18:53:20 +0100  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From 55cd514 to c1d07dd

2010-03-24 11:27:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlame.h:
	* ext/lame/gstlamemp3enc.h:
	* ext/lame/plugin.c:
	  build: Add all kinds of compiler warning flags and fix the resulting warnings

2010-03-23 19:46:43 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/icydemux/gsticydemux.c:
	* gst/icydemux/gsticydemux.h:
	  icydemux: Handle upstream Content-Type.
	  Allows us to handle ShoutCast TV (NSV) streams.
	  If the upstream caps have the 'content-type' field set to video/nsv, then
	  we shortcut the typefinding and set video/x-nsv directly.

2010-03-23 19:30:50 +0100  Edward Hervey <bilboed@bilboed.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Set the Content-Type HTTP header on the caps.
	  First step to fixing ShoutCast (NSV) streaming.

2010-03-23 02:38:43 -0400  Tristan Matthews <tristan@sat.qc.ca>

	* sys/osxaudio/gstosxaudioelement.c:
	* sys/osxvideo/Makefile.am:
	  osx: fix compiler warnings
	  Added void parameter to avoid old-style definition warning.
	  Added -Wno-aggregate-return flag to avoid erroneous aggregate return warning.
	  https://bugzilla.gnome.org/show_bug.cgi?id=613663

2010-03-23 00:15:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/videocrop.c:
	  tests: use loop test for long-running videocrop check
	  This should avoid timeouts on slow machines.
	  Fixes #597739.

2010-03-22 17:26:37 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/flac/gstflac.c:
	* ext/pulse/plugin.c:
	* ext/wavpack/gstwavpack.c:
	* gst-libs/gst/gettext.h:
	* gst/multifile/gstmultifilesink.h:
	  i18n: build fixes: #if -> #ifdef for ENABLE_NLS

2010-03-22 17:25:09 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst-libs/gst/gst-i18n-plugin.h:
	  i18n: fix the build
	  Don't inlcude locale.h which we include in gettext.h if needed. Guard the
	  inlcude like we do in the simillar headers in core.

2010-03-22 13:16:33 +0100  Benjamin Otte <otte@redhat.com>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	  Add -Wwrite-strings
	  and fix its warnings

2010-03-22 12:02:16 +0100  Benjamin Otte <otte@redhat.com>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  Add -Wredundant-decls flag
	  and fix warnings from it

2010-03-21 21:39:18 +0100  Benjamin Otte <otte@redhat.com>

	* gst/dtmf/gstrtpdtmfdepay.h:
	  Add -Wmissing-declarations -Wmissing-prototypes to configure flags
	  And fix all warnings

2010-03-21 17:46:06 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	  -Wold-style-definition is not valid for C++

2010-03-21 17:36:28 +0100  Benjamin Otte <otte@redhat.com>

	* gst/multifile/gstmultifile.c:
	  multifile: Include headers instead fo defining functions

2010-03-21 17:24:14 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	  Add a large set of warning flags.
	  None of them trigger warnings anymore, so nothing needed to be fixed.

2010-03-21 17:23:43 +0100  Benjamin Otte <otte@redhat.com>

	* gst/goom/config_param.c:
	* gst/goom/convolve_fx.c:
	* gst/goom/filters.c:
	* gst/goom/flying_stars_fx.c:
	* gst/goom/goom_config_param.h:
	* gst/goom/goom_core.c:
	* gst/goom/goom_filters.h:
	* gst/goom/goom_fx.h:
	* gst/goom/ifs.c:
	* gst/goom/ifs.h:
	* gst/goom/plugin_info.c:
	* gst/goom/tentacle3d.c:
	* gst/goom/tentacle3d.h:
	  Make goom not use aggregate returns

2010-03-21 15:17:46 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	* ext/annodex/gstcmmlutils.c:
	* ext/wavpack/gstwavpackparse.c:
	* gst/effectv/gstwarp.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/udp/gstmultiudpsink.c:
	* tests/check/elements/cmmldec.c:
	* tests/check/elements/cmmlenc.c:
	* tests/check/elements/deinterlace.c:
	* tests/check/elements/rglimiter.c:
	* tests/check/elements/rtp-payloading.c:
	* tests/check/elements/udpsink.c:
	* tests/check/elements/videofilter.c:
	* tests/check/elements/wavpackdec.c:
	* tests/check/generic/states.c:
	* tests/icles/v4l2src-test.c:
	  Add -Wold-style-definition flag
	  And fix the warnings

2010-03-20 00:54:14 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	* ext/hal/hal.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/wavpack/gstwavpackcommon.c:
	* gst/avi/gstavimux.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/flv/gstflvparse.c:
	* gst/goom/config_param.c:
	* gst/goom/goom_config_param.h:
	* gst/id3demux/id3tags.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/qtdemux/qtdemux.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videofilter/gstvideobalance.c:
	* sys/oss/gstossmixertrack.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/level.c:
	* tests/check/elements/rtpbin_buffer_list.c:
	* tests/check/pipelines/simple-launch-lines.c:
	  Add -Wwrite-strings to the configure flags
	  ... and fix all warnings

2010-03-21 11:14:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  shapewipe: Add support for the remaining ARGB formats
	  And handle AYUV like ARGB, we need no YUV specific handling.

2010-03-20 21:30:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Add support for RGB and xRGB input

2010-03-20 21:13:23 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Add support for ARGB input

2010-03-20 20:46:19 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Add support for generating ARGB output

2010-03-20 10:47:42 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/blend_mmx.h:
	* gst/videomixer/videomixer.c:
	  videomixer: Add support for ABGR and RGBA
	  Now all 4 ARGB variants are supported by videomixer.

2010-03-20 10:24:56 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Move chroma keying parameters into stack variables to prevent multiple pointer dereferences per pixel

2010-03-20 10:20:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Move color conversion matrixes into stack variables to speed up processing

2010-03-20 10:18:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Use correct matrixes to convert chroma keying color to YUV

2010-03-19 18:51:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Add support for different color matrixes

2010-03-19 18:21:19 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Rename and move functions as further preparation for supporting more color formats

2010-03-19 18:18:08 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: Remove some unneeded calculations and instance struct fields
	  And document the instance struct fields a bit better

2010-03-19 18:11:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: Some preparations for supporting more color formats

2010-03-19 17:09:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  h264pay: fix config-interval property
	  Use the same units for comparing the elapsed time against the interval.
	  Fixes #613013

2010-03-19 16:44:00 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  alphacolor: Implement color-matrix support and use integer arithmetic only
	  Alphacolor now uses the correct matrixes for SDTV and HDTV and can
	  convert between them.

2010-03-19 15:03:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* gst/rtsp/gstrtspsrc.c:
	  rtsp: use GType from -base and bump required version
	  Use the transport flags GType from -base and bump the required version of -base
	  because of this.

2010-03-19 00:05:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/apetag/Makefile.am:
	  apetag: minor Makefile.am surgery
	  -I$(top_srcdir)/gst-libs/ is already in $(GST_CFLAGS)

2010-03-18 17:30:26 +0100  Benjamin Otte <otte@redhat.com>

	* gst/audiofx/gststereo.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple

2010-03-18 17:30:26 +0100  Benjamin Otte <otte@redhat.com>

	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple

2010-03-04 22:12:35 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/raw1394/gst1394clock.c:
	  dv1394src: Fix internal clock
	  Fixes #593910.

2010-03-18 21:14:17 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/dv/Makefile.am:
	* ext/esd/Makefile.am:
	* ext/libcaca/Makefile.am:
	* ext/pulse/Makefile.am:
	* ext/shout2/Makefile.am:
	* ext/speex/Makefile.am:
	* ext/wavpack/Makefile.am:
	* gst/auparse/Makefile.am:
	* gst/avi/Makefile.am:
	* gst/flx/Makefile.am:
	* gst/icydemux/Makefile.am:
	* gst/interleave/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/qtdemux/Makefile.am:
	* gst/replaygain/Makefile.am:
	* gst/rtp/Makefile.am:
	* gst/udp/Makefile.am:
	* gst/videomixer/Makefile.am:
	* gst/wavparse/Makefile.am:
	* sys/directsound/Makefile.am:
	* sys/oss/Makefile.am:
	* sys/waveform/Makefile.am:
	* tests/examples/v4l2/Makefile.am:
	  build: Makefile.am cleanups
	  Mostly add $(GST_BASE_CFLAGS) where it was missing, but also fix up
	  order of flags and libs if needed (see docs/random/moving-plugins).

2010-03-18 18:49:24 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: fix redundant function redeclaration compiler warnings

2010-03-18 19:00:09 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: Remove remaining floating point arithmetic when processing a pixel

2010-03-18 18:55:34 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Refactor chroma keying into a single function
	  This reduces code duplication once we add support for more color formats.

2010-03-18 15:53:14 +0100  Benjamin Otte <otte@redhat.com>

	* ext/lame/gstlame.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple

2010-03-18 14:31:35 +0100  Benjamin Otte <otte@redhat.com>

	* ext/aalib/gstaasink.c:
	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttimeoverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/esd/esdmon.c:
	* ext/esd/esdsink.c:
	* ext/gconf/gstgconfaudiosink.c:
	* ext/gconf/gstgconfaudiosrc.c:
	* ext/gconf/gstgconfvideosink.c:
	* ext/gconf/gstgconfvideosrc.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/hal/gsthalaudiosink.c:
	* ext/hal/gsthalaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libmng/gstmng.h:
	* ext/libmng/gstmngdec.c:
	* ext/libmng/gstmngenc.c:
	* ext/libpng/gstpng.h:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/mikmod/gstmikmod.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/efence.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/negotiation.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/testplugin.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/level/gstlevel.c:
	* gst/median/gstmedian.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/qtdemux/gstrtpxqtdepay.c:
	* gst/qtdemux/qtdemux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspgoogle.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* gst/y4m/gsty4mencode.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstossmixerelement.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	  gst_element_class_set_details => gst_element_class_set_details_simple

2010-03-18 14:02:30 +0100  Benjamin Otte <otte@redhat.com>

	* gst/oldcore/Makefile.am:
	* gst/oldcore/gstaggregator.c:
	* gst/oldcore/gstaggregator.h:
	* gst/oldcore/gstelements.c:
	* gst/oldcore/gstfdsink.c:
	* gst/oldcore/gstfdsink.h:
	* gst/oldcore/gstmd5sink.c:
	* gst/oldcore/gstmd5sink.h:
	* gst/oldcore/gstmultifilesrc.c:
	* gst/oldcore/gstmultifilesrc.h:
	* gst/oldcore/gstpipefilter.c:
	* gst/oldcore/gstpipefilter.h:
	* gst/oldcore/gstshaper.c:
	* gst/oldcore/gstshaper.h:
	* gst/oldcore/gststatistics.c:
	* gst/oldcore/gststatistics.h:
	  Remove oldcore directory
	  The elements have been unused for ages and all important ones have been
	  replaced or copied elsewhere.

2010-03-18 13:45:08 +0100  Benjamin Otte <otte@redhat.com>

	* gst/avi/gstavidecoder.c:
	  avi: Remove old file
	  Seems to be leftover from the 0.4 days or so.

2010-03-18 12:44:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	  pulse: use #ifdef rather than #if conditionals

2010-03-18 12:20:17 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: do not call _push_ts with unneeded (and wrong) time parameter
	  Fixes #613206.

2010-03-18 11:33:59 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix typo in header validation check

2010-03-18 01:51:19 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: put more information in the metadata
	  Additional tags are: audiocodecid, videocodecid framerate and (in the
	  non-live case) filesize.
	  While at it, fix index rewriting to update duration and filesize
	  values even if the index is empty.
	  Fixes #613094.

2010-03-17 21:33:28 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	* ext/jpeg/gstjpegenc.c:
	* ext/speex/gstspeexenc.h:
	* gst/goom/goom_config.h:
	* gst/goom/mathtools.h:
	* tests/check/elements/level.c:
	  Add -Wundef to configure flags
	  and fix the resulting warnings

2010-03-17 20:02:16 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	  -Wmissing-prototypes is not valid for C++

2010-03-17 19:35:10 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	* ext/flac/gstflacdec.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/jpeg/gstjpeg.h:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/wavpack/gstwavpackdec.c:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/equalizer/gstiirequalizer.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/videomixer/videomixer.c:
	* sys/v4l2/v4l2src_calls.c:
	  Add -Wredundant-decls warning flag
	  Also fix compile issues

2010-03-17 18:49:11 +0100  Benjamin Otte <otte@redhat.com>

	* gst/monoscope/gstmonoscope.h:
	  Fix warnings in experimental plugins, too

2010-03-17 18:23:00 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	* ext/annodex/gstannodex.c:
	* ext/annodex/gstcmmldec.h:
	* ext/annodex/gstcmmlenc.h:
	* ext/annodex/gstcmmlparser.c:
	* ext/annodex/gstcmmlutils.c:
	* ext/dv/gstdvdec.c:
	* ext/flac/gstflacenc.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/pixbufscale.h:
	* ext/jpeg/Makefile.am:
	* ext/jpeg/gstjpeg.c:
	* ext/jpeg/gstjpeg.h:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/wavpack/gstwavpackstreamreader.c:
	* ext/wavpack/gstwavpackstreamreader.h:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	* gst/deinterlace/tvtime/greedyh.asm:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/mmx.h:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
	* gst/goom/goom_fx.h:
	* gst/goom2k1/filters.c:
	* gst/goom2k1/filters.h:
	* gst/law/mulaw-conversion.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/multipart/multipart.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	* gst/multipart/multipartmux.c:
	* gst/multipart/multipartmux.h:
	* gst/qtdemux/gstrtpxqtdepay.c:
	* gst/rtp/fnv1hash.c:
	* gst/rtp/fnv1hash.h:
	* gst/rtp/gstasteriskh263.h:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpac3depay.h:
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpbvdepay.h:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpbvpay.h:
	* gst/rtp/gstrtpceltdepay.h:
	* gst/rtp/gstrtpceltpay.h:
	* gst/rtp/gstrtpdvdepay.h:
	* gst/rtp/gstrtpdvpay.h:
	* gst/rtp/gstrtpg723depay.h:
	* gst/rtp/gstrtpg723pay.h:
	* gst/rtp/gstrtpg726depay.h:
	* gst/rtp/gstrtpg726pay.h:
	* gst/rtp/gstrtpg729depay.h:
	* gst/rtp/gstrtpg729pay.h:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtph263depay.h:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtph264pay.h:
	* gst/rtp/gstrtpilbcdepay.h:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpilbcpay.h:
	* gst/rtp/gstrtpj2kdepay.h:
	* gst/rtp/gstrtpj2kpay.h:
	* gst/rtp/gstrtpjpegdepay.h:
	* gst/rtp/gstrtpjpegpay.h:
	* gst/rtp/gstrtpmp1sdepay.h:
	* gst/rtp/gstrtpmp2tdepay.h:
	* gst/rtp/gstrtpmp2tpay.h:
	* gst/rtp/gstrtpmp4adepay.h:
	* gst/rtp/gstrtpmp4apay.h:
	* gst/rtp/gstrtpmp4gdepay.h:
	* gst/rtp/gstrtpmp4gpay.h:
	* gst/rtp/gstrtpmp4vdepay.h:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadepay.h:
	* gst/rtp/gstrtpmpapay.h:
	* gst/rtp/gstrtpmpvdepay.h:
	* gst/rtp/gstrtpmpvpay.h:
	* gst/rtp/gstrtppcmadepay.h:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmudepay.h:
	* gst/rtp/gstrtppcmupay.h:
	* gst/rtp/gstrtpqdmdepay.h:
	* gst/rtp/gstrtpsirendepay.h:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpsirenpay.h:
	* gst/rtp/gstrtpspeexdepay.h:
	* gst/rtp/gstrtpspeexpay.h:
	* gst/rtp/gstrtpsv3vdepay.h:
	* gst/rtp/gstrtptheoradepay.h:
	* gst/rtp/gstrtptheorapay.h:
	* gst/rtp/gstrtpvorbisdepay.h:
	* gst/rtp/gstrtpvorbispay.h:
	* gst/rtp/gstrtpvrawdepay.h:
	* gst/rtp/gstrtpvrawpay.h:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstmask.c:
	* gst/smpte/gstmask.h:
	* gst/videobox/gstvideobox.h:
	* gst/videocrop/gstvideocrop.h:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	* gst/wavenc/gstwavenc.h:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/gstv4l2vidorient.h:
	* sys/ximage/ximageutil.c:
	* tests/check/elements/aspectratiocrop.c:
	* tests/check/elements/audioamplify.c:
	* tests/check/elements/audiochebband.c:
	* tests/check/elements/audiocheblimit.c:
	* tests/check/elements/audiodynamic.c:
	* tests/check/elements/audioecho.c:
	* tests/check/elements/audioinvert.c:
	* tests/check/elements/audiopanorama.c:
	* tests/check/elements/audiowsincband.c:
	* tests/check/elements/audiowsinclimit.c:
	* tests/check/elements/avimux.c:
	* tests/check/elements/avisubtitle.c:
	* tests/check/elements/cmmldec.c:
	* tests/check/elements/equalizer.c:
	* tests/check/elements/level.c:
	* tests/check/elements/matroskamux.c:
	* tests/check/elements/multifile.c:
	* tests/check/elements/rganalysis.c:
	* tests/check/elements/rglimiter.c:
	* tests/check/elements/rgvolume.c:
	* tests/check/elements/shapewipe.c:
	* tests/check/elements/souphttpsrc.c:
	* tests/check/elements/spectrum.c:
	* tests/check/elements/videofilter.c:
	* tests/check/elements/wavpackdec.c:
	* tests/check/elements/wavpackenc.c:
	* tests/check/elements/wavpackparse.c:
	* tests/check/elements/y4menc.c:
	* tests/check/generic/states.c:
	* tests/check/pipelines/simple-launch-lines.c:
	* tests/check/pipelines/wavpack.c:
	* tests/examples/equalizer/demo.c:
	* tests/examples/level/level-example.c:
	* tests/examples/spectrum/spectrum-example.c:
	* tests/icles/v4l2src-test.c:
	  Add -Wmissing-declarations -Wmissing-prototypes warning flags
	  And fix all the warnings.

2010-03-17 16:23:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  mp4gdepay: improve constantDuration guessing
	  When no constantDuration has been given in the caps, try to derive one from the
	  timestamp difference between packets. Also keep doing this for each packet
	  because some broken streams might simply provide wrong timestamps.

2010-03-16 23:43:39 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: Put width and height in the metadata
	  Some players use that info to scale their display.
	  See #613094.

2010-03-16 23:32:45 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: don't put timestamps larger than G_MAXINT32 in the FLV tags
	  For non-live input respond by pushing EOS, for live wrap the
	  timestamps every G_MAXINT32 miliseconds.
	  Fixes #613003.

2010-03-16 23:40:12 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/soup/gstsouphttpsrc.c:
	  soup: also use g_value_set_static_string() here for static strings

2010-03-16 21:23:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: Fix RGBA<->AYUV conversion

2010-03-16 21:16:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: Remove redundant instance field

2010-03-16 21:10:08 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: Protect property values from changes during frame processing

2010-03-15 23:29:55 +0300  Руслан Ижбулатов <lrn1986@gmail.com>

	* ext/libpng/gstpngdec.c:
	  pngenc: Use png_get_io_ptr() instead of accessing io_ptr directly
	  Fixes #612700 (for the last time!)

2010-03-15 23:29:06 +0300  Руслан Ижбулатов <lrn1986@gmail.com>

	* configure.ac:
	  png: Check for libpng >= 1.2 instead of libpng12

2010-03-16 01:29:36 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Always put a duration tag in the metadata
	  Some Flash players (for instance JW Player) always expect a duration
	  tag, otherwise they don't start playback.
	  If duration can be queried from the sink pads or is provided as a tag,
	  use it. Otherwise try to determine it from the last seen timestamp of
	  the sink pads after EOS and rewrite it in the header before writing
	  the index.

2010-03-16 00:35:46 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Remove the send_codec_data field from GstFlvPad
	  That field is not used anymore after the changes in
	  9fdecbc1c11f4e5af6578bba32a9b32771029d33.

2010-03-16 13:53:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: get family of external sockets too
	  Get the family of externally configured sockets so that we can configure it
	  correctly.

2010-03-15 20:37:51 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: Add support for the remaining ARGB formats

2010-03-15 19:16:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: Simplify ARGB<->AYUV conversions by code generation macros

2010-03-15 19:07:28 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* gst/alpha/Makefile.am:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: Minor cleanups and move declarations into a separate header file

2010-03-15 18:58:51 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/Makefile.am:
	* gst/alpha/gstalpha.c:
	  alpha: Use GstVideoFilter as base class for automatic QoS support

2010-03-15 18:50:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  alphacolor: Add support for inplace conversions from AYUV to ARGB

2010-03-15 18:14:19 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  alphacolor: Use libgstvideo for caps parsing

2010-03-15 18:09:55 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/Makefile.am:
	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  alphacolor: Use GstVideoFilter as base class for automatic QoS support

2010-03-15 18:07:29 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: Some minor cleanup

2010-03-15 14:16:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexdec.h:
	  speexdec: Use speex_stereo_state_init() instead of the deprecated initialization macro
	  Fixes bug #612777.

2010-03-15 01:09:49 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: Correctly mark buffers as delta units
	  Mark video interframes, video codec data buffers and audio buffers (if
	  it's not an audio-only stream) as delta units.

2010-03-14 19:32:20 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: Support streamheaders
	  Put the FLV header, the metadata tag and (if present) codec
	  information in the streamheader to allow the muxer to be used for
	  streaming.

2010-03-14 01:38:21 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: Preallocate index space and fill it after finishing output
	  Make the index appear at the beginning of the file, which is what most
	  players are expecting.
	  Fixes #601236.

2010-03-15 13:47:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Minor coding style fixes and cleanup

2010-03-14 01:34:02 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Add a is-live property
	  If it is set, the muxer will not write the index. Defaults to false.

2010-03-14 01:25:42 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: Only put valid seek points in the index
	  For files containing video only video keyframes are valid points to
	  which a player can seek. For audio-only files any tag start is a valid
	  seek point.
	  See #601236.

2010-03-14 01:09:37 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix index building to make entries point to tag's start offset
	  Previous coding was wrongly incrementing the total byte count before
	  adding an index entry.

2010-03-15 13:40:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	  cairotextoverlay: Don't render text outside the frame boundaries
	  Fixes bug #611986.

2010-03-15 11:38:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't forget to send keepalive messages
	  When we operate in TCP mode, still send keepalive messages when we
	  need to.
	  Fixes #612696

2010-03-13 23:19:35 +0300  Руслан Ижбулатов <lrn1986@gmail.com>

	* ext/libpng/gstpngenc.c:
	  pngenc: Call png_jmpbuf() instead of accessing png_struct_ptr directly
	  Fixes #612700 (again)

2010-03-12 16:44:30 +0300  Руслан Ижбулатов <lrn1986@gmail.com>

	* ext/libpng/gstpngenc.c:
	  pngenc: Call png_error() instead of using longjmp() directly.
	  Fixes #612700

2010-03-12 13:57:28 +0100  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From e272f71 to 55cd514

2010-03-05 11:06:47 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: add XMP parsing support
	  Use xmp helpers to parse XMP metadata in udta atom.
	  Fixes #609539

2010-03-11 12:32:56 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudpnetutils.c:
	* gst/udp/gstudpnetutils.h:
	  udp: fix compilation errors on non-windows.

2010-03-10 22:23:43 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudpnetutils.c:
	* gst/udp/gstudpnetutils.h:
	  multiudpsink: avoid getting the socket family using getsockname()

2010-03-11 17:28:47 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix print statements for pointer differences.
	  This fixes it for both 32 and 64 bit

2010-03-11 17:28:35 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix unitialized variables

2010-03-11 17:03:47 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Fix printf formatting for macosx

2010-03-11 17:03:05 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Fix unitialized variables

2010-03-11 17:02:44 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix unitialized variable.

2010-02-19 13:39:04 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvparse.c:
	  flvparse: Make script tag parsing more flexible.
	  * The nb_elements for arrays is just an indication, we can therefore ignore
	  it and carry on parsing metadata items until we reach the end marker.
	  * If type == 3, then the script tag contains a list of object followed
	  by the end marker.
	  Refactor code slightly to handle both cases
	  https://bugzilla.gnome.org/show_bug.cgi?id=610447

2010-03-11 15:51:40 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/deinterleave.c:
	* tests/check/elements/interleave.c:
	  tests: fix metadata not writable warnings in interleave and deinterleave tests

2010-03-11 15:38:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/apev2mux.c:
	* tests/check/elements/id3v2mux.c:
	  tests: fix metadata not writable warnings with apev2mux and id3v2mux tests

2010-03-11 15:24:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix metadata writable warnings
	  Set metadata on buffer first, when the refcount is still 1, and only
	  ref again afterwards.

2010-03-11 15:02:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: ignore stream with invalid header time metadata

2010-03-08 14:57:17 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Set stream-format=raw on AAC caps
	  Set stream-format=raw for AAC caps, as that is the
	  expected AAC format to be in this container family.
	  Fixes #566250

2010-03-11 12:56:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: check for NULL before doing strcmp
	  Check the connection and address type for NULL before doing strcmp and
	  crashing.
	  Fixes #612553

2010-03-11 11:20:59 +0100  Benjamin Otte <otte@redhat.com>

	* common:
	  Automatic update of common submodule
	  From df8a7c8 to e272f71

2010-03-11 11:09:55 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/udp/gstudpnetutils.c:
	  build: include stdlib.h for atoi()

2010-03-11 10:33:00 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: move invariant check out of the inner loop
	  Improves performance for simple method.

2010-03-10 22:15:04 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	  Update CXXFLAGS, too, just like CFLAGS

2010-03-10 21:01:20 +0100  Benjamin Otte <otte@redhat.com>

	* configure.ac:
	* gst/rtpmanager/Makefile.am:
	* tests/check/Makefile.am:
	  Update for recent changes to common submodule
	  This just replaces every "$ERROR_CFLAGS" usage with a usage of
	  "$WARNING_CFLAGS $ERROR_CFLAGS" to get the same functionality as
	  previously.
	  Actually using that separation will happen later.

2010-03-10 21:52:09 +0100  Benjamin Otte <otte@redhat.com>

	* common:
	  Automatic update of common submodule
	  From 9720a7d to df8a7c8

2010-03-10 20:43:57 +0100  Benjamin Otte <otte@redhat.com>

	* common:
	  Automatic update of common submodule
	  From 0b6e072 to 9720a7d

2010-03-10 10:51:28 -0800  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Reset windows error code after getting corresponding error message.

2010-03-09 17:32:27 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avimux: put the codec_data blob into the actual data for MPEG4 video, to match other implementations in the wild.

2010-03-10 16:09:56 +0100  Benjamin Otte <otte@redhat.com>

	* common:
	  Automatic update of common submodule
	  From 7cc5eb4 to 0b6e072

2010-02-23 21:06:55 -0300  Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: send new_segment with GST_FORMAT_TIME format
	  Instead of using BaseSrc default format GST_FORMAT_BYTES, send it in
	  GST_FORMAT_TIME.
	  Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
	  Fixes #611659

2010-03-10 11:46:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: push mode; also report seekable without an element index
	  ... since recent code also seeks around to obtain required data
	  from avi index.

2010-03-09 18:06:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: add some check and standardized seek event handling in push mode

2010-03-09 18:05:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix offset handling in push mode seeking
	  Push mode seeking uses same index data as pull mode, and stores
	  offset to data in chunk, whereas push mode operates in chunks,
	  and as such needs offset consistently corresponding to chunk headers.
	  Also fix determining best matching stream for incoming newsegment event,
	  as well as setting some stream state accordingly.

2010-02-26 21:29:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: conduct index scan in task thread
	  ... rather than in seeking thread, which might then occupy mainloop
	  for some time with possible unresponsive side-effects.

2010-02-26 21:27:33 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  flvdemux: avoid indefinite index growth
	  That is, check for and do not add an index entry that has already
	  been added.

2010-02-18 14:57:39 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  flvdemux: also collect index info on-the-fly in pull mode

2010-02-18 12:42:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvparse.c:
	* gst/flv/gstflvparse.h:
	  flvdemux: incrementally build index in pull mode
	  Scan for needed part upon a seek as opposed to doing a complete scan
	  at startup, which may take some time depending on file and/or platform.
	  Also accept index metadata in pull mode and peek for some metadata
	  at the end of the file when deemed appropriate.

2010-02-18 12:26:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: some more variable cleanup

2010-03-09 18:25:23 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  flvdemux: refactor adding index entry

2010-02-17 11:36:13 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  flvdemux: fix setting DELTA_UNIT flag on outgoing buffers
	  ... which should not depend on having index available or not.
	  Also refactor resulting collapsed code.

2010-02-11 19:43:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: avoid erroneous codec-data overriding of stsd information

2010-02-01 22:37:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: adapt to new oggdemux
	  Remove all granulepos hacks and simply use upstream timestamps.

2010-02-01 22:36:02 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexdec.h:
	  speexdec: refactor granulepos hacks

2010-03-10 11:19:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: parse connection information
	  Parse the connection information from the SDP and use it to figure out if we are
	  dealing with ipv4 or ipv6 connections.

2010-03-09 17:53:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: require a destination for multicast
	  When setting up the multicast sockets, we need a destination address to listen
	  on or else we error.

2010-03-09 17:52:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: handle ipv6 listening ports when needed
	  Add some code to make udpsrc listen on an ipv6 address when needed. The
	  detection of IPV6 is not yet implemented.

2010-03-09 17:15:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsink.h:
	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udp: use uri parsing code
	  Use the uri parsing helper functions to manage the host and port pairs. This
	  adds support for IPV6.

2010-03-09 17:13:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpnetutils.c:
	* gst/udp/gstudpnetutils.h:
	  udpnetutils: add helper functions for udp uri handling
	  Add some helpers to parse udp uris. Make sure IPV6 is supported too.

2010-03-05 16:08:45 +0100  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: Make it possible to favor new sources in case of SSRC conflict
	  Add a "favor-new" property that tells the session to favor new sources when
	  there is a SSRC conflict. This is useful for SIP calls and other such cases
	  where a remote loop is extremely unlikely.
	  Fixes #607615

2010-03-05 15:46:48 +0100  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: Move SSRC conflicts lists into RTPSource
	  We will also need to track SSRC conflicts in remote sources.
	  See #607615

2010-02-26 17:13:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: send keep alive when paused
	  When we are paused, send keep alive messages to the server so that our session
	  doesn't time out when we go back to playing later.

2010-03-10 01:10:07 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 7aa65b5 to 7cc5eb4

2010-02-23 19:48:10 -0800  David Schleef <ds@schleef.org>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: Add key-frame option to next-file
	  This allows segmenting of MPEG-TS files at key frames, which is
	  exactly what is needed for Apple's HTTP streaming.

2010-03-09 21:32:47 +0000  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 44ecce7 to 7aa65b5

2010-03-08 20:17:58 +0000  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix autocropping for odd width/height differences

2010-03-08 20:02:19 +0000  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/Makefile.am:
	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: Use libgstvideo for format specific stuff

2010-03-08 19:28:47 +0000  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbaseiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	  audiofx: Sync properties to the stream time

2010-03-08 19:20:59 +0000  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/Makefile.am:
	* gst/videobox/gstvideobox.c:
	  videobox: Make properties controllable

2010-03-08 19:09:01 +0000  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Some cleanup

2010-02-28 15:47:50 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	  effectv: Use controller where possible, optimize a bit and make properties threadsafe

2010-02-26 16:35:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* pkgconfig/Makefile.am:
	  build: Make some more rules silent if requested

2010-02-26 15:41:52 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Use automake 1.11 silent rules instead of shave if available
	  This makes sure that we use something that is still maintained and
	  also brings back libtool 1.5 support.

2010-03-08 22:57:34 +0100  Benjamin Otte <otte@redhat.com>

	* ext/libpng/gstpngenc.c:
	  png: fractions don't allow doubles

2010-03-01 12:03:56 +0100  Benjamin Otte <otte@redhat.com>

	* gst/flx/gstflxdec.c:
	  flx: fix description
	  It's video, not audio

2010-03-09 17:45:27 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* win32/common/config.h:
	  Back to development

=== release 0.10.21 ===

2010-03-09 00:28:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.21

2010-03-09 00:24:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2010-03-09 00:09:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  Revert "Add 4:2:2, 4:1:1, and 4:4:4 output support"
	  This reverts commit 637c26f61a2bd8d7b01f8b6d081d94da65f74557.

=== release 0.10.20 ===

2010-03-08 23:42:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.20

2010-03-08 23:42:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2010-03-08 16:47:04 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: don't send second newsegment event in framed mode, fixes long playback delay
	  Don't send another newsegment event if the upstream muxer/parser has already
	  sent one (otherwise the sink will wait for $duration before starting playback).
	  Fixes long delay until playback starts with flac-in-ogg files.
	  Fixes #610959.

2010-03-05 13:49:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: configure multicast correctly
	  Take the transport destination for multicast.
	  Disable loop and autojoin for multicast on the udpsinks.

2010-03-05 13:47:33 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  multicast: always configure loop and ttl
	  Also configure TTL and loop parameters when we add a client after initializing
	  the sender.

2010-03-08 12:13:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  Revert "rtph263depay: baseclass handles timestamps for us"
	  This reverts commit 564581e1b88ecd5ec5da82c3cafb0e7a2d58b302.
	  If we don't call push_ts, there will be no timestamp at all on the outgoing
	  buffer.
	  Fixes #612154

2010-02-23 22:16:39 -0500  Benjamin M. Schwartz <bens@alum.mit.edu>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  Add 4:2:2, 4:1:1, and 4:4:4 output support

2010-03-02 13:21:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: use payload size to estimate bitrate
	  Use the length of the payload for estimating the receiver bitrate so that it
	  matches the calculations done on the sender side. Together with the number of
	  packets one can scale the bitrate with the header overhead of the lower
	  transport.

2010-03-02 12:39:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsource: refactor bitrate estimation
	  Don't reuse the same variable we need for stats for the bitrate estimation
	  because we're updating it.
	  Refactor the bitrate estimation code so that both sender and receivers use the
	  same code path.

2010-03-01 16:40:27 -0500  Tristan Matthews <tristan@sat.qc.ca>

	* gst/rtpmanager/rtpsource.c:
	  added bitrate estimation to receiver-side stats, fixes #611213

2010-03-01 16:01:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263pay.c:
	  h263pay: fix typo in debug

=== release 0.10.19 ===

2010-03-06 00:43:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.19

2010-03-06 00:42:09 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2010-03-03 20:29:30 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.18.4 pre-release

2010-03-02 18:29:41 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Make sure we don't send invalid newsegments
	  Fixes #611501

2010-03-02 14:09:14 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Mark streams as being EOS at the right time.
	  This allows us to stop streaming only when all streams have gone past the
	  segment.stop and not before.
	  Fixes #611501

2010-02-26 18:10:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Advance sparse streams only as much as required to keep the gap smaller than 500ms
	  Changing it to the newest timestamp that was ever pushed will
	  increase the segment start in 500ms jumps, which could be just
	  after the next sparse stream buffer. E.g.
	  Video at 1.0s, sparse stream at 0.5s would jump the
	  sparse stream to 1.0s. Now a new sparse stream buffer could
	  appear that has a timestamp of 0.9s and this would be
	  dropped for no good reason because of bad luck.

2010-02-24 01:36:07 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* po/es.po:
	* win32/common/config.h:
	  0.10.18.3 pre-release

2010-02-24 02:05:49 +0100  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  Make sure FLUSH_STOP is sent so not to leave downstream flushing.

2010-02-23 17:25:54 +0100  Volker Grabsch <bugzilla.gnome.org@v.notjusthosting.com>

	* configure.ac:
	  configure: Use $PKG_CONFIG instead of pkg-config to fix cross compilation
	  Fixes bug #610839.

2010-02-23 17:24:03 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Reset skew detection after instantiating the jitterbuffer
	  ...not only when going to READY. This sets high_level and friends to
	  a more useful value.

2010-02-23 17:19:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Return 100 if high-level is 0 instead of dividing by zero

2010-02-22 12:24:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: avoid division by 0
	  Avoid a division by 0 when no constantDuration was specified and when out two
	  timestamps are equal.
	  Fixes #610265

2010-02-22 18:20:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvdepay.h:
	  dvdepay: don't output frames until we have a header
	  Wait for the complete first 6 header DIF packets before outputting a frame.
	  Decoders need this info to correctly decode the data.
	  Fixes #610556

2010-02-22 20:55:29 +0100  David Hoyt <dhoyt@llnl.gov>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Fix invalid memory access by first checking and then reading
	  Fixes bug #610483.

2010-02-18 09:05:50 +0100  Philippe Normand <phil@base-art.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: gst_pulsesink_get_mute: set result earlier.
	  In the cases where no buffer was process yet or the index is not
	  available, get_pulsesink_get_mute() would unconditionally return
	  FALSE.
	  https://bugzilla.gnome.org/show_bug.cgi?id=610337

2010-02-19 12:35:29 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* pkgconfig/gstreamer-plugins-good-uninstalled.pc.in:
	  pkgconfig: fix gstreamer-plugins-good uninstalled .pc file
	  Fix gst-plugins-base reference/requirement. This caused spurious
	  problems with uninstalled -ugly/-bad not finding -good plugins in
	  their unit tests (when distchecking).

2010-02-19 01:03:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* po/lv.po:
	* win32/common/config.h:
	  0.10.18.2 pre-release

2010-02-19 00:54:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/.gitignore:
	* tests/examples/shapewipe/.gitignore:
	  Make git ignore shapewipe examples and tests

2010-02-19 00:46:40 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  flvdemux: minor micro-optimisation
	  We know these values don't change during the loop, but the compiler
	  doesn't and has to re-check them for every iteration.

2010-02-19 00:39:50 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  flvdemux: remove static keyword from variables that shouldn't be static
	  Multiple flvparse/flvdemux instances should be able to operate without
	  trampling over each other by accidentally re-using the same (static)
	  variables. (Spotted by Mark Nauwelaerts)

2010-02-16 02:07:07 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  docs: add Since: markers for new jitterbuffer properties

2010-02-18 18:20:24 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix off-by-one logic error in frame rate cap regression commit

2010-02-17 16:27:33 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Use the correct duration when comparing segments
	  Do not confuse QtDemuxSegments with GstSegments when
	  comparing the total file duration with the segment duration
	  Fixes #610296

2010-02-17 18:06:29 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: add durations modulo 1<<32
	  For calculating the durations of each sample, we are supposed to add each
	  duration modulo 1<<32 so make the elapsed time counter a uint32.
	  Fixes #610280

2010-02-16 21:05:24 +0100  Anders Skargren <anders.skargren at axis.com>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: improve header mime-type parsing
	  Make the handing of the mime type within the "boundary" a bit less naive.
	  The standard for MIME allows parameters to follow the "type" / "subtype"
	  clause separated from the mime type by ';'.
	  Modifies the multipartdemuxer's header parsing so it doesnt assume
	  the whole line after "content-type:" is the mime type and thus makes it a bit
	  more resilient to finding absurd mime types in the case where parameters are
	  added.
	  Fixes #604711

2010-02-16 19:53:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid stopping NULL tasks
	  Check the task for NULL, it could be paused and set to NULL before.

2010-02-16 16:22:28 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix ALAC codec-data handling
	  ALAC codec-data apparently comes in (at least) two flavours (mov, mp4),
	  so use atom based parsing to retrieve required data, rather than
	  aiming for a specific offset.
	  See also #580731.

2010-02-16 15:50:23 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix debug message

2010-02-11 19:39:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_types.h:
	  qtdemux: handle signed values in 3GPP location tag

2010-02-08 21:35:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix typo in debug message

2010-02-16 15:00:13 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: reset some more stream state after seek
	  In particular, fixes non-flushing seek.

2010-02-16 14:44:11 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix frame rate cap regression
	  Look for a non-zero min_duration during initialisation to avoid
	  incorrect frame rate caps.

2010-02-16 10:13:17 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: log more details in buffer pool finalize
	  Helps to align with the loggin from libv4l.

2010-02-16 10:11:40 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: init datastructures after pre-conditions checks

2010-02-16 10:10:45 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: add a fixme for handling other YUV variants

2010-02-16 01:40:19 +0000  Brian Cameron <brian.cameron@sun.com>

	* gst/matroska/matroska-demux.c:
	  matroska: fix GST_ELEMENT_ERROR usage
	  Fixes #610053.

2010-02-16 00:50:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: fix up GST_CXXFLAGS properly
	  We don't want C specific flags in GST_CXXFLAGS, so base it on the
	  GST_CFLAGS that only contains the pkg-config CFLAGS but none of
	  the GST_OPTION_CFLAGS. Also, we only need the local includes once.
	  Fix typo as well (GST_FLAGS -> GST_CFLAGS).

2010-02-15 23:13:46 +0200  Stefan Kost <ensonic@users.sf.net>

	* configure.ac:
	  configure: base GST_CXXFLAGS on --cflags from pkg-config
	  pkg-config sets GST_CFLAGS and GST_LIBS. We need to use CFLAGS as a starting
	  point for for both C and CXX settings.

2010-01-20 18:52:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin: remove use of ntp_ns_base

2010-01-20 18:22:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpbin: remove more ntpnstime and cleanups
	  Remove some code where we pass ntpnstime around, we can do most things with the
	  running_time just fine.
	  Rename a variable in the ArrivalStats struct so that it's clear that this is the
	  current system time.

2010-01-20 18:19:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: use running_time for jitter
	  Use the running_time to calculate the jitter instead of the ntp time. Part of
	  the plan to get rid of ntpnsbase.

2010-01-20 17:04:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpbin: change how NTP time is calculated in RTCP
	  Don't calculate the NTP time based on the running_time of the pipeline but from
	  the systemclock. This allows us to generate more accurate NTP timestamps in case
	  the systemclock is synchronized with NTP or similar.

2010-02-15 12:12:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: printf format string fix
	  The compiler wants a cast here even though the type is already
	  typedefed as 64-bit integer (presumably because glib has typedefed
	  guint64 to unsigned long here).

2010-02-15 10:33:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska: fix printf format string

2010-02-15 00:50:10 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/raw1394/gst1394clock.h:
	* gst/matroska/ebml-write.h:
	* gst/rtpmanager/gstrtpjitterbuffer.h:
	  raw1394, matroska, rtpmanager: remove padding from structures
	  None of these element and class structures are in public headers,
	  so don't need padding.

2010-02-15 00:47:11 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update for new translator comment

2010-02-15 00:45:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: add comment for translators for 'x by y' message
	  Fixes #609724.

2010-02-15 01:28:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	  cairorender: Fix leaking of pad templates

2010-02-15 00:50:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/shapewipe.c:
	  shapewipe: Fix unit test for latest changes
	  Now the alpha is multiplied with the already existing alpha
	  value instead of simply ignoring it and the luma/chroma values
	  are kept, even if the output is 100% transparent.

2010-02-15 00:47:08 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/shapewipe.c:
	  shapewipe: Improve unit test output on errors

2010-02-14 23:17:20 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 96dc793 to 44ecce7

2010-02-13 23:28:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: bump -base requirement to git
	  For GST_RIFF_TAG_JUNQ.

2010-02-12 16:11:30 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2.c:
	  v4l2sink: change rank to NONE so it is never autoplugged

2010-02-13 18:18:42 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvparse.c:
	  flvdemux: Audio tags without any content are valid.
	  We silently ignore them instead of erroring out.

2010-02-13 18:07:50 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvparse.c:
	  flvdemux: Fix GST_CLOCK_DIFF usage.
	  It was previously checking for DIFF(a, b > 6 * GST_SECOND) instead of
	  the proper DIFF(a,b) > 6 * GST_SECOND

2010-02-13 16:27:07 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't forget to reset the indexed variable when cleaning up

2010-02-13 11:01:53 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvparse.c:
	  flvdemux: Speedup GstIndex usage
	  Used the _add_associationv variant of GstIndex since we know how many
	  associations we're adding. Trims up to 50% from index generation time.
	  Note : It would be great if the index could be generated on the fly or
	  on request as opposed to being fully created at startup.

2010-02-12 19:32:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: don't resync to invalid timestamps
	  If we detect backward timestamps on the server, don't try to resync when we
	  don't have an input timestamp (such as when using RTSP over TCP) instead, do
	  nothing but assume the timestamp was ok, it will correct itself when time goes
	  forwards.

2010-02-12 17:21:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix typo

2010-02-12 16:47:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: start out active and not buffering
	  There is no need to set the latency in the jittebuffer in _init, we will set
	  that later when going to PAUSED.
	  Set the jitterbuffer active and not buffering when starting.

2010-01-27 17:57:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpbin: more buffering work
	  When deactivating jitterbuffers when the buffering starts, keep the current
	  percent of the jitterbuffer and also set the jitterbuffer in the buffering state
	  so that we know when it's filled again.
	  Add property to get the buffering percentage of the jitterbuffer.

2009-10-14 16:29:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: adjust latency in buffer mode
	  When we are in buffer mode, adjust the buffering low/high thresholds based on
	  the total configured latency. If we don't and there is a huge queue or element
	  with a big latency downstream we might drain the complete queue immediately and
	  start buffering again.

2009-10-12 11:54:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add ts-offset to timestamp
	  Add the ts-offset to the buffer timestamp to get the final output timestamp of
	  the buffer.

2009-10-08 19:23:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpjitterbuffer.h:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpbin: do more accurate buffer offsets
	  Return the next timestamp in the jitterbuffer.
	  Use the min-timestamp of the jitterbuffers to calculate an offset so that the
	  next timestamp is pushed with a timestamp equal to running_time.
	  Start producing timestamps from 0 in the buffering case too.

2009-10-08 18:42:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: only start buffering when < 100%
	  Only start buffering when the percentage message is < 100 %.

2009-10-06 13:34:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: keep track of elapsed pause time
	  Keep track of the time we spend pausing the jitterbuffers when they were
	  buffering and distribute this elapsed time to the jitterbuffers.
	  Also keep the latency in nanosecond precision.

2009-10-06 13:33:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpjitterbuffer.h:
	  jitterbuffer: keep track of offset
	  Keep track of an outgoing offset that we add to each outgoing buffer to
	  compensate for PAUSE when buffering.
	  Adjust the offset when activating.

2009-10-06 13:30:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: report level using high watermark

2009-10-05 21:31:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtsp/gstrtspsrc.c:
	  rtpbin: pass running_time to jitterbuffer pause
	  Pass the current running time to the jitterbuffer when pausing or resuming so
	  that it calculate the right offsets.
	  Small cleanups and comments.
	  Set the default rtspsrc latency to 2 seconds.

2009-10-05 20:09:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpbin: add some comments

2009-10-05 19:45:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpjitterbuffer.h:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpbin: more buffering updates
	  Add signal to pause the jitterbuffer. This will be emitted from gstrtpbin when
	  one of the jitterbuffers is buffering.
	  Make rtpbin collect the buffering messages and post a new buffering message with
	  the min value.
	  Remove the stats callback from jitterbuffer but pass a percent integer to
	  functions that affect the buffering state of the jitterbuffer. This allows us
	  then to post buffering messages from outside of the jitterbuffer lock.

2009-10-05 13:32:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  rtpbin: propagate buffer-mode property
	  Propagate buffer-mode property to the jitterbuffers.
	  Intercept BUFFERING messages in rtpbin

2009-10-01 17:14:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: do more buffering implementation
	  Add callback for buffering stats.
	  Configure the latency in the jitterbuffer instead of passing it with _insert.
	  Calculate buffering levels when pushing and popping
	  Post buffering messages.

2009-10-01 12:46:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: flesh out buffering mode some more
	  Add a buffering state to the jitterbuffer and wait until buffering ends before
	  pushing out packets.

2009-10-01 12:09:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: hook up the mode property
	  Expose a mode property on the jitterbuffer.
	  Fix the case where timestamps are -1 in the check for outgoing timestamps.

2009-10-01 11:20:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: add buffering mode options
	  Add getters and setters for different buffering modes that the jitterbuffer will
	  support. Default to the current slave mode.

2010-02-12 15:54:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2.c:
	  v4lsink: lower rank to MARGINAL

2010-02-12 16:06:45 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvparse.c:
	  flvdemux: Obtain the index from the end of an flv file in push mode
	  Allows for better support of seeking in flv files when in push mode

2010-01-21 11:55:15 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Drop video frames up to the desired keyframe after a seek
	  The audio packets in AVI are generally muxed ~0.5s before the
	  corresponding video packet. This changes causes downstream to only
	  receive packets with roughly corresponding timestamps.

2010-01-19 18:35:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: more DISCONT handling
	  Add some debug in the DISCONT handling code.
	  When we receive a DISCONT in push mode, mark all streams as DISCONT.

2010-01-19 10:51:08 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix _handle_seek_push () and new segement behaviour

2010-01-18 17:13:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: cleanups
	  Make sure we reset the demuxer correctly wrt parsing the index.
	  Don't leak pending seek events.
	  Rename some methods to reflect what they do and to avoid confusion with similar
	  method names.
	  Try to make the seeking threadsafe by protecting the setup code with a lock.
	  Make sure we post errors when a seek fails.

2010-01-18 11:45:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: rename some variables
	  seek_event -> seg_event
	  event_seek -> seek_event

2010-01-15 18:00:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: take fallback duration from avih
	  When we have not parsed any indexes yet, we don't know the length of the streams
	  and we must take the length given in the avih as a fallback.
	  Avoid some typechecking.

2009-12-04 15:13:12 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Push mode seeking support

2010-02-01 16:04:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: cleanup properties
	  Use more default constants.
	  Use static strings param flag.
	  Init properties explicitly instead of letting gobject do this.

2010-02-12 15:34:38 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/speex/gstspeexdec.c:
	  speex: add missing include

2010-02-05 13:28:53 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/debugutils/gsttaginject.c:
	  taginject: fix multi-value tag example
	  We need to use {} to specify a list.

2010-02-01 14:43:04 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	* gst/wavparse/gstwavparse.c:
	  avi,wav: also handle JUNQ chunk in addition to JUNK

2010-02-04 15:59:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	  rtppay: don't ignore result from set_outcaps
	  set_outcaps can fail and we need to propagate the result upstream.

2010-02-04 15:36:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  flvparse: fix confusing debug messages

2010-01-27 13:28:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add some more debug info

2010-01-27 13:26:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: avoid segfault when shutting down
	  when we are shutting down, we might still receive state updates from pulseaudio
	  but since we are unparented we should not do anything with the NULL parent
	  anymore.

2010-01-26 18:33:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: fix timestamp problems
	  When the pad with the highest framerate goes EOS, instead of not timestamping
	  output buffers, intepollate timestamps and durations from the last seen ones.
	  Fixes #608026

2010-02-12 11:32:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: Update documentation

2010-02-12 11:18:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* tests/check/Makefile.am:
	* tests/examples/Makefile.am:
	  Moved 'shapewipe' from -bad to -good
	  Fixes bug #584536.

2010-02-10 10:52:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 29/29] shapewipe: Preserve the input color values in all cases

2010-02-10 10:50:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 28/29] shapewipe: Scale mask alpha values by the source alpha values

2010-02-10 10:42:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 27/29] shapewipe: Fix ARGB processing

2010-02-10 10:34:24 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/shapewipe/shapewipe-example.c:
	  [MOVED FROM BAD 26/29] shapewipe: Print some more details on error/warning messages

2010-02-08 08:26:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 25/29] shapewipe: Improve/add debug output

2010-02-08 08:20:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 24/29] shapewipe: Always hold the mask mutex before signalling the GCond

2010-02-08 08:19:48 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 23/29] shapewipe: Move chain function error cases at the end of the function and add useful debug output

2010-02-08 08:12:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	  [MOVED FROM BAD 22/29] shapewipe: Fix race condition during shutdown that can lead to a deadlock

2010-02-08 08:11:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 21/29] shapewipe: Drop mask buffer on FLUSH events

2010-02-08 08:09:55 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	  [MOVED FROM BAD 20/29] shapewipe: Update copyright year

2010-02-08 08:08:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 19/29] shapewipe: Don't reset properties when going PAUSED->READY
	  Also use defines for the default values of the properties.

2010-01-16 16:52:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 18/29] shapewipe: Replace floating point arithmetic in the inner processing loops by integer arithmetic

2009-12-10 10:40:10 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 17/29] shapewipe: Don't do pointer dereferences in the processing loop
	  Lowers the time taken there in my testcase from 6.91% to 6.20%
	  as measured by callgrind.

2009-07-08 17:59:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 16/29] shapewipe: Add BGRA support for video in/output

2009-07-02 11:24:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	  [MOVED FROM BAD 15/29] shapewipe: Add support for ARGB video input/output

2009-06-23 18:23:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 14/29] shapewipe: Correctly handle 0/1 fps

2009-06-09 19:14:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	  [MOVED FROM BAD 13/29] shapewipe: Implement basic QoS
	  This change is based on Tim's QoS implementation
	  for jpegdec.

2009-06-09 18:45:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 12/29] shapewipe: Proxy queries on the video pads to the correct peers

2009-06-09 18:37:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 11/29] shapewipe: Proxy bufferalloc on the video sinkpad

2009-06-09 18:25:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 10/29] shapewipe: Try to work inplace if possible
	  This saves one new, large allocation per frame for the
	  most cases.

2009-06-04 08:56:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/shapewipe.c:
	  [MOVED FROM BAD 09/29] shapewipe: Increase timeout of the unit test

2009-06-01 21:24:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 08/29] shapewipe: Fix some issues that were exposed by the new unit test

2009-06-01 21:24:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/shapewipe.c:
	  [MOVED FROM BAD 07/29] shapewipe: Add unit test for shapewipe

2009-05-31 21:33:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 06/29] shapewipe: Add documentation and integrate into the build system

2009-05-29 21:07:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	  [MOVED FROM BAD 05/29] shapewipe: Adjust border to still have everything transparent at 1.0 and the other way around

2009-05-29 16:55:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	* tests/examples/shapewipe/shapewipe-example.c:
	  [MOVED FROM BAD 04/29] shapewipe: Divide the border value by two, otherwise we use a twice a wide border

2009-05-29 16:51:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	* tests/examples/shapewipe/shapewipe-example.c:
	  [MOVED FROM BAD 03/29] shapewipe: Add border property to allow smooth borders
	  ...and use a border of 0.01 in the example application.

2009-05-29 16:00:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/shapewipe/Makefile.am:
	  [MOVED FROM BAD 02/29] shapewipe: Fix Makefile of the example application

2009-05-29 15:32:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/shapewipe/Makefile.am:
	* gst/shapewipe/gstshapewipe.c:
	* gst/shapewipe/gstshapewipe.h:
	* tests/examples/shapewipe/Makefile.am:
	* tests/examples/shapewipe/shapewipe-example.c:
	  [MOVED FROM BAD 01/29] shapewipe: Add a simple shapewipe transition filter & example application

2010-02-06 18:19:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: Only flush the FLAC decoder if it wasn't created right before
	  If the FLAC decoder is flushed, its state will be set to frame-sync mode,
	  which will sync to the next *audio* frame and makes it ignore all headers.
	  This prevented tags and everything else to show up when using flacdec
	  in push mode.
	  Fixes bug #608843.

2010-02-11 01:12:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* MAINTAINERS:
	  Update MAINTAINERS

2010-02-12 00:03:09 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: back to development
	  Slushy freeze remains in effect.

=== release 0.10.18 ===

2010-02-10 23:18:22 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.18

2010-02-10 23:17:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2010-02-10 20:36:56 +0000  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: temporary safety check to avoid crashes with a certain file
	  Add temporary check to avoid crashes with a certain file when seeking
	  until the real cause of this is figured out. See #609405.

2010-02-05 18:05:39 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: skip unknown atoms when looking for moov
	  Fixes bug #609107

2010-02-05 02:13:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.17.3 pre-release

2010-02-04 19:10:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/bg.po:
	* po/hu.po:
	  po: update translations

2010-02-04 14:46:56 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: Set the segment start time to the requested seek time for non-keyframe seeks

2010-02-04 12:00:03 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix time returned for index at a byte offset
	  The logic for searching forwards/backwards was swapped

2010-02-01 19:22:24 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: initialize stereo decoding state

2010-01-28 18:58:08 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: improve stream synchronization
	  In particular, do not make it send newsegment updates that
	  sort-of contradict the indented playback segment (e.g. start time).

2010-01-28 18:53:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix bridging (time) gaps in streams
	  As a side effect, avoid sending newsegment updates with start times
	  that go back and forth, which leads to bogus downstream running_time.
	  Also fixes seeking in bug #606744.

2010-01-28 18:49:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix stream synchronization
	  .. by initializing streams starting at 0, as that is basically
	  where we 'seek to' at the start and assume streams to start elsewhere.
	  Also enables newsegment update events for subtitle streams.

2010-02-02 13:41:03 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	  jpeg: don't directly access message, some message have args
	  This caused bogus messages, such as reported in bug #607471.

2010-02-02 00:02:34 +0000  David Hoyt <dhoyt@llnl.gov>

	* ext/libpng/gstpngdec.c:
	  png: fix compilation with libpng 1.4
	  png_set_gray_1_2_4_to_8() has been deprecated for a while and was
	  finally removed in libpng 1.4.x. Use png_set_expand_gray_1_2_4_to_8()
	  instead.
	  Fixes #608629.

2010-02-01 16:46:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: free transports on errors
	  See #608564

2010-02-01 09:18:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: fix unportable printf format

2010-01-30 15:18:48 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 15d47a6 to 96dc793

2010-01-27 17:53:07 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: index timestamps should be in seconds, not milliseconds

2010-01-27 15:24:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: free some more when resetting
	  Fixes #608255.

2010-01-27 15:24:24 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpspeexpay.c:
	  rtpspeexpay: fix occasional buffer leak
	  Fixes #608255.

2010-01-27 15:22:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: prevent invalid arithmetic if not setup yet
	  Fixes #608255.

2010-01-27 16:34:21 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_mmx.h:
	  videomixer: Fix assembly register constraints
	  Fixes bug #608209.

2010-01-27 01:56:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/config.h:
	  0.10.17.2 pre-release

2010-01-27 01:52:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/LINGUAS:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update translations

2010-01-27 01:49:49 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/.gitignore:
	  checks: ignore deinterlace check binary

2010-01-27 01:18:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: purge all mention of CVS

2010-01-26 11:18:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: ignore streams that finished
	  When we receive an UNEXPECTED from a stream, move to the next stream and only go
	  EOS when all streams are EOS. When selecting a stream to push, ignore streams
	  that went EOS.
	  Fixes #607949

2010-01-25 17:23:43 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: don't deref NULL
	  Error out when the pool gets shutdown.

2010-01-25 17:21:13 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	* sys/v4l2/v4l2src_calls.c:
	* tests/check/Makefile.am:
	  Revert "v4l2src: don't deref NULL"
	  This reverts commit 3d9d34bd60faeb940b36d992a47168fc895036ba.

2010-01-25 14:16:22 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	* sys/v4l2/v4l2src_calls.c:
	* tests/check/Makefile.am:
	  v4l2src: don't deref NULL
	  Error out when the pool gets shutdown.

2010-01-23 15:32:48 -0800  Michael Smith <msmith@xiph.org>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: when creating an overflow buffer, copy timestamps.

2010-01-23 14:47:55 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: dmb1 is a valid fourcc for Motion-JPEG

2010-01-23 14:20:02 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdeux: IV32 is also used for Indeo 3 video streams

2010-01-22 16:48:01 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/icles/ximagesrc-test.c:
	  build: no unused variables when disabling asserts

2010-01-21 23:17:40 -0300  Roland Krikava <rkrikava@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Avoid negative overflow on keyframe search
	  Do not overflow negatively when searching a previous
	  "keyframe" on audio streams. Could cause infinite loops
	  on backwards playback
	  Fixes #607718

2010-01-21 17:22:38 -0800  Peter van Hardenberg <pvh@songbirdnest.com>

	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpegenc: enlarge buffer if libjpeg tells us it's out of space. Fixes buffer overflow on some high-quality, low-resolution jpeg encodes.

2010-01-21 19:24:22 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix compiler warnings under OS X.

2010-01-21 17:57:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: don't parse NULL indexes
	  for some streams we might fail to fetch the index offsets. Don't try to parse
	  NULL indexes in those cases.

2010-01-18 21:15:51 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: ptime should is in nanoseconds
	  https://bugzilla.gnome.org/show_bug.cgi?id=607403

2010-01-20 15:11:15 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	  wavenc: Post warning if file isnt finished properly
	  When the pipeline is shut down and the file isn't
	  finished properly, wavenc should post a warning.
	  Fixes #607440

2009-05-27 13:51:44 +0200  Arnout Vandecappelle <arnout@mind.be>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: make index size configurable.
	  Added the 'min-index-interval' property to matroskamux,
	  which determines how much time (nanoseconds) is left
	  between keyframes stored in the index.
	  Fixes #583985.

2010-01-20 16:28:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: scale spspps_interval to milliseconds
	  The spspps_interval is kept in seconds. Convert it to milliseconds before
	  comparing it to another value in milliseconds.

2010-01-20 15:18:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: always keep media segments within total duration
	  ... as opposed to only doing so following a seek.

2010-01-20 15:44:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: rename spspps-interval property
	  Rename the spspps-interval property to config-interval because it is nicer.

2010-01-19 18:37:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: skip RIFF and index in push mode
	  When we are in push mode, we can encounter RIFF and idx tags in the data chunk
	  when we are dealing with ODML files. In these cases, simply skip the chunks and
	  continue streaming instead of going EOS.

2010-01-20 11:27:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: more DISCONT handling
	  Add some debug in the DISCONT handling code.
	  When we receive a DISCONT in push mode, mark all streams as DISCONT.

2010-01-20 11:26:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: reset on flush events
	  When we receive a flush event on the sinkpad, reset the EOS state and the
	  flowreturn of all streams. Also mark the streams with a DISCONT.

2010-01-20 11:22:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: rename some variable
	  Rename the seek_event variable to seg_event because it really contains the
	  newsegment event that needs to be pushed.

2010-01-20 00:54:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 14cec89 to 15d47a6

2010-01-18 14:49:26 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Don't set profile-level-id in out caps
	  The profile-level-id represents restrictions on what can be sent, it does not
	  describe the stream. So it should be reflected in the sink caps of the
	  payloader, not the src caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=607353

2010-01-18 14:41:10 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Don't ignore the return value from set_outcaps
	  https://bugzilla.gnome.org/show_bug.cgi?id=607353

2010-01-18 17:43:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/greedyhmacros.h:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Fix license and copyright headers

2010-01-18 14:57:42 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: move G_END_DECLS to the end

2010-01-18 14:55:38 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: fix bufferpool file names in header comment

2010-01-15 18:15:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: avoid some typecasting

2010-01-15 18:13:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: avoid some type checks

2010-01-15 18:09:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: fallback to avih duration
	  when we have not yet parsed the indexes (in push mode, for example) use
	  the duration as given in the avih header instead of -1.

2010-01-15 13:32:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: g_free is NULL safe

2010-01-15 13:27:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use DEMUX errors, instead of DECODE
	  qtdemux should use DEMUX errors, and not DECODE
	  Conflicts:
	  gst/qtdemux/qtdemux.c

2010-01-14 19:16:19 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Minor refactor
	  Replace repeated code with a function call

2010-01-14 17:11:13 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: Handle another kind of redirect trak
	  Some traks might contain a redirect rtsp uri inside
	  hndl atom (which is a dref atom entry). This commit makes qtdemux
	  post a message when it finds one of these traks and there are
	  no other traks.
	  Fixes #597497

2010-01-14 16:13:08 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: Post error when reaching EOS without pads
	  Post an error when EOS is reached and there are no src pads

2010-01-14 14:13:50 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Do not post empty redirect messages
	  Some misinterpreted data could result in posting redirect messages
	  with empty redirect strings. It is better not to post them.
	  An example is the file on bug #597497

2010-01-14 18:19:25 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: polish last buffer end time usage
	  That is, reset it upon seek, and note that (rarely) last pushed buffer
	  time might precede segment start.

2010-01-13 16:48:46 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/videomixer/blend_mmx.h:
	  videomixer: use 'q' constraint instead of 'r'
	  This avoids the "bad register name `%dil'" compilation errors on 32bit where
	  because of 'r' gcc puts the value in a general purpose register and then tries
	  to access the lower part as %dil/%sil which is not existing on 32bit. 'q' requests
	  a-d registers

2010-01-13 16:44:58 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avi: add missing include for sscanf

2010-01-13 09:36:03 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer10bands.c:
	  equalizer: Fix property description for the 3rd band of the 10band equalizer
	  The frequency is actually 237 Hz, not 227 Hz.
	  Fixes bug #606692.

2010-01-13 09:22:20 +0100  Kipp Cannon <kcannon@ligo.caltech.edu>

	* gst/audiofx/audioamplify.c:
	  audioamplify: Allow negative amplifications
	  Fixes bug #606807.

2010-01-13 09:17:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/taglib/gstapev2mux.cc:
	  apev2mux: Don't call constructors directly, this leads to compiler errors with gcc 4.5

2010-01-12 17:39:05 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use G_GSIZE_FORMAT for platform independent gsize qualifier
	  Fixes build on macosx

2010-01-11 19:02:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: refactor eos sending when pausing loop
	  Also, prevent hanging if no pads yet on which to send eos by
	  posting a message instead.

2010-01-11 17:50:35 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: standardize seek handling
	  ... which implies fixing some corner cases.

2010-01-11 15:14:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: use more generic xiphN_streamheader_to_codecdata helper

2010-01-11 17:50:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: reflow audio and video setcaps and improve logging
	  Also ensure width and height are available as they are mandatory
	  in matroska specs.

2010-01-11 11:42:43 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix offset for type 2 mp4a sound sample descriptions.
	  Allows us to correctly find the esds (and thus the codec data) for such
	  mp4a files.

2010-01-11 15:45:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	  rtpmp4g(de)pay: Only handle raw aac
	  rtpmp4g(de)pay should only handle raw AAC streams

2010-01-11 18:59:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Implement basic QoS
	  This drops frames if they're too late anyway before blending and all
	  that starts but QoS events are not forwarded upstream. In the future
	  the QoS events should be transformed somehow and forwarded upstream.

2010-01-11 14:48:26 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	  rtpmp4a(de)pay: Only accept raw aac
	  rtpmp4a(de)pay should only handle raw aac to conform to the RFC

2010-01-11 18:35:47 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/blend_mmx.h:
	  videomixer: Add MMX implementations for I420 and all non-alpha RGB formats

2010-01-04 10:24:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_i420.c:
	* gst/videomixer/blend_mmx.h:
	* gst/videomixer/blend_rgb.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Refactor processing functions
	  This allows easier plugging of optimized processing functions
	  in the future, like for SSE or AltiVec.

2010-01-11 13:26:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/matroska/matroska-mux.c:
	  avimux: matroskamux: rename aac's stream-format to raw
	  AAC's none stream-format has been renamed to raw, rename
	  on avimux and matroskamux as well

2010-01-11 12:07:29 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Only accept raw aac
	  makes matroskamux reject aac streams that are not
	  in raw format (stream-format=none)
	  Fixes #598350

2010-01-11 12:08:55 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: Only accept raw aac
	  makes avimux reject aac streams that are not
	  in raw format (stream-format=none)
	  Fixes #598350

2010-01-11 10:38:10 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Oops. The gpointer cast is needed because of the const qualifiers on the data elements

2010-01-11 10:17:54 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Debug -> info level for a message for benchmarking index parsing
	  The extra message output at higher levels affects the accuracy of the
	  benchmark.

2010-01-11 10:05:10 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Don't check for NULL pointers or cast to gpointer as this is not needed

2010-01-08 13:55:05 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Refactor stbl sub-atom freeing. Free when index has been completely parsed.

2010-01-08 14:32:06 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Avoid whitespace commits due to inconsistent GNU indent behaviour

2010-01-11 00:10:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: remove newline at end of debug statement

2010-01-08 19:26:21 +0100  Havard Graff <havard.graff@tandberg.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Compiler warning fixes for Windows
	  Just simple missing casts
	  Fixes bug #606438.

2010-01-08 18:04:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: fix seekpoints property copy-and-paste documentation

2010-01-06 17:06:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flacenc: optionally add a seek table
	  API: GstFlacEnc:seekpoints
	  Fixes #351595.

2010-01-08 11:33:02 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Use more glib and be safer
	  Be safer on sscanf by limiting string format sizes.
	  Remove useless parameter and use g_strndup.

2010-01-08 10:44:44 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Simplifying code
	  Greatly simplify the IDIT chunk handling by using sscanf
	  instead of 'manually' parsing. Also replaces strncasecmp and
	  is_alpha/is_digit with glib versions.

2010-01-08 10:18:30 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: it's feb for february
	  Fix typo in last commit.

2010-01-08 09:17:22 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Parse and post IDIT dates
	  Parses and post date tags contained in IDIT chunks.
	  Fixes #503582

2010-01-07 17:25:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Add property for not draining the history on kernel changes
	  Currently this only works if the kernel size doesn't change, in the future
	  it will be possible to change the kernel size too without draining
	  the complete history and without loosing anything.
	  Partially based on a patch by
	  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

2010-01-07 16:58:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: remove weird memcmp code
	  Use plain memcmp for comparing memory instead of the custom buggy one.
	  Fixes #606198

2010-01-07 15:38:36 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/level/gstlevel.c:
	  level: fix typo in 'message' property description

2010-01-06 14:06:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: really use upstream timestamp if there is one
	  See/fixes #603471.

2010-01-06 13:45:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg728pay: remove unused adapter peek

2010-01-05 19:00:35 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Improve passthrough tests
	  Improve passthrough tests by forcing more specific
	  interlaced/deinterlaced caps to be tested

2010-01-05 18:22:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Adds some docs to the new tests
	  Adds some docs explaining the utility functions of the check
	  tests of deinterlace

2010-01-05 18:14:08 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Adds tests for passthrough
	  Adds tests for checking if the element really does
	  passthrough in disabled mode and in auto (if the input is
	  not interlaced)

2010-01-05 07:50:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/deinterlace.c:
	  deinterlace: Adds tests for caps acceptance
	  Adds check unit tests for deinterlace for validating
	  caps accepting and the expected caps output on the
	  other pad

2010-01-04 13:43:00 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/deinterlace.c:
	  deinterlace: Adds basic check test
	  Adds a basic check test for deinterlace element

2010-01-04 15:44:28 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/qtdemux/Makefile.am:
	* gst/qtdemux/qtdemux.c:
	  qtdemux: Add support for wave-style audio in qt.
	  Uses gstriff to parse the wave headers appropriately. Tested with MS-ADPCM
	  content.

2009-12-31 17:09:03 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* tests/check/elements/rtp-payloading.c:
	  tests: Add G.729 RTP payloader/depayloader test
	  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-31 16:52:30 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Simplify adapter usage
	  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-31 16:27:30 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Support ptime from caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-02 19:35:21 +0530  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/README:
	  rtp: Add maxptime to the README
	  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2010-01-05 19:03:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723depay.h:
	  rtpg723depay: add G723 depayloader

2010-01-05 19:02:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729depay.h:
	  rtpg729depay: remove unused variable

2010-01-05 18:33:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg723pay.h:
	  rtpg723pay: rewrite payloader
	  Handle all 3 packet sizes according to RFC 3551.
	  Totally untested, we don't have a G723 encoder.
	  Fixes #605882

2010-01-05 11:47:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix chunk counter

2010-01-04 19:44:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: more work at reducing loop overhead
	  Try to avoid derefs when parsing the index. Save the state into the structures
	  when we exit the loop instead of for each iteration.

2010-01-04 16:33:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: cleanups and make duration more accurate
	  Make the QtDemuxSample struct smaller by keeping the duration and the pts_offset
	  as their 32 bit values.
	  Make some macros to calculate PTS, DTS and duration of a sample.
	  Deref the sample index less often by keeping a ref to the sample we're dealing
	  with.

2010-01-04 13:41:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: simplify logic to calculate duration
	  Since we no longer store the timestamp and duration in nanoseconds, we can now
	  simply store the duration as-is.

2010-01-01 16:42:57 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Store timestamps in mov format in the index
	  This allows faster building of the index upon seeks so that scaling of
	  timestamps only occurs when actually needed.

2009-12-18 13:54:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: make seeking in push mode work
	  Move sample position checks into qtdemux_parse_samples where we can protect it
	  with a lock.
	  Refactor and make an qtdemux_ensure_index function.
	  Rename qtdemux_do_push_seek to qtdemux_seek_offset in order to avoid confusion
	  with gst_qtdemux_do_push_seek.

2009-12-18 12:44:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: move error code out of normal flow

2009-11-24 16:27:26 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: Add push mode seek support for seeking to obtain the moov atom

2010-01-05 12:22:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix on-npt-stop signal warnings for RDT
	  The RDT manager does not implement this signal so we need to check for it before
	  trying to connect to it.

2010-01-05 09:47:00 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: fix memory leak in new uri handler code
	  Don't leak a string everytime get_uri() is called and a device
	  has been set. There's a limited number of devices, so just
	  intern the string instead of doing more elaborate housekeeping
	  and storing it in the instance struct or so.

2010-01-01 14:10:49 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	  avimux: fix typo in warning message

2010-01-04 09:28:36 -0300  Robert Weidlich <gnomebugzilla@robert.weidlich.cc>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2send: Add 'public' property
	  Adds a property to set 'public' flag on libshout, making
	  the stream listed on the server's stream directory.
	  Fixes #605269

2009-12-30 14:14:55 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Add tags for average and maximum bitrate
	  Fixes #599300.

2009-12-26 16:59:14 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: do not try to alloc really large buffers
	  When nsamples_out is larger than nsamples_in, using unsigned
	  ints lead to a overflow and the resulting value is wrong and
	  way too large for allocating a buffer. Use signed integers
	  and returning immediatelly when that happens.

2009-12-25 12:38:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	  videomixer: optimize blend code some more
	  Use more efficient formula that uses less multiplies.
	  Reduce the amount of scalar code, use MMX to calculate the desired
	  alpha value.
	  Unroll and handle 2 pixels in one iteration for improved pairing.

2009-12-24 22:59:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_i420.c:
	* gst/videomixer/blend_rgb.c:
	  videomixer: scale and clamp
	  Scale and clamp to the max alpha values.

2009-12-24 22:50:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: scale and clamp alpha to its full extend
	  Convert the alpha value to 0->255 when setting and to 0->256 when using as
	  a scaling factor. This makes sure we can reach the full opacity value of 0xff in
	  all cases.

2009-12-24 22:23:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix some comments, remove property check
	  Fix some comments, clarify some FIXMEs
	  Remove the on-ntp-stop signal check now that the jitterbuffer is in
	  -good and we know that it supports this signal.

2009-12-24 20:27:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: some trivial cleanups

2009-12-24 17:04:28 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Parse all rtpinfo entries
	  Do not forget to parse all rtp-info entries, instead of
	  parsing the first one only.
	  Fixes #605222

2009-12-22 12:44:50 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: perf tag should map to GST_TAG_ARTIST

2009-12-24 17:03:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/interleave/interleave.c:
	  interleave: fix weird indentation

2009-12-24 17:01:54 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: use faster _adapter_copy() whem possible

2009-12-24 17:01:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/audiofx/firfilter-example.c:
	  tests: use right type when passing vararg value

2009-12-23 17:50:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: use a single decoder field for both push and pull mode

2009-12-23 17:03:32 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix possible hanging in pull mode seeking
	  A seek in multi-sink pipeline typically leads to several seek events in a row,
	  which could lead to sending several newsegments in a row without intermediate
	  flushing.  These would then accumulate, distort rendering times and as such
	  lead to 'hanging'.

2009-12-23 19:39:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: fix uninitialized variable

2009-12-23 13:09:54 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: use boilerplate

2009-12-23 00:38:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL16pay.h:
	  rtpL16pay: convert to baseaudiopayload
	  Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes
	  a bunch of problems that were already solved in the base class.
	  Fixes #853367

2009-12-23 00:30:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtppcmapay.c:
	  rtppcmapay: the boilerplate macro sets parent_class

2009-12-22 22:27:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpbin: avoid some structure copies
	  Don't make copied in the getter and setter for SDES in the RTPSource. This
	  avoids a couple of copies of the SDES structure when generating RTCP
	  packets.

2009-08-31 18:42:25 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpmanager: improve SDES handling
	  Store SDES internally as a struct to support multiple PRIV values.
	  Include all values set in SDES struct when sending RTCP SDES.

2009-12-22 14:41:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: add some fixmes

2009-12-22 14:35:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: baseclass handles timestamps for us

2009-12-22 14:27:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: reset start variable properly

2009-05-29 15:49:27 +0300  Marco Ballesio <marco.ballesio@nokia.com>

	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263depay.h:
	  Drop the whole frame if a packet is lost.
	  Fixes #582575

2009-12-21 20:39:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: add option to insert PPS/SPS in streams
	  Add a new spspps-interval property to instruct the payloader to insert
	  SPS and PPS at periodic intervals in the stream.
	  Rework the SPS/PPS handling so that bytestream and AVC sample code both use the
	  same code paths to handle sprop-parameter-sets. This also allows to have the AVC
	  code to insert SPS/PPS like the bytestream code.
	  Fixes #604913

2009-12-21 19:12:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 47cb23a to 14cec89

2009-12-21 12:01:53 -0300  Jonathan Conder <j@skurvy.no-ip.org>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	* gst/qtdemux/qtdemux_types.c:
	  qtdemux: Adds new tags
	  Adds some new tags mapping to qtdemux.
	  Fixes #599759

2009-12-21 15:05:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: add property to remove pads automatically
	  Add a property called autoremove to automatically remove the pads of sources
	  that timed out.
	  Fixes #554839

2009-12-21 14:55:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  ssrcdemux: fix comparison
	  A NULL means no pad was found.

2009-11-08 11:49:14 +0100  Edward Hervey <bilboed@bilboed.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Add GstURIHandler interface. Fixes #601143
	  This allows using v4l2://[<device>]

2009-12-20 17:24:47 -0800  Michael Smith <msmith@xiph.org>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: pass length parameter to g_convert

2009-12-18 12:44:50 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-demux.c:
	  matroska: Fix unitialized variable.
	  Yes, it's stupid, but macosx compilers are even more stupid.

2009-12-17 16:01:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	  videomixer: Fix assembly compilation on x86
	  Fixes bug #604814.

2009-12-17 17:37:03 +0100  Branko Čibej <brane at xbc.nu>

	* gst/replaygain/rganalysis.c:
	  rganalysis: fix timestamp rounding
	  Use scaling function to round and avoid overflows.
	  Fixes #604352

2009-12-17 17:27:42 +0100  Tiago Katcipis <tiago.katcipis@digitro.com.br>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg723pay.h:
	  rtp: add G723 payloader
	  Fixes #597823

2009-12-17 16:22:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_types.c:
	  qtdemux: Fix ALAC codec_data parsing
	  Fixes #604611

2009-12-16 17:28:30 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Remove cpp style coments
	  Removes // comments and replace them with /* */ comments

2009-12-16 12:48:02 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: also consider BlockNumber indicated in index when seeking

2009-12-16 12:43:27 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-read.h:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: support push based mode
	  Fixes #598610.

2009-12-16 12:44:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	  matroskademux: fix ebml read cache usage

2009-12-16 10:50:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	  videomixer: Use movzbl instead of movzxb for moving one byte to a l register
	  For some reason latest gcc/binutils accept movzxb here while
	  movzbl would be correct and is the only thing accepted by older
	  gcc/binutils.
	  Fixes bug #604679.

2009-12-16 06:59:01 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	  videomixer: src/dest are input and output of the AYUV blending MMX assembler

2009-12-15 18:18:54 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiowsincband.c:
	  audiowsincband: Use the same upper length limit as audiowsinclimit

2009-12-12 17:00:50 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	  audiowsinc{limit,band}: Allow much larger filter lengths now

2009-12-11 12:27:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Fix frequency response calculation

2009-12-08 14:57:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Remove dead assignments

2009-12-06 16:58:51 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Add special processing functions for Mono/Stereo
	  This provides another 7% speedup for the time domain convolution and 1.5%
	  speedup for the FFT convolution on Mono input.
	  This optimization assumes that the compiler simplifies calculations
	  and conditions on constant numbers and unrolls loops with a constant
	  number of repeats.

2009-12-04 09:25:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Add a "low-latency" mode
	  This will always use time-domain convolution, which lowers the latency.
	  With FFT convolution it's always a multiple of the kernel length,
	  with time domain convolution it's only the pre-latency of the filter kernel.

2009-12-04 09:00:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Remove obsolete TODO comments

2009-12-03 20:12:01 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Use samples everywhere instead of samples*channels sometimes

2009-12-03 17:27:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: FFT convolution implementation
	  This provides a great speedup, especially the relationship between kernel
	  length and processing size is now logarithmic instead of linear. Below a
	  kernel size of 32 it's a bit slower, afterwards it's much faster:
	  17     0.788000 -> 0.950000
	  33     1.208000 -> 1.146000
	  65     2.166000 -> 1.146000
	  ...
	  4097 107.444000 -> 1.508000
	  For sizes smaller 32 the normal time-domain convolution is chosen,
	  for larger sizes the FFT convolution is automatically used.
	  Fixes bug #594381.

2009-11-27 20:33:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Make most code parts independent of the processing functions and used convolution algorithm
	  Only remaining part is the residue pushing, which will be fixed later.

2009-11-26 15:17:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Optimize time-domain convolution
	  Remove some redundant calculations, move comparisions out of
	  inner loops, etc.
	  This makes the convolution about 3 (!) times faster but
	  processing time is of course still proportional to the
	  filter size.

2009-11-26 10:45:37 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once

2009-11-25 18:12:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Rewrite timestamp tracking
	  It's much simpler now and doesn't introduce accumulating rounding
	  errors.

2009-11-25 17:39:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Rename some variables and change comments

2009-11-24 20:06:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Add const qualifier to the source data array

2009-12-14 20:08:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/videomixer.c:
	  videomixer: Add MMX implementations of the AYUV blending and color filling functions
	  This provides a 20% speedup for blending and 100% for color filling.
	  The blending can probably be optimized even more.

2009-12-13 13:19:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3v2frames.c:
	  id3demux: prefer two letter ISO 639-1 code for extended comment

2009-12-13 13:10:12 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix up language code extraction some more
	  Quicktime uses ISO 639-2 for language codes, but GST_TAG_LANGUAGE
	  is supposed to hold a ISO 639-1 code, so convert as needed using
	  the new API from -base.
	  See #602126.

2009-12-13 12:45:22 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: fix language code writing and extraction
	  Matroska uses three-letter ISO 639-2B codes, but GST_TAG_LANGUAGE is
	  supposed to contain two-letter ISO 639-1 codes, so use new language
	  code mapping functions in -base to convert between those two as
	  needed.
	  Fixes #505823.

2009-12-07 20:54:07 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: minor debug message changes
	  Fix up a few debug messages so that it's clearer what they mean.

2009-12-12 17:44:04 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  Revert "qtdemux: Correctly parse classification tags"
	  This reverts commit cd883aa60c1133196a6ae052884d15c295c37dde.
	  Previous code was correct, 4 is due to table and language code,
	  not only language code

2009-12-12 16:28:36 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Correctly parse classification tags
	  In clsf atoms, the language code is 2 bytes long, not 4.

2009-12-12 16:55:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Dequeue current buffer on FLUSH_STOP and don't unref NULL buffers
	  ... NULL buffers shouldn't really happen anymore when popping the
	  buffer from GstCollectPads but better check for this and print a warning.

2009-12-11 13:11:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_i420.c:
	  videomixer: Fix stupid mistake in last commit

2009-12-11 12:35:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_i420.c:
	  videomixer: Don't do floating point math in the inner processing loop for I420 blending

2009-12-10 18:43:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle NULL and empty transport strings
	  When an RTSP extension returns NULL or an empty transport string, just ignore it
	  and try to get the next possible transport. Fixes playback of RealMedia streams.

2009-12-10 18:42:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: install event function on internal RTCP pad
	  Install a custom event function on the internal RTCP pad so that we can reply
	  TRUE to a latency event.

2009-12-10 10:48:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_rgb.c:
	  videomixer: Remove wrong comments, copied from the I420 blend function

2009-12-09 21:15:07 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: The queued duration is a signed integer
	  ...and it will really be negative sometimes.

2009-12-09 21:03:57 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Only pop buffers from collectpads after they're fully consumed
	  This decreases latency and memory usage because new buffers are only
	  accepted by collectpads if there's no queued buffer.

2009-12-09 20:42:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: Clean up position/duration handling
	  Also use the last end time for closing the segment, not the
	  start time of the last buffer.

2009-12-09 16:50:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Close the segment on EOS if the real duration is known

2009-12-09 16:46:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Update duration if current buffer is already after the old duration

2009-12-09 16:43:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Drop buffers that are after segment stop
	  ...and if this happened for all streams go EOS.

2009-12-09 16:41:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix position tracking and sending of filler segments

2009-12-09 16:15:09 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Use gst_util_uint64_scale_int() for fps to seconds per frame calculations

2009-12-08 17:34:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Keep the segment stop position for update newsegment events

2009-12-04 14:42:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/Makefile.am:
	* ext/ladspa/Makefile.am:
	* ext/ladspa/gstladspa.c:
	* ext/ladspa/gstladspa.h:
	* ext/ladspa/gstsignalprocessor.c:
	* ext/ladspa/gstsignalprocessor.h:
	* ext/ladspa/load.c:
	* ext/ladspa/search.c:
	* ext/ladspa/utils.h:
	  ladspa: Remove the sources from gst-plugins-good
	  It's disabled anyway and the latest version of it is in
	  gst-plugins-bad. Fixes bug #603779.

2009-12-04 13:50:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: init current_entry in push mode
	  Set the current_entry to 0 (instead of -1) in push mode so that we correctly
	  calculate the current frame number and timestamp.
	  Add some more debug info and fic the duration debug.

2009-12-04 11:14:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix major memory leak when playing back rtsp video streams
	  Don't forget to unref QoS, navigation and latency events when
	  dropping them.

2009-12-03 08:58:08 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: only send pending tags with newsegment events
	  Send pending tags only from the streaming thread, just after we've sent
	  the newsegment event, not with e.g. flush-start. This not only does the
	  right thing, but also makes sure we're not trampling over variables set
	  up in the streaming thread from the seeking thread in case someone tries
	  to issue a seek just as the demuxer is parsing the headers.
	  Fixes #601617. Spotted by Ognyan Tonchev.

2009-12-03 17:49:55 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix debug message printf args
	  Fixes debug message printf format to make it build in mac's gcc

2009-12-02 13:33:20 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/shout2/gstshout2.c:
	  shout2: Convert delay correctly
	  Use GST_MSECOND to convert delay in msecs to nanosecs
	  Fixes #603547

2009-12-02 11:21:22 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  lame: Avoid crash when seeking before negotiating
	  lame's 'lgv' variable is only initialized when the caps
	  is negotiated, whenever a seek happens before that, it would
	  attempt to call a function on an empty pointer, causing the crash.
	  Fixes #603515

2009-12-01 19:24:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: reset segment info after flush
	  Reset the segment info after a flush. We use the segment for handling QoS and if
	  we don't reset the segment, QoS is basically disabled after a flushing seek.

2009-12-01 15:07:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 87bf428 to 47cb23a

2009-12-01 14:15:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From da4c75c to 87bf428

2009-11-30 15:59:50 +0100  Aurelien Grimaud <gstelzz at yahoo dot fr>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: avoid buffer ref/unref pairs for CSRCs
	  We ref the buffer before pushing it downstream in order to get the CSRCs of it
	  after pushing. This causes performance problems when downstream elements want to
	  change the metadata because the buffer needs to be subbuffered.
	  Instead, read and store the CSRCs of the buffer in an array before pushing it
	  and process the array after pushing the buffer. This allows us to remove the
	  ref/unref pair.
	  Fixes #603376

2009-11-28 19:23:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2: use gstpoll for timeouts
	  Use our own GstPoll based timeout instead of the shout sleep so that we can
	  interrupt when doing a state change and shutting down.
	  Fixes #602887

2009-11-28 12:25:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  check: fix jitterbuffer check
	  Make sure we set a base_time on the element.
	  Fix the timeout to at least twice the jitterbuffer latency.
	  Enable previously failing tests.
	  Remove impossible checks.

2009-11-27 18:55:20 +0100  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From 53a2485 to da4c75c

2009-11-26 16:14:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: optionally merge NALUs into Access Units
	  ... which may be expected/desired by some downstream decoders
	  (and spec-wise highly recommended for at least non-bytestream mode).

2009-11-26 17:29:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix timestamp datatype

2009-11-25 10:38:23 -0600  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: avoid using wrong clock-rate
	  Check for a valid clock-rate before attempting to estimate the npt
	  stop time.

2009-11-25 10:37:30 -0600  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix typo in comments

2009-11-25 16:05:10 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffertest: add one more test and file a bug now
	  CHange the backwards test to always send first buffer first to have a define
	  basetime. Add another test that sends buffers backwards to assert that only
	  first sent buffer is keep and used as basetime. Disabled those tests still,
	  as its not passing/failing consitently and file a bug for jitterbuffer.

2009-11-25 10:17:34 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/elements/rtpjitterbuffer.c:
	  jitterbuffertest: improve the test
	  the tests are a bit more solid now but still not produce reliable results.
	  Wonder if they are still flawky or if its a bug in jitterbuffer.

2009-11-24 11:13:06 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: return error message on windows too.

2009-11-24 10:58:49 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: first phase of fixing up error reporting for windows.

2009-10-30 03:13:54 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: also set the suggested buf size for audio
	  We were only setting the suggested buf size for video,
	  we can set it for audio as well.
	  This and 195e14529d80ef318ce3a778c1995efb11f266cd
	  fix an issue that prevented seeking on large avi files
	  on WMP (non-recent versions).

2009-11-04 16:10:23 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avimux: fix indx duration for PCM audio
	  GstBuffers for PCM audio usually contains more than
	  1 sample, we need to get the total number of samples to set
	  the indx duration.

2009-11-04 16:04:10 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: Audio buffers should be picked earlier
	  Adds a 0.5s advantage for audio buffers to being
	  picked earlier for muxing.

2009-11-24 16:40:19 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix push mode by making sure stbl information is available in next_entry_size ()

2009-11-24 16:35:20 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix order of arguments in log message

2009-11-24 15:51:21 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: fix spelling in comment

2009-11-23 17:58:17 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* common:
	  build system: Fix wrongly committed change to common/

2009-11-10 10:26:07 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Ease debugging by removing a goto for an error message

2009-11-14 15:52:09 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* common:
	* gst/qtdemux/qtdemux.c:
	  qtdemux: Parse per sample rather than all at once but build complete index when seeking

2009-11-04 17:31:15 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Save atom data for later use so it doesn't get freed after initial parsing

2009-11-06 11:00:04 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Parse from the previously parsed sample up to sample n

2009-11-04 17:04:22 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Make qtdemux_parse_samples () parse up to n samples

2009-10-28 17:49:02 +0000  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Separate off stbl sub-atom initialisation

2009-10-26 22:42:36 +0000  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Move variables into context in preparation for refactorisation

2009-10-26 20:36:08 +0000  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix bug where stps is never parsed due to logic error

2009-11-04 17:31:15 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Port ctts from Gnode * to GstByteReader

2009-10-23 13:06:44 +0100  Robert Swain <robert.swain@gmail.com>

	* gst/qtdemux/qtatomparser.h:
	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_dump.c:
	* gst/qtdemux/qtdemux_dump.h:
	* gst/qtdemux/qtdemux_types.h:
	  qtdemux: Switch from QtAtomParser to GstByteReader

2009-11-23 12:53:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix typo and grammar

2009-11-22 19:30:58 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/dtmf/Makefile.am:
	  Clean up LDFLAGS, LIBS, CFLAGS
	  Fix order, fix variables that don't exist, like GST_LIBS_LIBS,
	  use $(LIBM) instead of -lm, and move _LIBS from LDFLAGS to LIBADD.
	  Spotted by Havard Graff.

2009-11-20 10:31:47 -0500  Olivier Crête <tester@tester.ca>

	* gst/dtmf/tone_detect.h:
	  dtmf: Use _stdint.h from configure
	  https://bugzilla.gnome.org/show_bug.cgi?id=602465

2009-11-20 10:30:00 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix typo in mode enum description

2009-11-20 11:25:49 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpbin.c:
	  docs: more links and better short description
	  Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change
	  the short description to be more meaningful.

2009-11-20 09:58:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/wavpackparse.c:
	  wavpackparse: Fix unit test for recent position reporting changes

2009-11-19 20:33:07 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/tone_detect.c:
	* gst/dtmf/tone_detect.h:
	  dtmf: Update dtmfdetect to make it MSVC friendly
	  https://bugzilla.gnome.org/show_bug.cgi?id=602465

2009-11-19 16:09:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/wavpack/gstwavpackparse.c:
	  wavpackparse: After pushing a frame, update last_stop to the end of the frame
	  This improves position reporting, especially because of the fact that
	  WavPack frames are usually 0.5-1.0 seconds long.

2009-11-19 16:08:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/wavpack/gstwavpackparse.c:
	  wavpackparse: Allow pulling the last WavPack frame of a file
	  Because of a >= instead of a >, that last frame of a WavPack file
	  would never be parsed in pull mode.

2009-11-19 10:30:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 0702fe1 to 53a2485

2009-10-29 08:29:38 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: Add more fields to SVQ3 caps
	  qtdemux only added the whole stsd atom as 'codec_data'
	  in its output caps for SVQ3. This patch makes it add
	  the SEQH (inside a SMI atom) and a gamma field (taken
	  from the gama atom) if available.
	  Fixes #587922

2009-11-18 17:55:42 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Raise rank of muxer to PRIMARY

2009-11-18 17:54:16 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/y4m/gsty4mencode.c:
	  y4m: Raise rank of encoder to PRIMARY

2009-11-18 17:54:02 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/law/alaw.c:
	* gst/law/mulaw.c:
	  law: Raise rank of encoders to PRIMARY

2009-11-12 19:11:18 +0000  Bastien Nocera <hadess@hadess.net>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  Add user-id and user-pw properties
	  So that one doesn't need to modify the URL to have access
	  to authenticated RTSP streams.
	  fixes #601728

2009-11-18 12:22:10 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: use acquired flag when checking valid state
	  Use the acquired field of the ringbuffer in get_time to know when we are in an
	  invalid state. We don't clear the rate flag when releasing the ringbuffer so
	  this values is not usable.
	  Avoids some error messages being posted because the pulseaudio connection is
	  down.

2009-11-18 10:17:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: bump core requirement to 0.10.25.1 as well
	  Make implicit requirement explicit.

2009-11-18 12:53:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix bogus memory chunk size check

2009-11-18 12:01:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: implement some more callbacks
	  Implement some more callbacks for debugging purposes.

2009-11-11 15:50:19 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: release lock before emiting signals
	  Release the jbuf lock before emiting the request-pt-map signal to avoid
	  deadlocks. We also need to catch the shutdown case when locking again.
	  Fixes #593354

2009-11-11 11:59:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvdepay.h:
	  rtp: add BroadcomVoice depayloader

2009-11-11 11:38:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpbvpay.c:
	  rtpbvpay: add rfc reference

2009-11-11 11:37:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpbvpay.h:
	  rtp: add BroadcomVoice payloader

2009-11-09 12:17:34 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: properly finish the ECMA array
	  The ECMA array with the file index was missing a mandatory end marker.
	  Fixes bug #601242.

2009-11-18 02:15:15 +0000  Jan Schmidt <thaytan@noraisin.net>

	* gst/deinterlace/gstdeinterlace.c:
	  Use new still-frame API from gst-plugins-base

2009-11-18 02:14:46 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	  Bump gst-plugins-base requirement to 0.10.25.1

2009-11-17 17:59:13 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: identify IMA adpcm in qt properly.

2009-11-18 01:27:37 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* win32/common/config.h:
	  Back to development -> 0.10.17.1

2009-11-17 01:53:08 +0000  Jan Schmidt <thaytan@noraisin.net>

	* gst-plugins-good.doap:
	  Add release 0.10.17 to the doap file

=== release 0.10.17 ===

2009-11-17 01:25:30 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  Release 0.10.17

2009-11-17 00:18:22 +0000  Jan Schmidt <thaytan@noraisin.net>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2009-11-13 02:07:25 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	* win32/common/config.h:
	  0.10.16.3 pre-release

2009-11-10 11:52:24 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Make sure to initialize variables before using them

2009-11-09 20:06:03 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ChangeLog:
	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	* win32/common/config.h:
	  0.10.16.2 pre-release

2009-11-09 15:20:00 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: free temporary buffer when changing state to NULL
	  Free temporary allocations in the state change function and not
	  only when the object is finalised.

2009-11-09 11:40:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: only allocate as much temporary memory as needed for indirect decoding
	  When we can't decode directly into the output buffer, make our temp buffers
	  only as big as needed instead of allocating for the worst case scenario (well,
	  we still alloc more than strictly needed for some cases, but significantly
	  less than before).

2009-11-05 23:46:58 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: printf format fix

2009-11-05 23:44:27 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/raw1394/gst1394clock.c:
	* ext/raw1394/gsthdv1394src.c:
	  raw1394: printf format fixes

2009-11-05 23:40:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: printf format fix

2009-11-04 22:19:58 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/Makefile.am:
	* gst/dtmf/gstdtmf.c:
	* gst/dtmf/gstdtmfdetect.c:
	* gst/dtmf/gstdtmfdetect.h:
	* gst/dtmf/tone_detect.c:
	* gst/dtmf/tone_detect.h:
	  dtmfdetect: Add DTMF tone detector
	  It looks at raw audio data and emits messages when DTMF is detected.
	  The dtmf detector is the same Goertzel implementation used in FreeSwitch
	  and Asterisk. It is in the public domain.

2009-11-05 12:13:44 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: do not write empty INFO list
	  avoid writing an empty INFO list chunk, both because
	  it is useless and because vlc refuses to play the
	  resulting file.

2009-11-05 10:54:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Notify about band property changes caused by changing number of bands

2009-11-05 10:45:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer.h:
	* gst/equalizer/gstiirequalizernbands.c:
	  equalizer: Make changes to band properties and the number of bands threadsafe

2009-11-05 10:30:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Fix stupid off by two bug

2009-11-05 08:18:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Add band property to select the band filter type
	  This allows per band configuration of a peak, low shelf or
	  high shelf filter, which can be very useful if the band frequencies
	  and widths are manually configured.

2009-11-05 08:17:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Fix code style

2009-11-05 08:03:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/equalizer/gstiirequalizernbands.c:
	  equalizer: Some cleanup

2009-11-04 22:21:35 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  dtmfsrc: Reject empty caps

2009-11-04 22:21:22 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  dtmfsrc: Use log level for repeated debug messages

2009-11-04 20:05:17 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  dtmfsrc: Allow for any samplerate

2009-10-07 09:31:19 -0400  Gabriel Millaire <gabriel.millaire@collabora.co.uk>

	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	  celtpay/depay : change GST_DEBUG_OBJECT to GST_LOG_OBJECT in pay_handle_buffer and depay_process

2009-10-02 17:04:43 -0400  Gabriel Millaire <gabriel.millaire@collabora.co.uk>

	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltdepay.h:
	* gst/rtp/gstrtpceltpay.c:
	  celtpay/depay: Negotiate parameters through caps
	  celtdepay : added default framesize(480) channels(1) and clockrate(32000)
	  depay_setcaps : now gets channels and framesize from string with default value
	  depay_process : now adds timestamp to outbuf
	  Added frame_size to GstRtpCeltDepay
	  Changed some GST_DEBUG to GST_DEBUG_OBJECT or GST_LOG_OBJECT
	  celtpay : getcaps : gets channel and framesize and sets caps
	  Added frame-size to static caps for audio/x-celt

2009-11-04 15:58:34 +0000  Jan Schmidt <thaytan@noraisin.net>

	* gst/deinterlace/Makefile.am:
	  deinterlace: Pull in CFLAGS and LIBS flags from -base before core before system.

2009-10-15 16:33:24 +0100  Jan Schmidt <thaytan@noraisin.net>

	* po/Makevars:
	  po: Don't create backup .po files
	  As well as preventing creation of useless backup files, it works
	  around a bug in gettext 0.17 on OS/X

2009-11-04 16:47:42 +0100  Edward Hervey <bilboed@ihatesteve.local>

	* gst/qtdemux/qtdemux_dump.c:
	  qtdemux: init variables to make compiler on osx build bot happy

2009-11-03 16:04:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux_dump.c:
	  qtdemux: init variables to make compiler on osx build bot happy

2009-11-03 17:35:15 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: don't allocate big arrays on the stack
	  Add the arrays to the instance data and allocate on first use.

2009-11-01 15:57:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: remove pointless call to gst_element_no_more_pads()

2009-11-01 00:29:57 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: fix decay to be smooth
	  The length not having any fractional part as it was promoted to gdouble after
	  dividing two guint64.

2009-11-01 00:29:24 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	  level: calculate the message-intervall when it changes

2009-11-01 00:14:08 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: clocktime is a guint64, use right macro to init fields

2009-11-01 00:10:01 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: use more g-style types

2009-10-30 09:27:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	  pulsesink: Only set the volume on stream connection if pulse >= 0.9.20 is available
	  In older versions the volume set during stream connection had
	  no defined sematic and usually it was a relative volume. What
	  was needed for our use case is an absolute volume though, otherwise
	  the volume will be always decreased on stream connection if it's
	  less than 100%.
	  Since pulse 0.9.20 that volume is always an absolute volume if
	  flat volumes are used and relative otherwise, which is the same
	  as for pa_context_set_sink_input_volume().
	  Relevant pulse changesets:
	  http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=f27a50691c8fe45bac7dd6b21fac91a359def3a1
	  http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=2501687579e359d5032a4d165b2ffc8f5b1b8ba6

2009-10-27 18:07:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: use segment_full when we can
	  Use segment_full so that we can pass the applied rate to the segment values. We
	  will change the applied rate when we implement skip mode.

2009-10-18 00:16:06 +0100  Robert Swain <robert.swain@gmail.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Fix buffer offset by moving length incrementation

2009-10-23 18:31:14 -0700  Michael Smith <msmith@songbirdnest.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: Create the video NSView in READY->PAUSED rather than NULL->READY

2009-10-23 18:28:22 -0700  Michael Smith <msmith@songbirdnest.com>

	* sys/osxvideo/Makefile.am:
	  osxvideo: explicitly link to GST_LIBS

2009-10-23 18:09:43 -0700  Michael Smith <msmith@songbirdnest.com>

	* gst/avi/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/wavparse/Makefile.am:
	  Add dependencies of gstriff to things that link to gstriff, needed on Win32.

2009-10-23 17:25:17 -0700  Michael Smith <msmith@songbirdnest.com>

	* tests/examples/rtp/client-PCMA.c:
	* tests/examples/rtp/server-alsasrc-PCMA.c:
	  rtp examples: remove executable bits from C files.

2009-10-23 11:21:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: disable all jitterbuffer tests for now
	  Since even the one enabled seems to fail.

2009-10-22 13:39:58 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: also include the new test for prev commit

2009-10-22 13:19:07 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	  tests: add a jitterbuffer test
	  Tests pushing a few buffers in various order and asserting the order sent by the
	  jitterbuffer. Contains two disabled tests that need more work.

2009-10-22 12:30:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Dirac "muxing" units end on EOS too
	  A Dirac muxing unit are all non-picture, non-end-of-sequence
	  packets up to and including the first picture or eos packet.
	  See http://www.diracvideo.org/wiki/index.php/ContainerFormatMappingGuidelines

2009-10-22 02:09:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix compilation with debugging disabled
	  total_idx is always evaluated.

2009-10-19 21:59:46 +0300  Priit Laes <plaes@plaes.org>

	* ext/libcaca/gstcacasink.h:
	  cacasink: minor cleanups for header.
	  Use G_BEGIN_DECLS macros, remove unused variables and fix typo.
	  See #599018.

2009-10-19 21:59:23 +0300  Priit Laes <plaes@plaes.org>

	* ext/libcaca/gstcacasink.c:
	  cacasink: exit properly when invalid driver has been selected.
	  See #599018.

2009-10-20 18:23:28 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Stop scanning at the last entry... and not the one before :)
	  This ensures we actually push out everything

2009-10-20 17:20:55 +0200  Andy Wingo <wingo@oblong.net>

	  qtdemux: unpack more information into image/x-j2c caps
	  * gst/qtdemux/qtdemux_fourcc.h: Add new fourccs for use by the mj2
	  unpacker.
	  * gst/qtdemux/qtdemux.c (qtdemux_parse_trak): Unpack JPEG2000 component
	  mapping and channel definitions from the jp2h header. Will add
	  component-map and channel-definitions elements to the caps if the
	  component maps or channel definitions are nonstandard, where standard
	  order means RGB, 444 packed YUV, or greyscale, with no alpha channel.
	  Fixes #598915.

2009-10-20 17:33:41 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/check/elements/deinterleave.c:
	  tests: include stdio.h for sscanf

2009-10-19 15:21:57 +0100  Bastien Nocera <hadess@hadess.net>

	* ext/pulse/pulsesink.c:
	  Fix the StreamVolume interface not being advertised
	  gst_pulsesink_interface_supported() was missing a check for it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=598933

2009-10-16 21:14:14 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: code cleanup
	  Use gdouble instead of double. Calculate falloff_time once instead of twice.

2009-10-18 15:52:02 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: MEMDUMP the junk blobs
	  It will only actually pull the junk blobs from upstream if the memdump
	  level is activated

2009-10-18 15:51:34 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Some avi files have INFO lists in the headers.

2009-10-18 16:02:01 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Don't seek on empty streams

2009-10-18 15:50:39 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Ensure _calculate_durations_from_index only uses valid streams

2009-10-18 15:49:29 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Only call convert function if we have strf.auds

2009-10-18 15:48:06 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Use first indexed stream for seeking.
	  In the future, main_stream can be adjusted to contain the optimal stream
	  as mentionned in the FIXME line 3440

2009-10-18 15:46:48 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Only expose streams that actually have something in it.
	  This guarantees that in pull-mode, all streams have a valid index to
	  work with.

2009-10-18 15:40:37 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Properly mark presence of index.
	  Instead of blindly saying we have an index, only do so if we have a
	  non-empty index.

2009-10-17 02:18:53 +0200  Lennart Poettering <lennart@poettering.net>

	* ext/pulse/pulsesink.c:
	  pulse: never apply volume more than once
	  Generally decisions on the volume of the stream should be done inside of
	  PA, not inside of Gst. Only PA knows how volumes translate between
	  devices and s on.
	  This patch makes sure that all volumes set via the volume property are
	  only applied *once* to the underlying stream. After applying them the
	  client side will not store them anymore. This should make sure that
	  really only user-triggered volume changes are forwarded to server, but
	  the client never tries to save/restore the volume internally.
	  Fixes bug #595231.

2009-10-17 08:55:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/plugin.c:
	  pulsesink: Initialize gettext for the translated strings in plugin_init()

2009-10-17 00:10:30 +0200  Lennart Poettering <lennart@poettering.net>

	* ext/pulse/pulsesink.c:
	  pulse: use 'performer' as a fallback for 'artist' tag

2009-10-17 00:09:36 +0200  Lennart Poettering <lennart@poettering.net>

	* ext/pulse/pulsesink.c:
	* po/POTFILES.in:
	  pulse: when constructing a stream title from tag data make sure it is translatable

2009-10-17 00:06:15 +0200  Lennart Poettering <lennart@poettering.net>

	* ext/pulse/pulsemixerctrl.c:
	  pulse: loop while connecting to server
	  pthread does not guarantee that there are no spurious condition variable
	  wakeups, neither does pa_threaded_mainloop_xxx() which is a wrapper
	  around it. So we need to loop around the _wait() function to make sure
	  we get the right wakeup.
	  Also, unify the order of the wait loops across the file.

2009-10-17 00:05:10 +0200  Lennart Poettering <lennart@poettering.net>

	* ext/pulse/pulsemixerctrl.c:
	* ext/pulse/pulseprobe.c:
	  pulse: mainloop creation can fail too, so handle that

2009-10-17 00:03:06 +0200  Lennart Poettering <lennart@poettering.net>

	* ext/pulse/pulsemixerctrl.c:
	  pulse: adjust CHECK_DEAD_GOTO macro to glib style

2009-10-16 17:28:42 +0200  Lennart Poettering <lennart@poettering.net>

	* ext/pulse/pulsemixerctrl.c:
	* ext/pulse/pulsemixerctrl.h:
	* ext/pulse/pulseprobe.c:
	* ext/pulse/pulseprobe.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.h:
	  pulse: make a few things smaller by making them bitfields

2009-10-16 17:26:41 +0200  Lennart Poettering <lennart@poettering.net>

	* configure.ac:
	  pulse: bump minimum libpulse version to 0.9.10
	  Older versions than 0.9.10 are really really old and buggy. Drop
	  compatibility with them. Nobody should run anything that old.
	  Also see: https://bugzilla.gnome.org/show_bug.cgi?id=595029

2009-10-16 18:18:31 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/debugutils/gstdebug.c:
	  debugutils: register pushfilesrc element

2009-10-16 17:28:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avimux: support (some) VBR audio muxing
	  AVI format can handle VBR audio provided audio chunks are of fixed duration
	  (cfr fixed duration video frames).  Apply this approach to (always) parsed
	  raw AAC and (if parsed) to MPEG-1/2 audio.
	  See #368681.

2009-10-16 13:41:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix branch hints
	  Remove inappropriate branching hints and add some new ones.

2009-10-16 12:33:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix regression in indirect decode path
	  Revert variable name back to what it was before the G_LIKELY was
	  added (in commit 69c24fb9). The code works better that way.

2009-10-16 02:47:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix regression with certain formats
	  Fix regression introduced by previous commit (#598517).

2009-10-15 19:49:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: don't use decompress structure members we shouldn't be using

2009-10-14 17:53:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.h:
	  jpegdec: remove some unused members from jpegdec instance structure

2009-10-16 11:53:38 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/Makefile.am:
	* gst/udp/Makefile.am:
	  build: use gst-glib-gen.mak to fix the glib build rules.
	  The build rules in glib-gen.mak were using pattern rules in a non save way.

2009-10-16 10:15:35 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 85d1530 to 0702fe1

2009-10-15 21:04:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: adjust flow return aggregation to updated loop_data
	  In particular, each stream is now treated separately, and one stream's
	  EOS should not lead to overall EOS.

2009-10-15 11:52:35 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: check some more atom sizes prior to parsing

2009-10-15 13:19:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: handle events in TCP mode
	  We need to handle events in TCP mode so that we can reply to the LATENCY event
	  with TRUE.

2009-10-15 11:24:45 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: add missing argument in debug message

2009-10-14 18:58:06 +0200  Marvin Schmidt <marv@exherbo.org>

	* tests/check/elements/flvmux.c:
	  flvmux: Use loop test to prevent timeout on slow machines
	  Partially fixes bug #597739.

2009-10-14 16:15:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: forward events into the rtpbin
	  Only catch the SEEK event on the srcpad and let other events enter the rtpbin.

2009-10-14 11:33:24 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix late tags finding
	  Use the correct taglist variable when notifying of late tags.

2009-10-14 13:09:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: use GstIndex for (limited) seeking in push mode
	  ... but disable this for now.  Although it basically works fine,
	  user experience might be shaky (depending on taste), since there
	  is no keyframe info in push mode.

2009-10-14 13:08:47 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: add GstIndex support

2009-10-14 11:55:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: also determine duration in push mode

2009-10-14 11:54:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: add GstIndex support

2009-10-14 07:38:26 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Set duration on buffers
	  Use framerate to estimate duration of buffers.
	  Fixes #590362

2009-10-14 12:28:55 +0200  Håvard Graff <havard.graff at tandberg.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: only forward the lost-event to the last seen pt-number
	  forward all events on all pads except for the PacketLost event, which we want to
	  forward to the last seen pt pad.
	  Fixes #598377

2009-10-06 22:28:50 +0300  René Stadler <mail@renestadler.de>

	* ext/pulse/pulsesink.c:
	  pulsesink: set desired minreq value to segsize/latency-time
	  If we let the daemon decide freely by passing -1, we end up always getting 20ms.
	  We want to set this value because in some cases we want to select a higher
	  latency-time in order to save power.
	  Fixes #597601

2009-10-14 10:41:21 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From a3e3ce4 to 85d1530

2009-10-13 18:33:34 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/pipelines/flacdec.c:
	  tests/pipeline/flac: Fix build on macosx 10.5

2009-10-13 18:19:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: demote some warnings to debug

2009-10-13 17:47:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/avi-ids.h:
	  avi: add new avi flag we might want to use

2009-10-13 17:46:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: calculate suggested buffer size
	  Calculate the suggested buffer size based on the largest chunk in the file.
	  See #597847

2009-10-13 17:45:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: add jpeg2000 to allowed caps

2009-10-13 17:41:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: add debug for the superindex offsets

2009-10-13 16:02:37 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix uninitialized variable warning
	  Fix another bogus may-be-used-uninitialized warning in qtdemux

2009-10-13 13:08:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avi: lower max file size
	  Make a constant of the max file size and lower the value to what ffmpeg does,
	  hopefully improving compatibility with windows media player.
	  See #597847

2009-10-13 01:02:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix bogus warning about discont flag on first buffer
	  The very first buffer should always have the DISCONT flag set, no
	  need to warn about that. Only warn if we get a DISCONT buffer in
	  non-packetised mode and we already have some data.

2009-10-13 00:41:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix crash for unusual vertical chroma subsampling factors
	  Fixes #597351.

2009-10-13 00:12:42 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix uninitialized variable warnings
	  The gcc on the OS/X buildbot complains about these variables not being
	  initialized, even though they can't possibly actually be used
	  uninitialized.

2009-10-11 11:35:23 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  dtmf: fix warnings in macosx snow leopard

2009-10-10 00:37:08 +0200  Josep Torra <n770galaxy@gmail.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fixes warning building in snow leopard

2009-10-09 17:12:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: also consider Quicktime text subtitles

2009-10-09 17:02:57 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: provide language tag for stream

2009-10-09 16:30:57 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: refactor common parts in track parsing

2009-10-09 16:21:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: refactor buffer processing and sending
	  ... so it can be used in both pull and push based mode.

2009-10-08 13:39:25 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: extract palette data for dvd subpicture streams
	  ... and send it downstream using custom dvd event

2009-10-07 14:03:17 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: support 3GPP timed text subtitles
	  In particular, also make subtitle support less subp(icture)-centric.

2009-10-07 16:15:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: NULL is not a valid taglist

2009-09-23 17:20:25 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: recognize some more encypted track cases

2009-10-09 15:59:25 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/id3demux/id3tags.c:
	  id3: fixes warnings building on macosx
	  Another round on the formating of that debug line.

2009-10-09 14:44:02 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/id3demux/id3tags.c:
	  id3: cast pointer math results to glong

2009-10-09 14:37:32 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/flac/gstflacdec.c:
	  flac: apparently on some platforms a FLAC__uint64!=guint64

2009-10-09 14:21:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  buikd: explicitely cast, to tell some compilers that this is not long int

2009-10-09 13:38:17 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/flac/gstflacdec.c:
	* gst/id3demux/id3tags.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  build: don't cast, but use the right format specified instead
	  This correct some of the previous macos fixes.

2009-10-09 12:40:47 +0200  Josep Torra <n770galaxy@gmail.com>

	* ext/dv/gstdvdemux.c:
	  dv: fix warnings on macosx

2009-10-09 12:25:19 +0200  Josep Torra <n770galaxy@gmail.com>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	  flac: fix warnings on macosx

2009-10-09 12:19:35 +0200  Josep Torra <n770galaxy@gmail.com>

	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	  annodex: fix warnings in macosx

2009-10-09 12:14:22 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxvideo/cocoawindow.m:
	  osxvideo: fix a warning doing a cast

2009-10-09 12:11:12 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxaudio/gstosxringbuffer.c:
	  osxaudio: fix warnings on macosx

2009-10-09 12:01:10 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: fix warning on macosx

2009-10-09 11:57:59 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: fix warning on macosx

2009-10-09 11:54:03 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix warnings building on macosx

2009-10-09 11:42:36 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/id3demux/id3tags.c:
	  id3demux: fix printf warnings on macosx

2009-10-09 11:30:00 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix warning in macosx making the format portable

2009-10-09 10:51:29 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofx: use G_GUINT64_FORMAT to fix warnings on OSX

2009-10-09 10:11:38 +0200  Josep Torra <n770galaxy@gmail.com>

	* sys/osxaudio/gstosxringbuffer.c:
	  osxaudio: Fixes build on macosx snow leopard.

2009-10-09 11:34:16 +0200  Pau Garcia i Quiles <pgquiles@elpauer.org>

	* sys/v4l2/gstv4l2object.h:
	  v4l2: Include sys/ioctl.h for the V4L ioctl requests
	  Old videodevice2.h kernel headers used ioctl stuff without
	  including ioctl.h, making compilation fail on older systems.
	  Note: Including ioctl.h here is only a workaround for old kernel
	  headers, should be removed once everybody has new enough headers.
	  Fixes bug #597867.

2009-10-09 00:14:07 +0100  Jan Schmidt <jan.schmidt@sun.com>

	* configure.ac:
	* tests/check/elements/level.c:
	  check: Make the level unit test succeed on Solaris 10
	  Add a configure check for functional isinf() and fpclass(), and
	  use fpclass() where possible when isinf() is not available.

2009-05-16 13:52:50 +0300  René Stadler <rene.stadler@nokia.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix strstr() usage on possibly unterminated string

2009-10-08 16:16:14 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/Makefile.am:
	* tests/check/elements/level.c:
	  check: Link against LIBM and include math.h for isinf()

2009-10-07 21:51:38 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/oss/gstossaudio.c:
	  oss: Downgrade the rank of osssrc to SECONDARY
	  which is the same rank as osssink has.
	  Fixes bug #597730.

2009-10-08 10:59:53 +0100  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Automatic update of common submodule
	  From 19fa4f3 to a3e3ce4

2009-10-08 10:20:09 +0100  Jan Schmidt <jan.schmidt@sun.com>

	* gst/avi/gstavidemux.c:
	* gst/wavparse/gstwavparse.c:
	  avi/wav: Fix some compiler warnings about incompatible pointers.

2009-10-05 17:36:55 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/multifile/gstmultifile.c:
	  multifile: Fix plugin description

2009-10-07 14:03:20 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/annodex/gstcmmlutils.c:
	* ext/jpeg/gstjpegdec.h:
	* ext/jpeg/gstjpegenc.h:
	* gst/apetag/gstapedemux.c:
	* gst/debugutils/tests.c:
	* gst/id3demux/id3v2frames.c:
	* gst/qtdemux/qtdemux.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtsp/gstrtpdec.c:
	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	* tests/examples/spectrum/spectrum-example.c:
	  build: fprintf, sprintf, sscanf need stdio.h

2009-10-07 00:33:49 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: use shelfing filters for first and last band
	  Refactor the filter setup. Add two new filters with shelf characteristics for
	  first and last band. Change gain calculation as recommended in the quoted
	  document (no qrt needed). Rename variables to match the formulas in the
	  document.

2009-10-02 23:51:29 +0300  René Stadler <mail@renestadler.de>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: guard fragment size with a lower limit based on latency-time
	  In case that the pulse daemon runs the source device at a relatively low fixed
	  fragment size compared to the requested latency-time, configure the ring buffer
	  segsize to the largest integer multiple of the fragment size that is still
	  smaller than or equal to the requested latency-time.
	  Fixes bug #597463.

2009-10-06 17:40:47 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: comment/logging cleanups and more branch guides

2009-10-05 22:43:11 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: fix filter history usage. Fixes #597397
	  The process functions where overwriting the history for each channel. Also pull
	  some static things out of the inner loop.

2009-10-05 16:07:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: use locking around the sessions

2009-10-05 11:46:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: make sure compatible brands buffer exists before dereferencing it

2009-10-04 21:59:24 +0200  Robert Swain <robert.swain@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix printf warnings on OSX
	  Cast variables passed to printf to avoid warnings about incorrect
	  formats (most likely caused by sizeof returning a size_t).
	  Fixes #597348.

2009-10-02 00:23:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: remove internal genre table
	  No need to maintain our own genre table in qtdemux. The genres are
	  identical to the ID3 genres, so we can just use libgsttag's
	  gst_tag_id3_genre_get() to look them up.

2009-10-03 17:18:28 +0200  Robert Swain <robert.swain@gmail.com>

	* gst/avi/gstavidemux.c:
	  Fix printf formats to avoid warnings in avidemux. Fixes #597214
	  https://bugzilla.gnome.org/show_bug.cgi?id=597214

2009-10-03 09:52:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Change one GST_WARNING to a GST_DEBUG

2009-10-02 14:37:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvparse.c:
	  flvdemux: If there's no audio stream after 6 seconds of video signal no-more-pads
	  ...and the other way around. Also ignore any audio/video streams that appear
	  after no-more-pads.
	  Fixes bug #597091.

2009-10-02 14:37:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvparse.c:
	  flvdemux: Make sure to only signal no-more-pads a single time

2009-10-02 22:55:45 +0300  René Stadler <mail@renestadler.de>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulse: rename pa_buffer_attr variables
	  Makes it much easier to see what is going on and is a lot less error prone.

2009-10-02 18:25:16 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtp/gstrtpjpegdepay.c:
	  rtp: add missing include to fix the build

2009-10-02 13:15:59 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	  videofilter: add G_OBJECT_WARN_INVALID_PROPERTY_ID to property setter

2009-10-02 13:10:44 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: don't give wrong number of fields in the message docs

2009-10-01 12:52:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: cache latency in nanoseconds
	  Cache the latency in nanoseconds units to avoid having to convert the
	  milliseconds value to nanoseconds all the time.

2009-10-01 12:12:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: handle -1 input timestamps
	  Don't try to check a -1 timestamp against the max delay.

2009-10-01 10:54:55 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avi: don't misues perf-category and remove unused ext category
	  The performance category is meant to be used to audit codepaths that lead to bad
	  performance (e.g. copies, conversion that can be avoided).
	  Remove the event category which is not used.

2009-09-16 14:23:24 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay/depay: Demote per-buffer debug messages to log level

2009-09-16 14:16:27 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Don't leak incoming buffers after subbuffering them

2009-09-16 13:57:05 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay/depay: Add debug categories

2009-09-16 13:55:19 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Remove long unneeded define replacement

2009-09-30 18:06:07 +0100  Christian F.K. Schaller <christian.schaller@collabora.co.uk>

	* ext/dv/Makefile.am:
	  Update makefile with missing header file

2009-09-30 18:45:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/examples/rtp/client-H263p-AMR.sh:
	* tests/examples/rtp/client-H263p-PCMA.sh:
	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-PCMA.sh:
	* tests/examples/rtp/server-alsasrc-PCMA.sh:
	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  rtp: Use autoaudio{sink,src} instead of alsa in the examples

2009-09-29 17:51:04 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: don't leak output buffers on decoding errors
	  The setjmp handles libjpeg error. Free the outputbffer if we don't need it.

2009-09-29 00:01:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix 'unused variable' compiler warning when compiling with GST_DISABLE_GST_DEBUG

2009-09-23 14:25:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: small cleanups

2009-09-23 13:57:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: fix timestamping in some audio streams
	  For vbr audio streams we need to use the number of blocks to calculate the
	  timestamps.
	  When the allocation of additional index memory fails, don't throw away what
	  we had before.
	  Various cleanups.

2009-09-23 12:56:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: add support for ODML indexes again

2009-09-22 22:12:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avi: implement index scanning
	  Implement scanning of the file when we can parse the index.
	  Some refactoring of common code.
	  Cleanups and comments.
	  Remove some reimplemented code.
	  Remove index massage code and put a FIXME where we should do something
	  equivalent later.

2009-09-22 18:18:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: fix reverse playback

2009-09-22 17:42:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: fix prev keyframe search and cleanups

2009-09-22 14:51:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: remove code that got converted

2009-09-22 14:44:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avi: more cleanups
	  Remove some duplicate counters.
	  Be smarter when updateing the current the timestamp and offset in the stream
	  because we can reuse previously calculated values when simply go forward one
	  step.
	  Correctly set metadata on outgoing buffers.

2009-09-22 12:35:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: small cleanups

2009-09-22 01:28:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: fix read offset and cleanups

2009-09-21 18:04:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avi: rewrite index playback
	  disable code, start on reimplementing loop based operation.
	  Rewrite the index handling so that all streams use their own index for decoding
	  media.

2009-09-21 15:35:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: add new index parsing code
	  Add a new function and datastructure to parse and hold the index entries on a
	  per stream base. Also avoid doing too much work trying to figure out the
	  timestamps and durations as we can trivially do that later.
	  Less information in the entries makes them 2 times smaller and not doing too
	  much work makes this code about 12 times faster than the regular case.
	  Hook in the new function alongside the existing function for comparison until
	  the rest of the code is updated to handle the new index datastructure.

2009-09-28 16:29:45 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	  jpeg: handle more libjpeg return values, add some more branch hints
	  Also remove unused size variable in _chain().

2009-09-25 19:21:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: some optional QT specified stsd MPEG-4 atoms also apply to H264
	  Fixes #596319.

2009-09-25 16:40:31 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: only send tag events downstream after newsegment

2009-09-25 14:14:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: if transport protocol unsupported, try another one
	  Also change error message to more accurately reflect cases in which
	  it can occur.

2009-09-25 11:54:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: add durations modulo 1<<32
	  For calculating the durations of each sample, we are supposed to add each
	  duration modulo 1<<32 so make the elapsed time counter a uint32.
	  Fixes #595942

2009-09-24 20:38:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: small cleanup

2009-09-24 19:33:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtatomparser.h:
	  qtdemux: don't use core API that doesn't exist yet
	  There's no gst_byte_reader_has_remaining() yet. Fixes build.

2009-09-24 13:20:50 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtatomparser.h:
	  qtdemux: map some atomparser functions to their new bytereader equivalents
	  Now that GstByteReader has unchecked and inlined variants as well, map
	  atomparser functions to their respective bytereader equivalents.

2009-08-25 12:11:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtatomparser.h:
	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_dump.c:
	  qtdemux: add qt_atom_parser_has_chunks() and fix indentation

2009-08-20 18:21:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: bail out instead of trying to alloc silly index sizes
	  If it looks like we would be allocating a silly size for our sample
	  index, just bail out instead of trying to allocate it. Helps with
	  broken or fuzzed files where we might end up trying to malloc a
	  couple of hundred MBs otherwise.

2009-08-20 16:47:25 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: error out correctly if we don't even have enough bytes for an atom header

2009-08-20 15:39:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: init fourcc to 0 as well to avoid invalid reads when printf'ing error message

2009-08-20 01:39:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtatomparser.h:
	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_dump.c:
	  qtdemux: add qt_atom_parse_has_remaining() to avoid overflows with _get_remaining()

2009-08-20 01:21:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use GstByteReader when parsing tkhd atom

2009-08-19 19:13:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use unsigned ints for node length and do more sanity checking of the atom length

2009-08-19 01:36:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtatomparser.h:
	* gst/qtdemux/qtdemux_dump.c:
	* gst/qtdemux/qtdemux_dump.h:
	* gst/qtdemux/qtdemux_types.h:
	  qtdemux: use GstByteReader for atom dumping and fix a few bugs

2009-08-21 14:21:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: move stco, stts, stss and stps atom parsing over to GstByteReader
	  Make sure we don't read beyond the atom boundary. Note that the code
	  behaves slightly differently in the corner case where there is not
	  enough atom data for the specified number of samples (n_samples_time)
	  in the atom, but still enough data to fill the pre-allocated index of
	  n_samples entries: before we would just stop parsing the stts data
	  and continue, whereas now we will likely error out. This should not
	  be a problem in practice though. We could maintain the old behaviour
	  by doing reads with a size check inside the loop if needed.

2009-06-30 19:51:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use bytereader to parse stsz and stsc atoms
	  Use GstByteReader to parse stsz and stsc chunks, and check size of
	  available data before parsing it, instead of blindly assuming there
	  will be enough data. Fixes crashes with some fuzzed/broken files.

2009-08-15 20:38:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtatomparser.h:
	  qtdemux: add qt_atom_parser_get_offset() and optimise _peek_sub()

2009-07-01 13:49:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/Makefile.am:
	* gst/qtdemux/qtatomparser.h:
	* gst/qtdemux/qtdemux.c:
	  qtdemux: add QtAtomParser, an inlined GstByteReader variant

2009-09-23 17:19:34 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: use proper order for no-more-pads and newsegment and tag sending

2009-09-23 09:50:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: sprinkle a few branch prediction macros

2009-09-22 15:03:20 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* ext/jpeg/gstjpegdec.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvparse.c:
	* gst/id3demux/id3v2frames.c:
	  Fix compile warnings with gcc 4.0.1.

2009-09-22 11:48:50 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Don't get stuck in an infinite loop with Dirac
	  At the end, Dirac streams have an EOS packet with 0 length.
	  Don't ever sit in an infinite loop when processing one. Allows
	  muxing Dirac into mkv to complete successfully.

2009-09-22 11:03:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* .gitignore:
	  Update .gitignore

2009-09-22 11:02:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	  videomixer: fix up Makefile some more
	  Remove CFLAGS from LIBADD and make order of the various CFLAGS and
	  LIBS at least consistent with each other.

2009-09-22 08:02:48 +0200  Brian Cameron <brian.cameron@sun.com>

	* gst/videomixer/Makefile.am:
	  videomixer: Add $(GST_PLUGINS_BASE_LIBS) to LDFLAGS for linking libgstvideo
	  Fixes bug #595897.

2009-09-21 18:09:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: fix timestamps in push mode

2009-09-18 17:26:42 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: add a G_UNLIKELY and put perf-cat log to code path that copies

2009-09-21 12:32:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: add some performance measurements
	  Measure the performance of various index and header parsing steps to the
	  PERFORMANCE debug category.

2009-09-18 11:53:12 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: allow for unknown varying number of frames per buffer
	  In particular, this caters for RTP payloads with multiple frames
	  per packet.

2009-09-18 11:45:06 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: use correct sample size in conversions

2009-09-18 11:43:46 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: fix buffer time and duration for multiple frames per packet

2009-09-18 14:22:02 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avidemux: some logging cleanup to help understanding the index parsing overhead

2009-09-16 13:28:27 -0700  David Schleef <ds@schleef.org>

	* sys/osxaudio/Makefile.am:
	  osxaudio: link against GST_BASE_LIBS

2009-09-15 17:24:24 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Fix adapter leak
	  The adapter would be leaked if it was empty and the data could be pushed out directly.

2009-09-15 10:04:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Don't dereference NULL pointers
	  pa_stream_get_timing_info() can return NULL.
	  Fixes bug #595220.

2009-09-15 10:01:54 +0200  David Henningsson <gnome.web@epost.diwic.se>

	* ext/pulse/pulsesink.c:
	  pulsesink: Don't dereference NULL pointers
	  pa_stream_get_timing_info() can return NULL.
	  Fixes bug #595220.

2009-09-14 16:05:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: handle stream events
	  Handle stream events and request a PAUSE/PLAY state change from the application
	  when we receive a CORK/UNCORK event.

2009-09-13 12:30:34 -0700  David Schleef <ds@schleef.org>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: Add next-file property
	  Add a property to allow control over what event causes a file
	  to finish being written and a new file start.  The default is
	  the same as before -- each buffer causes a new file to be
	  written.  Added is a case where buffers are written to the
	  same file until a discontinuity in the stream.

2009-09-13 15:55:02 -0700  David Schleef <ds@schleef.org>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: Use values from decoder structure directly
	  Don't store the same values in the GstDvDemux.  This
	  fixes a bug where dvdemux would detect a stream as PAL
	  instead of NTSC, and silently parse it wrong.

2009-09-13 12:20:23 -0700  David Schleef <ds@schleef.org>

	* ext/dv/Makefile.am:
	* ext/dv/gstsmptetimecode.c:
	* ext/dv/gstsmptetimecode.h:
	* ext/dv/smpte_test.c:
	  dvdemux: Add code to parse SMPTE time codes
	  Code to convert time codes to/from timestamps and frame numbers.

2009-09-13 12:01:27 -0700  David Schleef <ds@schleef.org>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: Fix detection of new media
	  There are 5 or 6 AAUX source control packs in a frame, and any
	  of them could have REC_ST cleared, indicating a recording start
	  point.  libdv only checks the first.

2009-09-12 19:25:36 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/dv/gstdvdemux.c:
	  dvdemux: Set DISCONT flag on buffers when REC_ST flag is set.
	  Also add a few branch prediction macros

2009-09-12 00:13:04 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/elements/souphttpsrc.c:
	* tests/check/elements/y4menc.c:
	  check: Fix a couple of tests.
	  The souphttpsrc test wasn't compiling. The soup-misc.h header is needed for
	  soup_ssl_supported.
	  Fix the y4menc test to use a 'progressive' header for the test data now that
	  the element outputs correct interlacing info.

2009-09-11 13:32:39 -0700  Michael Smith <msmith@songbirdnest.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: treat a zero-sized data chunk as extending to the end of the file.
	  This fixes playback of some files that don't have a valid data chunk length,
	  apparently some program creates these.

2009-09-11 22:24:47 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2src: add a function pointer for get_frame function and optimize a bit
	  Use a function-pointer for mmap/read, as this can't change during capture. Also
	  sprinkle a few G_LIKELY/UNLIKELY to improve the error-less code path.

2009-09-11 22:15:01 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2: log buffer copies on queue underrun in perf category
	  v4l2src has a slow path where it does buffer-copies when it runs out of queued
	  buffers. Log this to performance category to help monitoring it.

2009-09-11 15:14:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Implement GstStreamVolume interface

2009-09-11 16:09:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Implement mute property

2009-09-11 13:33:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	  gdkpixbufsink: fix docs refering to send-messages

2009-09-11 13:28:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	* gst/spectrum/gstspectrum.h:
	  spectrum: add post-messages property
	  Add a post-messages property and deprecate the less descriptive message
	  property.

2009-09-11 13:20:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.h:
	  pixbufsink: add post-messages property
	  Add post-messages and deprecate send-messages as the former is more
	  descriptive of what actually happens.

2009-09-11 13:12:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: rename silent to post-messages
	  Use the post-messages property name instead of silent as it is more
	  descriptive.

2009-09-11 12:16:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: post messages for each buffer
	  Add a silent property that can be set to FALSE to post messages on the bus for
	  each written file.
	  Do some more cleanups.
	  Add some docs.
	  Fixes #594663

2009-09-09 18:13:29 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: Allocate Boundry structs on the stack instead of the heap to avoid leaks
	  Fixes bug #594691.

2009-09-10 10:28:48 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	  docs: fix gtk-doc warnings

2009-09-10 10:26:23 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	  docs: fix gtk-doc warnings

2009-09-09 17:51:19 -0700  David Schleef <ds@schleef.org>

	* ext/raw1394/Makefile.am:
	* ext/raw1394/gst1394clock.c:
	* ext/raw1394/gst1394clock.h:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gstdv1394src.h:
	  dv1394src: Add a clock based on isochronous cycle counter
	  Partial fix for #169383.

2009-09-09 16:02:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Fix AYUV->I420 conversion
	  For this fix the averaging of the chroma values. It should't be (a/2 + b)/2
	  but just (a + b)/2.
	  Fixes bug #594599.

2009-09-09 16:25:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	  pulsesink: remove ringbuffer reset compensation
	  Remove the code to deal with a ringbuffer reset as this code is now in the base
	  class.
	  Bump the -base requirement as we need the new baseaudiosink code to function
	  properly.

2009-09-09 16:24:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.h:
	  pulsesink: whitespace fixes

2009-09-09 10:27:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2colorbalance.h:
	  whitespace fixes

2009-09-08 19:34:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsemixer.c:
	* ext/pulse/pulsemixerctrl.c:
	* ext/pulse/pulseprobe.c:
	  pulse: small cleanups
	  Add some debug info
	  Fix the state changes

2009-09-08 18:29:35 +0200  Marc-André Lureau <mlureau@flumotion.com>

	* gst/multipart/multipartmux.c:
	  multipartmux: mark data buffer as delta-unit
	  So that multifdsink always start sending header buffer first
	  Fixes #594520

2009-09-08 17:37:15 +0200  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: add ignore-pt parameter
	  Add a parameter 'ignore-pt' that disables creating a gstrtpptdemux module and
	  ghosts the pads of gstrtpjitterbuffer instead of the ones of gstrtpptdemux.
	  Fixes #594490

2009-09-04 13:51:37 +0200  Marvin Schmidt <marvin_schmidt@gmx.net>

	* tests/check/elements/souphttpsrc.c:
	  checks: only run HTTPS test if libsoup has SSL support

2009-09-08 13:59:56 +0200  Håvard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: propagate payload-type-change signal from demuxer
	  fixes #594254

2009-08-31 18:46:25 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: change severity of clock-rate change debug
	  Make log GST_DEBUG under normal circumstances, GST_WARNING otherwise.
	  Fixes #594253

2009-09-08 13:39:31 +0200  Håvard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: avoid throwing reordered buffers with same timestamps
	  When we receive a reordered packet with the same timestamp as the previous one
	  (which can happen for fragmented packets) don't consider the packet as lost but
	  instead wait for the reordered packet to arrive.
	  Switch the warning-level, so that a reordering does not get a warning, only
	  an actual produced lost-packet.
	  Fixes #594251

2009-08-31 21:16:54 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/rtp/gstrtpjpegdepay.c:
	  rtpjpegdepay: add missing math.h include
	  Fixes #594247

2009-09-08 13:30:29 +0200  Arnout Vandecappelle <arnout@mind.be>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix memory leak
	  In gst_rtspsrc_parse_digest_challenge(), rtspsrc does a g_strndup of the auth
	  header items and then passes them to gst_rtsp_connection_set_auth_param()
	  without freeing.
	  Fixes #594133

2009-09-08 13:18:29 +0200  Stig Sandnes <stig.sandnes@tandberg.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: make free_session() remove stream references
	  When receiving a sync-packet, all sessions with the same cname will be compared
	  and synced together. In this process, there could still be references to a
	  session that has been shut down in the meanwhile.
	  This patch makes sure that these references are removed when shutting down a
	  session, so that the syncing can be done safely.
	  Fixes #594283

2009-08-31 18:46:51 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: use locked state on internal bins
	  Set the locked state on internal elements to make sure that they don't change
	  back to another state when shutting down.
	  Fixes #594248

2009-09-07 18:28:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: add support for mpeg formats

2009-09-05 20:51:14 -0700  Zaheer Merali <zaheerabbas@merali.org>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  y4menc: Add interlaced support
	  Fixes #591713
	  Signed-off-by: David Schleef <ds@schleef.org>

2009-08-24 13:42:42 -0700  David Schleef <ds@schleef.org>

	* ext/gconf/gstgconfaudiosink.c:
	* ext/gconf/gstgconfaudiosrc.c:
	* ext/gconf/gstgconfvideosink.c:
	* ext/gconf/gstgconfvideosrc.c:
	* gst/apetag/gstapedemux.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* sys/v4l2/gstv4l2src.c:
	  Remove Ronald Bultje from Authors field
	  Replaced with "GStreamer maintainers
	  <gstreamer-devel@lists.sourceforge.net>" or just removed,
	  depending on the number of other authors.

2009-09-05 10:21:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 00a859e to 19fa4f3

2009-09-04 13:42:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: prevent a spurious debug warning

2009-09-04 09:32:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Define V4L2_FMT_FLAG_EMULATED if it's not defined yet
	  libv4l2 already uses this flag, even on Linux kernel versions
	  before 2.6.32.

2009-09-04 07:10:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Correctly handle NULL GstIndex

2009-09-03 20:40:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Fix stupid typo in last commit

2009-09-03 20:38:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Put emulated formats behind native formats
	  Fixes bug #593764.

2009-09-03 19:37:10 +0200  Laurent Glayal <spglegle at yahoo.fr>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: fix memleak
	  Don't leak the input buffer when the received and expected seqnum are different when
	  in probation.
	  fixes #594039

2009-09-02 15:21:02 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Lock clock_rate variable
	  The priv->clock_rate variable could become -1 between when its checked to not
	  be -1 and when its used, causing an assertion. Fixed by taking the mutex
	  earlier in the chain() function.
	  Fixes #593955

2009-09-03 19:12:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: whitespace fixes

2009-09-03 19:09:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmpapay.c:
	  rtpmpapay: whitespace fixes

2009-09-03 19:08:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: whitespace fixes

2009-09-03 17:33:28 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Avoid unnecessary processing until we have a full picture.
	  This is for non-packetized mode, when we know the upstream size in bytes.

2009-09-03 14:40:20 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/flv/gstflvmux.c:
	  flvmux: fully use tagsetter to manage the tags. Fixes #563221
	  There is no need to manage a separate taglist.

2009-09-03 14:13:43 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/speex/gstspeexenc.c:
	  speexenc: small taglist handling cleanup
	  Don't eventualy leak the list and instead assert (like in other elements).

2009-09-02 23:12:41 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: also guard reseting subscribe callback with ifdefs
	  It is conditionaly set, so do the same when unsetting.

2009-09-01 15:06:46 +0200  Peter Kjellerstedt <pkj@axis.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpmanager: Fixed a copy & paste error

2009-09-01 13:21:23 +0200  Peter Kjellerstedt <pkj@axis.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpmanager: Removed unused variable priv
	  The variable priv was initialized in a lot of functions but then never
	  used for anything.

2009-09-01 13:03:57 +0200  Peter Kjellerstedt <pkj@axis.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpmanager: A little clean up
	  Make the code flow of gst_rtp_session_send_rtcp() and
	  gst_rtp_session_sync_rtcp() identical.

2009-09-01 12:47:51 +0200  Peter Kjellerstedt <pkj@axis.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpmanager: Make sure that used caps are not freed already (take 2)
	  This reintroduces the fix for bug #593391. It also applies it in
	  gst_rtp_session_sync_rtcp() which has very similar code to
	  gst_rtp_session_send_rtcp().

2009-09-01 12:41:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.h:
	  jitterbuffer: make sure time does not go backwards
	  When we construct a timestamp that would result in a timestamp that is earlier
	  than when the packet was received, reset the skew calculation as this is
	  probably a sign that the sender restarted or paused.
	  Fixes #593354

2009-09-01 11:32:41 +0200  Peter Kjellerstedt <pkj@axis.com>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpmanager: Set caps in gst_rtp_session_send_rtcp() correctly again
	  The test for when to set an RTCP caps on the output pad in
	  gst_rtp_session_send_rtcp() accidentally got inverted in the last commit.

2009-09-01 10:26:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Add support for QCELP audio
	  Fixes bug #593757.

2009-08-31 18:10:11 +0200  Peter Kjellerstedt <pkj@axis.com>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstvertigo.c:
	  effectv: Fix compilation with gcc 3
	  Recent changes in gst-plugins-good/gst/effectv prevents it from being compiled
	  with gcc 3. The problem is that the new code uses preprocessor conditionals
	  within a macro call which does not work with older versions of gcc.
	  Fixes bug #593688.

2009-08-31 16:20:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  docs: small clean-ups in -sections.txt
	  Remove duplicate entry for warptv; there is no taglibmux element.

2009-08-27 15:46:52 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: consider (optional) auxiliary data when parsing

2009-08-27 15:46:15 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gdepay.h:
	  rtpmp4gdepay: handle broken AU-Index in non-interleaved streams
	  In case of non-interleaved (= sequentially payloaded) streams,
	  the AU-Index serves little purpose (that is not already covered by
	  RTP fields).  (Broken) Payloaders might consider this field then
	  to be disregarded and have non spec compliant values, e.g. each
	  RTP packet having AU-Index 2 (rather than 0).  As such, ensure/force
	  simple sequential sending of non-interleaved streams.

2009-08-18 17:17:28 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: also extract ftyp info in push mode

2009-08-13 16:11:59 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: consider 3gpp style tag parsing in some more cases
	  3GPP specs define a number of tags along with precise layout. While these
	  are normally expected to be found in a container whose major brand is a
	  3GPP brand, this may also happen when a 3GPP brand is only mentioned as a
	  compatible brand.  Apply some checks, heuristic and fallbacks to extract
	  such tags as well.

2009-08-11 13:56:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: reflow exit, and fix some leaks

2009-08-11 13:54:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: push mode; add pad if needed so downstream gets EOS

2009-08-10 16:19:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  wavparse: push mode; fix/improve chunk handling
	  Handle large, invalid or otherwise unusual chunk sizes.
	  Verify some chunk sizes to be at least the size they are
	  expected to be and round up some sizes to even number for
	  e.g. offset administration, which must also be properly
	  tracked in push mode.

2009-08-08 21:54:00 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: push mode; cater for unusual chunk sizes

2009-08-31 16:34:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: use proper locking for pads and caps
	  Use the sesion lock and shotdown variable to protect and ref the pads we are
	  going to push on.
	  fixes #561825

2009-08-31 16:33:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: whitespace fixes

2009-08-31 13:38:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: clean up adapter properly
	  Reflow code so we don't try to clear or re-use an already-freed adapter.

2009-08-31 13:07:53 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflactag.c:
	* gst/wavparse/gstwavparse.c:
	  flactag, wavparse: GstAdapter is not a GstObject

2009-08-31 12:28:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update plugin docs to git version

2009-08-31 11:32:39 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Fix tests warning from setting a NULL index
	  Setting a null index in the tests was causing warnings by unreffing
	  NULL pointers. This is a bug exposed by a recent change in core, it
	  seems.

2009-08-31 13:02:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: add slope estimation code and debug
	  Add some code to measure the sender speed vs the receiver speed. This can be
	  used to detect bursts.

2009-08-31 12:57:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: reset skew when timestamps change
	  Refactor the jitterbuffer resync code.
	  Reset the skew correction when we detect a big timestamp discont.
	  See #593354

2009-08-31 12:47:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: make sure time never goes invalid
	  Since the skew can be negative, we might end up with invalid timestamps. Check
	  for negative results and clamp to 0.
	  See #593354

2009-08-31 12:16:01 +0200  Jarkko Palviainen <jarkko.palviainen at sesca.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudpnetutils.c:
	  udpsink: Add ttl multicast property
	  Add a new ttl-mc property to control the TTL on multicast addresses.
	  Fixes #588245

2009-08-31 12:13:07 +0200  Jarkko Palviainen <jarkko.palviainen at sesca.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpnetutils.c:
	* gst/udp/gstudpnetutils.h:
	  udp: split out TTL and loop options
	  Split setting the TTL and loop parameters in 2 methods as they are not related.
	  Fix setting the TTL correctly for multicast streams.
	  See #588245

2009-08-27 12:36:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	  rtp: whitespace fixes

2009-08-14 13:45:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins.args:
	  videobox: Correctly add to the docs

2009-08-14 13:40:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/Makefile.am:
	* gst/videobox/gstvideobox.c:
	* gst/videobox/gstvideobox.h:
	  videobox: Split declarations into a header file and add autocrop stuff to the docs

2009-08-14 13:26:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: Reconfigure basetransform if something changes again
	  For this invent a new lock and don't abuse the basetransform lock,
	  otherwise we'll end up in deadlocks.

2009-08-14 13:15:57 +0200  Stephen Jungels <stephen@jungels.net>

	* gst/videobox/gstvideobox.c:
	  videobox: Add support for autocropping according to the caps
	  Fixes bug #582238.

2009-08-30 21:57:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Make sure that used caps are not freed already
	  Fixes bug #593391.

2009-08-26 17:02:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpstats.c:
	  rtp: Use new gst_iterator_new_single() for the internal linked pads iteration

2009-08-19 16:57:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Use iterate internal links instead of deprecated get internal links

2009-08-19 16:48:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: Use iterate internal links instead of deprecated get internal links

2009-08-19 16:37:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Use iterate internal links instead of deprecated get internal links

2009-08-30 23:27:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Update common

2009-08-30 23:26:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  Back to hacking -> 0.10.16.1

=== release 0.10.16 ===

2009-08-29 12:05:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Release 0.10.16

2009-08-26 00:58:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  0.10.15.5 pre-release

2009-08-25 16:53:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: don't use relative seeks
	  Don't use relative seeks, it's too hard to track where we are after a flush
	  etc.
	  fixes #593015

2009-08-24 17:50:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* po/LINGUAS:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  0.10.15.4 pre-release

2009-08-24 16:22:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: don't discard the result of _set_caps()
	  Use the result of gst_pad_set_caps() instead of assuming success.
	  See #590678

2009-08-21 11:44:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: add support for agsm
	  Fixes #592530

2009-08-18 17:16:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix qt style string tag extraction
	  QT style tags are tested on starting with (C) symbol using >>,
	  and (unsigned) int (may) have different >> behaviour.
	  Fixes #592232.

2009-08-17 15:48:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/smokecodec.c:
	  smokeenc: don't crash when compiled against libjpeg7
	  Set parameters so that we don't crash with libjpeg7. Based on
	  Stefan Kost's fix for jpegenc. Fixes #591951.

2009-08-14 20:18:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  0.10.15.3 pre-release

2009-08-14 13:45:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  checks: add test for leak to rtpbin unit test
	  See #591476.

2009-08-11 14:47:12 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Fix reference leak
	  Fixes #591476.

2009-08-14 13:34:53 +0100  Zaheer Merali <zaheerabbas@merali.org>

	* ext/dv/gstdvdec.c:
	  dvdec: set bottom field first on PAL interlaced content, not top field first
	  DV interlaced content is always bottom field first. Fixes #591712.

2009-08-14 12:44:06 +0100  Hans de Goede <jwrdegoede@fedoraproject.org>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: fix 'hang' with some cameras caused by bad timestamping if no framerate is available
	  For cameras/drivers that don't support e.g. VIDIOC_G_PARM we'd end up without
	  a framerate and would try to divide by 0, causing run-time warnings and all
	  frames to be timestamped with 0, which makes sinks that sync against the clock
	  drop them, causing 'hangs' (observed with the pwc driver and a Logitech QuickCam
	  Pro 4000). So if we do not know the framerate, simply don't adjust the
	  timestamps. Fixes #591451.

2009-08-14 10:11:25 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2src: clear format list in READY->NULL
	  Clear format list and probed caps when going to NULL so if a new device
	  is set we'll probe the formats again instead of using previously
	  detected ones. Fixes bug #591747.

2009-08-11 16:42:51 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  dtmfsrc: Empty event queue on finalize

2009-08-11 16:39:42 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmf: Use GSlice for internal event structures

2009-08-11 16:23:20 -0400  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: Cleanup events on finalize
	  Problem found by Laurent Glayal
	  Fixes bug #591440

2009-08-11 16:23:20 -0400  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  rtpdtmfsrc: Cleanup events on finalize
	  Problem found by Laurent Glayal
	  Fixes bug #591440

2009-08-11 17:30:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* po/LINGUAS:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  0.10.15.2 pre-release

2009-08-11 15:25:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* MAINTAINERS:
	  Add myself to MAINTAINERS file and update Wim's e-mail.

2009-08-11 03:08:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/Makefile.am:
	  v4l2: fix make distcheck by disting some more headers

2009-08-11 02:42:16 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  docs: update

2009-08-11 02:31:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* gst-plugins-good.spec.in:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/pipelines/.gitignore:
	  Move rtpmanager from -bad to -good.
	  Hook up build infrastructure (autotools, docs, unit test).

2009-08-06 19:26:21 +0200  ric <csxnju at sogou.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: avoid buffer leak on bad seqnum
	  Fixes #590797

2009-07-28 18:18:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: allow for NULL caps on buffers
	  Add the NULL caps check where it matters and also cover another case of
	  potential NULL caps.
	  Fixes #590030

2009-07-28 11:59:56 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Incoming buffers do not always have caps

2009-07-27 15:46:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: avoid doing lip-sync in BYE
	  When we get a BYE packet, don't do lip-sync with the SR inside because some
	  senders have trouble constructing valid SR packets after BYE.

2009-07-27 13:17:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpbin: don't do lip-sync after a BYE
	  After a BYE packet from a source, stop forwarding the SR packets for lip-sync
	  to rtpbin. Some senders don't update their SR packets correctly after sending a
	  BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with
	  the current lip-sync instead.

2009-07-27 12:43:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpbin: only reconsider once for BYE
	  When iterating the sources of a BYE packet, don't signal a reconsideration for
	  each of them but signal after we handled all sources.

2009-07-21 15:33:41 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Free conflicting addresses on finalize

2009-07-01 12:55:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpbin: use new method for netaddress to string

2009-06-29 18:48:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* tests/check/elements/rtpbin.c:
	  rtpbin: do better cleanup of the src ghostpads
	  Connect to the pad-removed signal of the ptdemux elements so that we remove the
	  ghostpads for them. Fixes cleanup when going to NULL as well as when releasing
	  the sinkpads.
	  Fixes #561752

2009-05-28 19:08:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: add a comment

2009-06-29 16:37:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin: add SDES property
	  Remove all individual SDES properties and use one sdes property that takes a
	  GstStructure instead. This will allow us to add more custom stuff to the SDES
	  messages later.

2009-06-29 16:21:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpbin: add SDES property that takes GstStructure
	  Remove all individual SDES properties and use one sdes property that takes a
	  GstStructure instead. This will allow us to add more custom stuff to the SDES
	  messages later.

2009-06-02 17:46:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpclient.c:
	* gst/rtpmanager/gstrtpclient.h:
	* gst/rtpmanager/gstrtpmanager.c:
	  rtpbin: removed old gstrtpclient

2009-06-19 19:09:19 +0200  Branko Subasic <branko.subasic at axis.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* tests/check/elements/rtpbin_buffer_list.c:
	  rtpbin: add support for buffer-list
	  Add support for sending buffer-lists.
	  Add unit test for testing that the buffer-list passed through rtpbin.
	  fixes #585839

2009-06-19 16:21:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Make build without warnings with debugging disabled

2009-05-28 17:37:44 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Transform the right session sdes message
	  Fixes #584165

2009-05-28 17:33:10 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  Add ssrc to application/x-rtp-source-sdes structure

2009-05-27 11:03:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsouce: the network address is in network order
	  Bring the network address in netowkr byte order to the host order.

2009-05-26 15:40:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: byteswap the port from GstNetAddress
	  Since the port in GstNetAddress is in network order we might need to byteswap it
	  before adding it to the source statistics.

2009-05-25 13:46:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: remove ptdemux ghostpads

2009-05-25 13:33:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  tests: add receive rtpbin unit test

2009-05-22 16:41:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: add to new signal to remove SSRC pads

2009-05-22 16:35:20 +0200  Ali Sabil <ali.sabil at gmail.com>

	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	  ssrcdemux: emit signal when pads are removed
	  Add action signal to clear an SSRC in the ssrc demuxer.
	  Add signal to notify of removed ssrc.
	  See #554839

2009-05-22 15:45:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: use our ghostpads instead of its target
	  Since we keep a reference to our ghostpads, we can use them to track sessions.
	  This avoid us having to mess with the target of the ghostpad.

2009-05-22 15:37:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  tests: more rtpbin checks

2009-05-22 15:36:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: don't warn when getting request pads twice
	  Allow getting the request pads multiple times, just return the previously
	  created pads.

2009-05-22 13:47:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: add RTP and RTCP source address
	  Add the RTP and RTCP sender addresses in the stats structure.

2009-05-22 13:45:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: reuse source code for SDES
	  Reuse the RTPSource object property instead of duplicating code.

2009-05-22 13:44:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  tests: add more rtpbin tests

2009-05-22 12:23:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpbin.c:
	  tests: add rtpbin unit test
	  Add the beginnings of an rtpbin unit test
	  Add some more stuff to .gitignore

2009-05-22 12:20:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: set target state on new elements
	  Set the state on newly added elements to the state of the parent.
	  Add some debug info and do some cleanups

2009-05-22 11:59:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: unref requests pads after releasing

2009-05-22 01:43:50 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Implement releasing the streams
	  See #561752

2009-05-22 01:16:11 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Keep jb signals handler
	  Keep the signal handlers so they can be disconnected at release time
	  See #561752

2009-05-22 01:12:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: use the right lock for the sessions
	  Use the right lock when iterating the sessions.

2009-05-22 01:03:55 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Free session if request pads are released
	  Free the session when all the request pads are released.
	  Don't mess with the session list in free_session as it is called from a foreach
	  on that list.
	  Set the state of the upstream element to NULL first.
	  See #561752

2009-05-22 00:51:53 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Implement relasing of the rtp recv pad

2009-05-22 00:44:51 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Implement releasing of rtp send pads

2009-05-22 00:34:36 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Implement release of the recv rtcp pad
	  See #561752

2009-05-22 00:16:19 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Implement releasing of rtcp src pad
	  See #561752

2009-05-05 16:48:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: drop unexpected RTCP packets
	  We usually only get SR packets in our chain function but if an invalid packet
	  contains the SR packet after the RR packet, we must not fail but simply ignore
	  the malformed packet.
	  Fixes #581375

2009-04-27 11:09:08 +0200  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  rtpsouce: make WARNING into LOG
	  Since neither rtpmanager nor any of the payloaders properly implement
	  pad allocation, there is no way for the rtpmanager to inform downstream elements
	  of the new SSRC if there is an SSRC collision. So the warning is emitted all the
	  time and it is confusing.
	  Fixes #580144

2009-04-27 11:06:01 +0200  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: notify when SSRC changes
	  Emit a g_object_notify when the SSRc changes because of a collision.
	  Fixes #580144

2009-04-17 16:16:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: join the RTCP thread
	  Avoid a case where a joinable thread would be left unjoined, which leaked the
	  thread structure.
	  Fixes #577318.

2009-04-15 18:14:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: prevent overflow in EOS estimation
	  Use a guint64 instead of a guint to hold a 64bit value to prevent completely
	  bogues EOS estimation values due to overflows.

2009-04-15 17:44:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: we should not provide a clock
	  There is no need to provide a clock.

2009-04-15 17:28:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: more estimated EOS fixes
	  Do more accurate EOS estimate and guard against backward timestamps.

2009-04-15 17:25:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: release lock before pushing EOS
	  Make sure we release the jitterbuffer lock before we start pushing out data
	  because else we might deadlock.

2009-03-27 17:44:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpjitterbuffer.h:
	  rtpbin: add on_npt_stop signal
	  Add the on_npt_stop signal to rtpbin and rtpjitterbuffer to notify the
	  application that the NPT stop position has been reached.

2009-03-13 15:59:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin: don't return FALSE on seek events
	  Silently ignore the seek event instead of returning FALSE.

2009-02-26 13:10:29 +0100  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpbin: Don't forward revc events to sender
	  Don't send events from the receiver to the sender side.
	  Fixes #572900.

2009-02-25 11:45:05 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  docs: various doc fixes
	  No short-desc as we have them in the element details.
	  Also keep things (Makefile.am and sections.txt) sorted.
	  Reword ambigous returns. No text after since please.

2009-01-23 12:13:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpstats.c:
	  Send BYE packets immediatly for small sessions
	  When the number of participants is less than 50, the RFC allows for sending the
	  BYE packet immediatly instead of using the regular BYE timeout.
	  Fixes #567828.

2009-01-22 13:33:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Unlock the jitterbuffer before pushing out the packet-lost events. Move some code before we do the unlock to make the jitterbuffer state consistent while we are unlocked.

2009-01-02 17:40:06 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/: When an SSRC is found on the caps of the sender RTP, use this as the internal SSRC. Fixes #565910.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_setcaps_send_rtp), (create_send_rtp_sink):
	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
	  When an SSRC is found on the caps of the sender RTP, use this as the
	  internal SSRC. Fixes #565910.

2009-01-02 16:50:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Rename a method to better reflect what it really does.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_event_send_rtp_sink),
	  (gst_rtp_session_getcaps_send_rtp):
	  * gst/rtpmanager/rtpsession.c: (check_collision),
	  (rtp_session_schedule_bye_locked), (rtp_session_schedule_bye):
	  * gst/rtpmanager/rtpsession.h:
	  Rename a method to better reflect what it really does.

2008-12-29 15:49:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Use method to get the internal SSRC.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_getcaps_send_rtp):
	  Use method to get the internal SSRC.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (rtp_session_set_property), (rtp_session_get_property):
	  Add property to congiure the internal SSRC of the session.
	  Fixes #565910.

2008-12-29 15:21:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/rtpsession.c: Only change the SSRC of the session and reset the internal source when the SSRC actually...
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
	  Only change the SSRC of the session and reset the internal source when
	  the SSRC actually changed. See #565910.

2008-12-29 14:21:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/rtpsource.*: When no payload was specified on the caps but there was a clock-rate, assume the clock-ra...
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
	  (rtp_source_update_caps), (get_clock_rate):
	  * gst/rtpmanager/rtpsource.h:
	  When no payload was specified on the caps but there was a clock-rate,
	  assume the clock-rate corresponds to the first payload type found in the
	  RTP packets. Fixes #565509.

2008-12-23 11:39:59 +0000  Arnout Vandecappelle <arnout@mind.be>

	  gst/rtpmanager/rtpjitterbuffer.*: Keep track of the last outgoing timestamp and of the last sender-side time.  Timest...
	  Original commit message from CVS:
	  Patch by: Arnout Vandecappelle <arnout at mind dot be>
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
	  (calculate_skew):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Keep track of the last outgoing timestamp and of the last sender-side
	  time.  Timestamps can only go forward if they do at the sender
	  side, can only go back if they do at the sender side, and remain the
	  same if they remain the same at the sender side. Fixes #565319.

2008-11-26 12:40:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/rtpsession.c: Make obtain_source return an aditional ref so that we don't lose our ref to it when a se...
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsession.c: (obtain_source),
	  (rtp_session_create_source), (rtp_session_process_rtp),
	  (rtp_session_process_sr), (rtp_session_process_rr),
	  (rtp_session_process_sdes), (rtp_session_process_bye):
	  Make obtain_source return an aditional ref so that we don't lose our ref
	  to it when a session cleanup occurs when we are emiting a signal.
	  Emit the on_new_ssrc signal for the CSRC, not the SSRC.
	  Fixes #562319.

2008-11-26 12:02:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Reset the sync parameters when clearing the payload type map too.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_reset_sync),
	  (gst_rtp_bin_clear_pt_map):
	  Reset the sync parameters when clearing the payload type map too.
	  Fixes #562312.

2008-11-26 11:44:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.*: Remove a lot of per stream state that is not needed and pass new info in the method call.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (get_client),
	  (gst_rtp_bin_reset_sync), (gst_rtp_bin_associate),
	  (gst_rtp_bin_handle_sync), (create_stream),
	  (gst_rtp_bin_class_init), (new_ssrc_pad_found):
	  * gst/rtpmanager/gstrtpbin.h:
	  Remove a lot of per stream state that is not needed and pass new info in
	  the method call.
	  Add signal to reset sync parameters.
	  Avoid parsing the caps to get a clock_base, we get this from the sync
	  signal now.

2008-11-25 15:12:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Fix event leak.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_event_send_rtcp_src):
	  Fix event leak.

2008-11-22 15:31:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/rtpsession.c: Add property to configure the RTCP MTU.
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (rtp_session_init), (rtp_session_set_property),
	  (rtp_session_get_property):
	  Add property to configure the RTCP MTU.

2008-11-22 15:24:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/rtpsession.c: Add G_PARAM_STATIC_STRINGS.
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (copy_source), (rtp_session_create_sources),
	  (rtp_session_get_property):
	  Add G_PARAM_STATIC_STRINGS.
	  Add property to return a GValueArray of all known RTPSources in the
	  session.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
	  (rtp_source_create_sdes), (rtp_source_set_property),
	  (rtp_source_get_property):
	  Remove properties to set the various SDES items, an application is never
	  supposed to change the RTPSource data.
	  Change the SDES getter properties to one SDES property that returns all
	  SDES items in a GstStructure.

2008-11-22 13:17:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Also unref the target pad for unknown pads.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
	  Also unref the target pad for unknown pads.

2008-11-21 16:17:22 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/gstrtpbin.c: Release the right pads on rtpbin. Fixes #561752.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
	  Release the right pads on rtpbin. Fixes #561752.

2008-11-20 18:41:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when processing RTP packets.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (get_current_times),
	  (rtcp_thread), (gst_rtp_session_chain_recv_rtp):
	  Pass the running time to the session when processing RTP packets.
	  Improve the time function to provide more info.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (rtp_session_init), (update_arrival_stats),
	  (rtp_session_process_rtp), (rtp_session_process_sdes),
	  (rtp_session_process_rtcp), (session_start_rtcp),
	  (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsession.h:
	  Mark the internal source with a flag.
	  Use running_time instead of the more useless timestamp.
	  Validate a source when a valid SDES has been received.
	  Pass the current system time when processing SR packets.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
	  (rtp_source_init), (rtp_source_create_stats),
	  (rtp_source_get_property), (rtp_source_send_rtp),
	  (rtp_source_process_rb), (rtp_source_get_new_rb),
	  (rtp_source_get_last_rb):
	  * gst/rtpmanager/rtpsource.h:
	  Add property to get source stats.
	  Mark params as STATIC_STRINGS.
	  Calculate the bitrate at the sender SSRC.
	  Avoid negative values in the round trip time calculations.
	  * gst/rtpmanager/rtpstats.h:
	  Update some docs and change some variable name to more closely reflect
	  what it contains.

2008-11-20 08:19:15 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Initialize return value to fix compiler warning about uninitialized variable.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain_rtcp):
	  Initialize return value to fix compiler warning about uninitialized
	  variable.

2008-11-19 16:48:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Mark signal arg as static scope.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init):
	  Mark signal arg as static scope.

2008-11-19 09:06:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Remove internal sync pad, use signals instead to get lip-sync notifications.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
	  (gst_rtp_bin_handle_sync), (create_stream), (free_stream),
	  (new_ssrc_pad_found):
	  Remove internal sync pad, use signals instead to get lip-sync
	  notifications.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_base_init),
	  (gst_rtp_jitter_buffer_class_init),
	  (gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink),
	  (remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad),
	  (gst_rtp_jitter_buffer_release_pad),
	  (gst_rtp_jitter_buffer_sink_rtcp_event),
	  (gst_rtp_jitter_buffer_chain_rtcp),
	  (gst_rtp_jitter_buffer_get_property):
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  Make it possible to send SR packets to the jitterbuffer.
	  Check if the SR timestamps are valid by comparing them to the RTP
	  timestamps.
	  Signal the SR packet and the timing information to listeners.
	  * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
	  (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query):
	  Remove some unused code.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
	  (calculate_skew), (rtp_jitter_buffer_get_sync):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Keep track of the last seen RTP timestamp so that we can filter out
	  invalid SR packets.

2008-11-17 19:47:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/rtpmanager/rtpsource.c: Fix GST_DEBUG call to only have as many arguments as required by the format string. Fixes...
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsource.c: (get_clock_rate):
	  Fix GST_DEBUG call to only have as many arguments as required
	  by the format string. Fixes a compiler warning.

2008-11-17 15:17:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ourselves but simply get the value from the ji...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
	  (gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found):
	  Do not try to keep track of the clock-rate ourselves but simply get the
	  value from the jitterbuffer.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
	  (gst_rtp_jitter_buffer_get_sync):
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  Add some debug info.
	  Pass the clock-rate to the jitterbuffer.
	  Also pass the clock-rate along with the rtp timestamp when getting the
	  sync parameters.
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
	  Fix some debug.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
	  (calculate_skew), (rtp_jitter_buffer_get_sync):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Keep track of clock-rate changes and return the clock-rate together with
	  the rtp timestamps used for sync.
	  Don't try to construct timestamps when we have no base_time.
	  * gst/rtpmanager/rtpsource.c: (get_clock_rate):
	  Request a new clock-rate when the payload type changes.
	  Reset the jitter calculation when the clock-rate changes.

2008-11-13 15:48:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Small cleanups and some more debug info.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_jitter_buffer_sink_parse_caps),
	  (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain):
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
	  (calculate_skew):
	  Small cleanups and some more debug info.

2008-11-10 15:26:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Also configure the next expected output seqnum when we get a seqnum-base on the ...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
	  Also configure the next expected output seqnum when we get a seqnum-base
	  on the caps.

2008-11-04 12:42:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Don't install static libs for plugins. Fixes #550851 for -bad.
	  Original commit message from CVS:
	  * ext/alsaspdif/Makefile.am:
	  * ext/amrwb/Makefile.am:
	  * ext/apexsink/Makefile.am:
	  * ext/arts/Makefile.am:
	  * ext/artsd/Makefile.am:
	  * ext/audiofile/Makefile.am:
	  * ext/audioresample/Makefile.am:
	  * ext/bz2/Makefile.am:
	  * ext/cdaudio/Makefile.am:
	  * ext/celt/Makefile.am:
	  * ext/dc1394/Makefile.am:
	  * ext/dirac/Makefile.am:
	  * ext/directfb/Makefile.am:
	  * ext/divx/Makefile.am:
	  * ext/dts/Makefile.am:
	  * ext/faac/Makefile.am:
	  * ext/faad/Makefile.am:
	  * ext/gsm/Makefile.am:
	  * ext/hermes/Makefile.am:
	  * ext/ivorbis/Makefile.am:
	  * ext/jack/Makefile.am:
	  * ext/jp2k/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/lcs/Makefile.am:
	  * ext/libfame/Makefile.am:
	  * ext/libmms/Makefile.am:
	  * ext/metadata/Makefile.am:
	  * ext/mpeg2enc/Makefile.am:
	  * ext/mplex/Makefile.am:
	  * ext/musepack/Makefile.am:
	  * ext/musicbrainz/Makefile.am:
	  * ext/mythtv/Makefile.am:
	  * ext/nas/Makefile.am:
	  * ext/neon/Makefile.am:
	  * ext/ofa/Makefile.am:
	  * ext/polyp/Makefile.am:
	  * ext/resindvd/Makefile.am:
	  * ext/sdl/Makefile.am:
	  * ext/shout/Makefile.am:
	  * ext/snapshot/Makefile.am:
	  * ext/sndfile/Makefile.am:
	  * ext/soundtouch/Makefile.am:
	  * ext/spc/Makefile.am:
	  * ext/swfdec/Makefile.am:
	  * ext/tarkin/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/timidity/Makefile.am:
	  * ext/twolame/Makefile.am:
	  * ext/x264/Makefile.am:
	  * ext/xine/Makefile.am:
	  * ext/xvid/Makefile.am:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/dshow/Makefile.am:
	  * gst/aiffparse/Makefile.am:
	  * gst/app/Makefile.am:
	  * gst/audiobuffer/Makefile.am:
	  * gst/bayer/Makefile.am:
	  * gst/cdxaparse/Makefile.am:
	  * gst/chart/Makefile.am:
	  * gst/colorspace/Makefile.am:
	  * gst/dccp/Makefile.am:
	  * gst/deinterlace/Makefile.am:
	  * gst/deinterlace2/Makefile.am:
	  * gst/dvdspu/Makefile.am:
	  * gst/festival/Makefile.am:
	  * gst/filter/Makefile.am:
	  * gst/flacparse/Makefile.am:
	  * gst/flv/Makefile.am:
	  * gst/games/Makefile.am:
	  * gst/h264parse/Makefile.am:
	  * gst/librfb/Makefile.am:
	  * gst/mixmatrix/Makefile.am:
	  * gst/modplug/Makefile.am:
	  * gst/mpeg1sys/Makefile.am:
	  * gst/mpeg4videoparse/Makefile.am:
	  * gst/mpegdemux/Makefile.am:
	  * gst/mpegtsmux/Makefile.am:
	  * gst/mpegvideoparse/Makefile.am:
	  * gst/mve/Makefile.am:
	  * gst/nsf/Makefile.am:
	  * gst/nuvdemux/Makefile.am:
	  * gst/overlay/Makefile.am:
	  * gst/passthrough/Makefile.am:
	  * gst/pcapparse/Makefile.am:
	  * gst/playondemand/Makefile.am:
	  * gst/rawparse/Makefile.am:
	  * gst/real/Makefile.am:
	  * gst/rtjpeg/Makefile.am:
	  * gst/rtpmanager/Makefile.am:
	  * gst/scaletempo/Makefile.am:
	  * gst/sdp/Makefile.am:
	  * gst/selector/Makefile.am:
	  * gst/smooth/Makefile.am:
	  * gst/smoothwave/Makefile.am:
	  * gst/speed/Makefile.am:
	  * gst/speexresample/Makefile.am:
	  * gst/stereo/Makefile.am:
	  * gst/subenc/Makefile.am:
	  * gst/tta/Makefile.am:
	  * gst/vbidec/Makefile.am:
	  * gst/videodrop/Makefile.am:
	  * gst/videosignal/Makefile.am:
	  * gst/virtualdub/Makefile.am:
	  * gst/vmnc/Makefile.am:
	  * gst/y4m/Makefile.am:
	  * sys/acmenc/Makefile.am:
	  * sys/cdrom/Makefile.am:
	  * sys/dshowdecwrapper/Makefile.am:
	  * sys/dshowsrcwrapper/Makefile.am:
	  * sys/dvb/Makefile.am:
	  * sys/dxr3/Makefile.am:
	  * sys/fbdev/Makefile.am:
	  * sys/oss4/Makefile.am:
	  * sys/qcam/Makefile.am:
	  * sys/qtwrapper/Makefile.am:
	  * sys/vcd/Makefile.am:
	  * sys/wininet/Makefile.am:
	  * win32/common/config.h:
	  Don't install static libs for plugins. Fixes #550851 for -bad.

2008-10-16 13:05:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Fix problem with using the output seqnum counter to check for input seqnum disco...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_jitter_buffer_sink_parse_caps),
	  (gst_rtp_jitter_buffer_flush_start),
	  (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain),
	  (gst_rtp_jitter_buffer_loop):
	  Fix problem with using the output seqnum counter to check for input
	  seqnum discontinuities.
	  Improve gap detection and recovery, reset and flush the jitterbuffer on
	  seqnum restart. Fixes #556520.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert):
	  Fix wrong G_LIKELY.

2008-10-16 09:51:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, make LATENCY event return
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src):
	  Install event handler on the rtcp_src pad, make LATENCY event return
	  TRUE.

2008-10-07 18:54:41 +0000  Håvard Graff <havard.graff@tandberg.com>

	  gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.
	  Original commit message from CVS:
	  Patch by: Håvard Graff <havard dot graff at tandberg dot com>
	  * gst/rtpmanager/gstrtpbin-marshal.list:
	  Add marshaller for new action signal.
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session),
	  (gst_rtp_bin_class_init):
	  * gst/rtpmanager/gstrtpbin.h:
	  Add action signal to retrieve the internal RTPSession object.
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (gst_rtp_session_get_property), (gst_rtp_session_release_pad):
	  Add property to access the internal RTPSession object.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (check_collision):
	  * gst/rtpmanager/rtpsession.h:
	  Add action signal to retrieve an RTPSource object by SSRC.
	  See #555396.

2008-10-07 11:33:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Release pads of the session manager.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (find_session_by_pad),
	  (free_session), (gst_rtp_bin_dispose), (remove_recv_rtp),
	  (remove_recv_rtcp), (remove_send_rtp), (remove_rtcp),
	  (gst_rtp_bin_release_pad):
	  Release pads of the session manager.
	  Start implementing releasing pads of gstrtpbin.
	  * gst/rtpmanager/gstrtpsession.c: (remove_recv_rtp_sink),
	  (remove_recv_rtcp_sink), (remove_send_rtp_sink),
	  (remove_send_rtcp_src), (gst_rtp_session_release_pad):
	  Implement releasing pads in gstrtpsession.

2008-10-07 10:02:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Only update the seqnum-base when it was not already configured for the streams.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_jitter_buffer_sink_parse_caps):
	  Only update the seqnum-base when it was not already configured for the
	  streams.

2008-09-30 15:08:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/rtpsession.c: Ref the rtpsource object before we release the session lock when we emit the signals.
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
	  (on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes),
	  (on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout):
	  Ref the rtpsource object before we release the session lock when we emit
	  the signals.

2008-09-23 18:13:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Fix some docs.
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert),
	  (rtp_jitter_buffer_get_sync):
	  * gst/rtpmanager/rtpsession.c: (on_sender_timeout),
	  (session_cleanup):
	  * gst/rtpmanager/rtpsource.c:
	  Fix some docs.

2008-09-17 13:59:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Fix compiler warnings on OS/X
	  Original commit message from CVS:
	  * ext/jack/gstjackaudiosink.c: (jack_process_cb):
	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
	  Fix compiler warnings on OS/X

2008-09-13 01:37:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_session),
	  (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
	  Do not try to adjust the offset of streams for which we have not yet
	  seen an SR packet. Avoids large ts-offsets in some cases.

2008-09-05 13:52:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
	  (create_session), (gst_rtp_bin_associate),
	  (gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
	  (gst_rtp_bin_request_new_pad):
	  * gst/rtpmanager/gstrtpbin.h:
	  Add signal to notify listeners when a sender becomes a receiver.
	  Tweak lip-sync code, don't store our own copy of the ts-offset of the
	  jitterbuffer, don't adjust sync if the change is less than 4msec.
	  Get the RTP timestamp <-> GStreamer timestamp relation directly from
	  the jitterbuffer instead of our inaccurate version from the source.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
	  (gst_rtp_jitter_buffer_get_sync):
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  Add G_LIKELY macros, use global defines for max packet reorder and
	  dropouts.
	  Reset the jitterbuffer clock skew detection when packets seqnums are
	  changed unexpectedly.
	  * gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
	  (gst_rtp_session_class_init), (gst_rtp_session_init):
	  * gst/rtpmanager/gstrtpsession.h:
	  Add sender timeout signal.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
	  (calculate_skew), (rtp_jitter_buffer_insert),
	  (rtp_jitter_buffer_get_sync):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Add some G_LIKELY macros.
	  Keep track of the extended RTP timestamp so that we can report the RTP
	  timestamp <-> GStreamer timestamp relation for lip-sync.
	  Remove server timestamp gap detection code, the server can sometimes
	  make a huge gap in timestamps (talk spurts,...) see #549774.
	  Detect timetamp weirdness instead by observing the sender/receiver
	  timestamp relation and resync if it changes more than 1 second.
	  Add method to report about the current rtp <-> gst timestamp relation
	  which is needed for lip-sync.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (on_sender_timeout), (check_collision), (rtp_session_process_sr),
	  (session_cleanup):
	  * gst/rtpmanager/rtpsession.h:
	  Add sender timeout signal.
	  Remove inaccurate rtp <-> gst timestamp relation code, the
	  jitterbuffer can now do an accurate reporting about this.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
	  (rtp_source_update_caps), (calculate_jitter),
	  (rtp_source_process_rtp):
	  * gst/rtpmanager/rtpsource.h:
	  Remove inaccurate rtp <-> gst timestamp relation code.
	  * gst/rtpmanager/rtpstats.h:
	  Define global max-reorder and max-dropout constants for use in various
	  subsystems.

2008-08-28 15:21:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs us to.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
	  (gst_rtp_session_event_send_rtp_sink):
	  Send EOS when the session object instructs us to.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsession.h:
	  Make it possible for the session manager to instruct us to send EOS. We
	  currently will EOS when the session is a sender and when the sender part
	  goes EOS. This is not entirely correct behaviour because the session
	  could still participate as a receiver.
	  Fixes #549409.

2008-08-13 14:31:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we detect a gap when the clock_base changed.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
	  (gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
	  Reset rtp timestamp interpollation when we detect a gap when the
	  clock_base changed.
	  Don't try to adjust the ts-offset when it's too big (> 3seconds)
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
	  * gst/rtpmanager/gstrtpsession.h:
	  Add method to set session SSRC.
	  * gst/rtpmanager/rtpsession.c: (check_collision),
	  (rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
	  (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsession.h:
	  Added debugging for the collision checks.
	  Add method to change the internal SSRC of the session.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
	  Reset the clock base when we detect large jumps in the seqnums.

2008-08-11 07:20:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/rtpmanager/gstrtpbin.c: Print the pad-name in debug log.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c:
	  Print the pad-name in debug log.
	  * sys/dshowsrcwrapper/gstdshowaudiosrc.c:
	  * sys/dshowsrcwrapper/gstdshowvideosrc.c:
	  Use "-" instead of "_" in property names. Can we call them just
	  "device" like everywhere else?

2008-08-05 09:42:53 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Make the buffer metadata writable before inserting it in the jitterbuffer becaus...
	  Original commit message from CVS:
	  Based on patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
	  Make the buffer metadata writable before inserting it in the
	  jitterbuffer because the jitterbuffer will modify the timestamps.
	  * gst/rtpmanager/rtpjitterbuffer.c:
	  Update method comment about requiring writable metadata on buffers.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
	  (rtp_session_process_rtcp):
	  Make the RTCP buffer metadata writable because we want to modify the
	  metadata.
	  Fixes #546312.

2008-08-05 09:00:50 +0000  Håvard Graff <havard.graff@tandberg.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Fix debug by logging the right seqnum.
	  Original commit message from CVS:
	  Patch by: Håvard Graff <havard dot graff at tandberg dot com>
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain):
	  Fix debug by logging the right seqnum.

2008-08-05 08:58:27 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/gstrtpbin.c: Release lock before emitting the request-pt-map signal.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtpmanager/gstrtpbin.c: (get_pt_map):
	  Release lock before emitting the request-pt-map signal.
	  Fixes #543480.

2008-07-03 14:44:51 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtpmanager/: Corrected a typo (interpollate -> interpolate).
	  Original commit message from CVS:
	  * ChangeLog:
	  * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop):
	  * gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr):
	  Corrected a typo (interpollate -> interpolate).

2008-07-03 14:31:10 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
	  (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
	  (gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp),
	  (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp):
	  * gst/rtpmanager/rtpsession.c: (source_push_rtp),
	  (rtp_session_send_rtp):
	  * gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter),
	  (rtp_source_process_rtp), (rtp_source_send_rtp):
	  Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a
	  pipeline is running normally.

2008-07-03 13:47:19 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtpmanager/: Do not mix the use of g_get_current_time() with gst_clock_get_time().
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
	  (gst_rtp_session_finalize), (rtcp_thread),
	  (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp),
	  (gst_rtp_session_event_send_rtp_sink),
	  (gst_rtp_session_chain_send_rtp):
	  * gst/rtpmanager/rtpsession.c: (check_collision),
	  (update_arrival_stats), (rtp_session_process_rtp),
	  (rtp_session_process_rtcp), (rtp_session_send_rtp),
	  (rtp_session_send_bye_locked), (rtp_session_send_bye),
	  (rtp_session_next_timeout), (session_report_blocks), (session_cleanup),
	  (is_rtcp_time), (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsession.h:
	  Do not mix the use of g_get_current_time() with gst_clock_get_time().

2008-06-16 07:30:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Final round of doc updates.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/speed/gstspeed.c:
	  * gst/speexresample/gstspeexresample.c:
	  * gst/videosignal/gstvideoanalyse.c:
	  * gst/videosignal/gstvideodetect.c:
	  * gst/videosignal/gstvideomark.c:
	  * sys/dvb/gstdvbsrc.c:
	  * sys/oss4/oss4-mixer.c:
	  * sys/oss4/oss4-sink.c:
	  * sys/oss4/oss4-source.c:
	  * sys/wininet/gstwininetsrc.c:
	  Final round of doc updates.

2008-06-16 07:03:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: More doc updates. More xrefs.
	  Original commit message from CVS:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/rtpmanager/gstrtpbin.c:
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/rtpmanager/gstrtpptdemux.c:
	  * gst/rtpmanager/gstrtpsession.c:
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  * gst/sdp/gstsdpdemux.c:
	  More doc updates. More xrefs.

2008-06-12 14:49:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Do not use short_description in section docs for elements. We extract them from element details and there will be war...
	  Original commit message from CVS:
	  * ext/dc1394/gstdc1394.c:
	  * ext/ivorbis/vorbisdec.c:
	  * ext/jack/gstjackaudiosink.c:
	  * ext/metadata/gstmetadatademux.c:
	  * ext/mythtv/gstmythtvsrc.c:
	  * ext/theora/theoradec.c:
	  * gst-libs/gst/app/gstappsink.c:
	  * gst/bayer/gstbayer2rgb.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/rawparse/gstaudioparse.c:
	  * gst/rawparse/gstvideoparse.c:
	  * gst/rtpmanager/gstrtpbin.c:
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/rtpmanager/gstrtpptdemux.c:
	  * gst/rtpmanager/gstrtpsession.c:
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  * gst/selector/gstinputselector.c:
	  * gst/selector/gstoutputselector.c:
	  * gst/videosignal/gstvideoanalyse.c:
	  * gst/videosignal/gstvideodetect.c:
	  * gst/videosignal/gstvideomark.c:
	  * sys/oss4/oss4-mixer.c:
	  * sys/oss4/oss4-sink.c:
	  * sys/oss4/oss4-source.c:
	  Do not use short_description in section docs for elements. We extract
	  them from element details and there will be warnings if they differ.
	  Also fixing up the ChangeLog order.

2008-06-06 13:01:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Fix deadlock when shutting down, use a new lock instead to properly shutdown.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_init),
	  (gst_rtp_bin_finalize), (gst_rtp_bin_change_state):
	  Fix deadlock when shutting down, use a new lock instead to properly
	  shutdown.

2008-05-27 16:48:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Break out of callbacks when we are shutting down.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c:
	  (gst_rtp_bin_propagate_property_to_jitterbuffer),
	  (gst_rtp_bin_change_state), (new_payload_found),
	  (new_ssrc_pad_found):
	  Break out of callbacks when we are shutting down.
	  Make sure no state changes can happen when we reconfigure.

2008-05-26 10:09:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the jitterbuffer if the gap is too big, we need ...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
	  When checking the seqnum, reset the jitterbuffer if the gap is too big,
	  we need to do this so that we can better handle a restarted source.
	  Fix some comments.
	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
	  (rtp_jitter_buffer_insert):
	  Tweak the skew resync diff.
	  Use our working seqnum compare function in -base.
	  Rework the jitterbuffer insert code to make it clearer and more
	  performant by only retrieving the seqnum of the input buffer once and by
	  adding some G_LIKELY compiler hints.
	  Improve debugging for duplicate packets.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
	  Fix a comment, we don't do skew correction here..

2008-05-26 10:00:24 +0000  Håvard Graff <havard.graff@tandberg.com>

	  gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to the jitterbuffers when they are changed o...
	  Original commit message from CVS:
	  Patch by: Håvard Graff <havard dot graff at tandberg dot com>
	  * gst/rtpmanager/gstrtpbin.c:
	  (gst_rtp_bin_propagate_property_to_jitterbuffer),
	  (gst_rtp_bin_set_property):
	  Propagate the do-lost and latency properties to the jitterbuffers when
	  they are changed on rtpbin.

2008-05-26 09:57:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Don't use _gst_pad().
	  Original commit message from CVS:
	  * examples/switch/switcher.c: (switch_timer):
	  * gst/replaygain/gstrgvolume.c: (gst_rg_volume_init):
	  * gst/rtpmanager/gstrtpclient.c: (create_stream):
	  * gst/sdp/gstsdpdemux.c: (gst_sdp_demux_stream_configure_udp),
	  (gst_sdp_demux_stream_configure_udp_sink):
	  * tests/check/elements/deinterleave.c: (GST_START_TEST),
	  (pad_added_setup_data_check_float32_8ch_cb):
	  * tests/check/elements/rganalysis.c: (send_eos_event),
	  (send_tag_event):
	  Don't use _gst_pad().

2008-05-16 19:56:30 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  docs/Makefile.am: Don't attempt to build plugin docs when they're disabled.
	  Original commit message from CVS:
	  * docs/Makefile.am:
	  Don't attempt to build plugin docs when they're disabled.
	  * gst/bayer/Makefile.am:
	  Add libgstvideo to the link.
	  * gst/rtpmanager/Makefile.am:
	  Fix link order, and move LIBS things to _LIBS

2008-05-14 21:02:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a warning instead of just posting an error and ...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain):
	  Simply drop bad RTP packets with a warning instead of just posting an
	  error and stopping. This is a perfectly recoverable event and we don't
	  force people to use an rtpbin to filter out bad packets first.

2008-05-13 09:06:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
	  Actually add the do-lost property to the object.

2008-05-12 18:43:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) duration when the last packet has a lower ti...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_loop):
	  Avoid waiting for a negative (huge) duration when the last packet has a
	  lower timestamp than the current packet.

2008-05-12 14:28:09 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memor...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src):
	  Make sure to unref the rtpsession returned by gst_pad_get_parent() to
	  prevent a memory leak.

2008-05-12 14:12:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_loop):
	  Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.

2008-05-09 07:41:58 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to prevent a memory leak.
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
	  Make sure to unref the caps used by RTPSource to prevent a memory leak.

2008-05-08 09:43:33 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our callbacks.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtpmanager/rtpsession.c: (source_clock_rate),
	  (rtp_session_process_bye), (rtp_session_send_bye_locked):
	  Unlock the session lock when calling one of our callbacks.
	  Fixes #532011.

2008-05-08 06:23:39 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_event_send_rtp_sink):
	  Send RTP BYE command on EOS. Fixes bug #531955.

2008-04-25 11:32:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.*: Expose new jitterbuffer property in rtpbin too.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init),
	  (gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
	  * gst/rtpmanager/gstrtpbin.h:
	  Expose new jitterbuffer property in rtpbin too.

2008-04-25 11:22:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Disable sending out rtp packet lost events by default and make a property to ena...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
	  (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property),
	  (gst_rtp_jitter_buffer_get_property):
	  Disable sending out rtp packet lost events by default and make a
	  property to enabe it. We will likely enable it by default when the base
	  depayloaders have a default handler for them so that we don't send these
	  events all through the pipeline for now.

2008-04-25 09:35:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Remove private version of a function that is in -base now.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop),
	  (gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain),
	  (gst_rtp_jitter_buffer_loop):
	  Remove private version of a function that is in -base now.
	  Add src event handler.
	  Rework the jitterbuffer pushing loop so that it can quickly react to
	  lost packets and instruct the depayloader of them. This can then be used
	  to implement error concealment data.

2008-04-25 08:21:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the RTCP and sync pads because the defaults ...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink),
	  (create_send_rtcp_src):
	  Set up some internal links functions for the RTCP and sync pads because
	  the defaults are really not correct.
	  Implement a query handler for the RTCP src pad, mostly to correctly
	  report about the latency.

2008-04-25 08:15:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Also keep track of the first buffer timestamp together with the first
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
	  (gst_rtp_bin_sync_chain):
	  * gst/rtpmanager/rtpsession.c: (update_arrival_stats),
	  (rtp_session_process_sr), (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
	  (calculate_jitter):
	  * gst/rtpmanager/rtpsource.h:
	  * gst/rtpmanager/rtpstats.h:
	  Also keep track of the first buffer timestamp together with the first
	  RTP timestamp as they both are needed to construct the timing of
	  outgoing packets in the jitterbuffer and are therefore also needed to
	  manage lip-sync. This fixes lip-sync if the first RTP packets arrive
	  with a wildly different gap.

2008-04-21 08:26:37 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
	  (new_ssrc_pad_found):
	  Ref caps when inserting into the cache.
	  Don't leak pads.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_get_clock_rate),
	  (gst_rtp_jitter_buffer_query):
	  Avoid a caps leak.
	  Don't leak refcount in query.
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
	  (gst_rtp_pt_demux_chain):
	  Avoid caps leaks.
	  * gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
	  (gst_rtp_session_init), (return_true),
	  (gst_rtp_session_clear_pt_map), (gst_rtp_session_cache_caps),
	  (gst_rtp_session_clock_rate):
	  Ref caps when inserting into the cache.
	  Fix some more caps leaks. Fixes #528245.

2008-04-17 07:31:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a refcount leak.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (get_pt_map), (free_client),
	  (gst_rtp_bin_associate), (gst_rtp_bin_get_free_pad_name):
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_get_clock_rate):
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps):
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
	  Unset GValues after g_signal_emitv so that we avoid a refcount leak.
	  Don't leak a padname.
	  Don't leak client streams list.
	  Lock rtpbin when associating streams. Fixes #528245.

2008-04-09 22:27:50 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtpmanager/: Avoid leaking pads in the RTP manager.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (free_session):
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize):
	  Avoid leaking pads in the RTP manager.

2008-03-11 12:40:58 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/rtpsession.*: Implement collision and loop detection in rtpmanager.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtpmanager/rtpsession.c: (find_add_conflicting_addresses),
	  (check_collision), (obtain_source), (rtp_session_create_new_ssrc),
	  (rtp_session_create_source), (rtp_session_process_rtp),
	  (rtp_session_process_sr), (rtp_session_process_rr),
	  (rtp_session_process_sdes), (rtp_session_process_bye),
	  (rtp_session_send_bye_locked), (rtp_session_send_bye),
	  (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsession.h:
	  Implement collision and loop detection in rtpmanager.
	  Fixes #520626.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_reset),
	  (rtp_source_init):
	  * gst/rtpmanager/rtpsource.h:
	  Add method to reset stats.

2008-03-11 11:36:03 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked d...
	  Original commit message from CVS:
	  Based on patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
	  (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread),
	  (join_rtcp_thread), (gst_rtp_session_change_state):
	  Avoid a deadlock when joining the RTCP thread in PAUSED because it might
	  be blocked downstream. Also avoid spawning multiple rtcp threads.
	  Fixes #520894.

2008-03-11 10:43:32 +0000  Stefan Kost <ensonic@users.sf.net>

	  gst/rtpmanager/rtpjitterbuffer.c: Don't try to reset the clock skew when we have no timestamps.
	  Original commit message from CVS:
	  Patch by: Stefan Kost <ensonic@users.sf.net>
	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
	  Don't try to reset the clock skew when we have no timestamps.
	  Fixes #519005.

2008-02-20 09:33:25 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/gstrtpbin.c: Fix small memory leak, leaking caps. Fixes #bug 517571.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
	  Fix small memory leak, leaking caps. Fixes #bug 517571.

2008-02-14 16:25:51 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/gstrtpbin.c: Ignore streams that did not receive an SR packet when doing synchronisation. Fixes #516160.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester@tester.ca>
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate):
	  Ignore streams that did not receive an SR packet when doing
	  synchronisation. Fixes #516160.

2008-01-29 18:57:27 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Try to get the new clock-rate from the buffer caps when we receive a new payload...
	  Original commit message from CVS:
	  Patch by: Thijs Vermeir  <thijsvermeir at gmail dot com>
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain):
	  Try to get the new clock-rate from the buffer caps when we receive a new
	  payload type instead of always firing the signal. Fixes #512774.

2008-01-25 16:58:00 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not provided with caps but with a signal.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester@tester.ca>
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
	  (create_stream), (payload_type_change), (new_ssrc_pad_found):
	  Also handle lip-sync when the clock-rate is not provided with caps but
	  with a signal.

2008-01-25 16:00:52 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/: Remove the fixed clock-rate from the jitterbuffer and extend it so that a clock-rate can be provided...
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester@tester.ca>
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
	  (rtp_jitter_buffer_insert):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Remove the fixed clock-rate from the jitterbuffer and extend it so that
	  a clock-rate can be provided with each buffer instead. Fixes #511686.

2008-01-25 15:49:55 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester@tester.ca>
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state),
	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
	  Remove old unused variable.
	  Track pt on input buffers and get the clock-rate when it changes.
	  Ignore packets with unknown clock-rate. See #511686.

2008-01-25 01:44:27 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function.  Fixes #511920
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester@tester.ca>
	  * gst/rtpmanager/rtpsource.c: Fix unref of buffer using the
	  wrong function.  Fixes #511920

2008-01-11 17:02:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to parse the clock-rate instead of returning...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
	  If we find the caps in the cache, use it to parse the clock-rate instead
	  of returning an error. Fixes a TODO as found by Youness Alaoui.

2008-01-11 16:45:57 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	  gst/rtpmanager/: Make it possible to use different user_data for each of the callbacks.
	  Original commit message from CVS:
	  Patch by: Youness Alaoui <youness dot alaoui at collabora dot co dot uk>
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
	  (rtp_session_set_process_rtp_callback),
	  (rtp_session_set_send_rtp_callback),
	  (rtp_session_set_send_rtcp_callback),
	  (rtp_session_set_sync_rtcp_callback),
	  (rtp_session_set_clock_rate_callback),
	  (rtp_session_set_reconsider_callback), (source_push_rtp),
	  (source_clock_rate), (rtp_session_process_bye),
	  (rtp_session_process_rtcp), (rtp_session_send_bye),
	  (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsession.h:
	  Make it possible to use different user_data for each of the callbacks.
	  Fixes #508587.

2008-01-10 20:57:17 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c:
	  Fix documentation for latest patch

2008-01-10 14:34:30 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515)
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c:
	  Allow request_new_pad with name NULL (bug #508515)

2008-01-09 14:39:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do everything the upsteam peer pad can renegot...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (create_send_rtp_sink):
	  Don't set fixed caps, we can basically do everything the upsteam peer
	  pad can renegotiate to. Fixes #507940.

2008-01-04 18:47:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Don't unref the popped buffer when we don't have ownership.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_loop):
	  Don't unref the popped buffer when we don't have ownership.
	  Fixes #507020.

2007-12-31 13:12:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpssrcdemux.c: Don't clean up pads when going to PAUSED.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  (gst_rtp_ssrc_demux_change_state):
	  Don't clean up pads when going to PAUSED.

2007-12-12 16:59:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Clean up the dynamic pads when going to READY.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize),
	  (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
	  (gst_rtp_pt_demux_change_state):
	  * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset),
	  (gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query),
	  (gst_rtp_ssrc_demux_change_state):
	  Clean up the dynamic pads when going to READY.

2007-12-12 12:11:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Fix some leaks.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_finalize),
	  (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
	  (gst_rtp_bin_handle_message):
	  * gst/rtpmanager/rtpsession.c: (rtp_session_finalize),
	  (rtp_session_send_bye):
	  * gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
	  Fix some leaks.

2007-12-10 18:36:04 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Post a message when the SDES infor changes for a source.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
	  (gst_rtp_bin_handle_message):
	  * gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
	  (on_ssrc_sdes):
	  Post a message when the SDES infor changes for a source.
	  * gst/rtpmanager/rtpsession.c:
	  * gst/rtpmanager/rtpsource.c:
	  Update some comments.

2007-12-10 15:34:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Add signal to notify of an SDES change.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (on_ssrc_sdes), (create_session),
	  (gst_rtp_bin_class_init):
	  * gst/rtpmanager/gstrtpbin.h:
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpclient.h:
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  * gst/rtpmanager/gstrtpmanager.c:
	  * gst/rtpmanager/gstrtpptdemux.c:
	  * gst/rtpmanager/gstrtpptdemux.h:
	  * gst/rtpmanager/gstrtpsession.c: (on_ssrc_sdes),
	  (gst_rtp_session_class_init), (gst_rtp_session_init):
	  * gst/rtpmanager/gstrtpsession.h:
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  * gst/rtpmanager/gstrtpssrcdemux.h:
	  * gst/rtpmanager/rtpjitterbuffer.c:
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (on_ssrc_sdes), (rtp_session_process_sdes):
	  * gst/rtpmanager/rtpsession.h:
	  * gst/rtpmanager/rtpsource.c:
	  * gst/rtpmanager/rtpsource.h:
	  * gst/rtpmanager/rtpstats.c:
	  * gst/rtpmanager/rtpstats.h:
	  Add signal to notify of an SDES change.
	  Fix object type in the signal callbacks.

2007-12-10 14:03:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.*: Expose SDES items as properties and configure the session managers with them.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_session),
	  (gst_rtp_bin_class_init), (gst_rtp_bin_init), (sdes_type_to_name),
	  (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
	  (gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
	  * gst/rtpmanager/gstrtpbin.h:
	  Expose SDES items as properties and configure the session managers with
	  them.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
	  (rtp_source_set_property):
	  Fix SSRC property.

2007-12-10 11:08:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Update comment.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_session):
	  * gst/rtpmanager/rtpjitterbuffer.c:
	  Update comment.
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (gst_rtp_session_set_property), (gst_rtp_session_get_property):
	  Define some GObject properties to set SDES and other configuration.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (rtp_session_init), (rtp_session_finalize),
	  (rtp_session_set_property), (rtp_session_get_property),
	  (on_ssrc_sdes), (rtp_session_set_bandwidth),
	  (rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction),
	  (rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string),
	  (rtp_session_get_sdes_string), (obtain_source),
	  (rtp_session_get_internal_source), (rtp_session_process_sdes),
	  (rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes),
	  (is_rtcp_time):
	  * gst/rtpmanager/rtpsession.h:
	  Add signal when new SDES infor has been found for a source.
	  Create properties for SDES and other info.
	  Simplify the SDES API.
	  Add method for getting the internal source object of the session.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
	  (rtp_source_finalize), (rtp_source_set_property),
	  (rtp_source_get_property), (rtp_source_set_callbacks),
	  (rtp_source_get_ssrc), (rtp_source_set_as_csrc),
	  (rtp_source_is_as_csrc), (rtp_source_is_active),
	  (rtp_source_is_validated), (rtp_source_is_sender),
	  (rtp_source_received_bye), (rtp_source_get_bye_reason),
	  (rtp_source_set_sdes), (rtp_source_set_sdes_string),
	  (rtp_source_get_sdes), (rtp_source_get_sdes_string),
	  (rtp_source_get_new_sr), (rtp_source_get_new_rb):
	  * gst/rtpmanager/rtpsource.h:
	  Add GObject properties for various things.
	  Don't leak the bye reason.

2007-11-22 09:08:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer can buffer an unlimited amount of time and thus has no max_latency ...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_query):
	  jitterbuffer can buffer an unlimited amount of time and thus has no
	  max_latency requirements.

2007-11-02 21:45:38 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).
	  Original commit message from CVS:
	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
	  * gst/rtpmanager/gstrtpsession.c:
	  Fix bad function signatures (#492798).

2007-10-09 10:01:39 +0000  Laurent Glayal <spglegle@yahoo.fr>

	  gst/rtpmanager/gstrtpbin.c: Fix memleak. Fixes #484990.
	  Original commit message from CVS:
	  Patch by: Laurent Glayal <spglegle at yahoo dot fr>
	  * gst/rtpmanager/gstrtpbin.c: (create_stream),
	  (gst_rtp_bin_class_init):
	  Fix memleak. Fixes #484990.

2007-10-08 17:46:45 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/: Fix compiler warnings shown by Forte.
	  Original commit message from CVS:
	  * gst/librfb/rfbbuffer.c: (rfb_buffer_new_and_alloc):
	  * gst/librfb/rfbbuffer.h:
	  * gst/librfb/rfbdecoder.c: (rfb_socket_get_buffer):
	  * gst/mpegvideoparse/mpegvideoparse.c: (gst_mpegvideoparse_chain):
	  * gst/nsf/nes6502.c: (nes6502_execute):
	  * gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
	  * gst/real/gstrealvideodec.c: (open_library):
	  * gst/real/gstrealvideodec.h:
	  * gst/rtpmanager/gstrtpsession.c: (create_recv_rtp_sink),
	  (create_recv_rtcp_sink), (create_send_rtp_sink):
	  Fix compiler warnings shown by Forte.

2007-10-08 10:39:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (get_pt_map),
	  (gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init):
	  Fix caps refcounting for payload maps.
	  When clearing payload maps, also clear sessions and streams payload
	  maps.
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
	  (gst_rtp_pt_demux_clear_pt_map), (gst_rtp_pt_demux_chain),
	  (find_pad_for_pt):
	  Implement clearing the payload map.
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_event_send_rtp_sink):
	  Forward flush events instead of leaking them.
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  (gst_rtp_ssrc_demux_rtcp_sink_event):
	  Correctly refcount events before pushing them.

2007-10-05 17:26:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/rtpsession.c: When reconsidering RTCP timeouts, set the next timeout against the last report time inst...
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsession.c: (rtp_session_next_timeout),
	  When reconsidering RTCP timeouts, set the next timeout against the last
	  report time instead of the current clock time so that we don't end up
	  reconsidering forever.

2007-10-05 12:07:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead of popping it off, which allows us to grea...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
	  Only peek at the tail element instead of popping it off, which allows
	  us to greatly simplify things when the tail element changes.
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_event_recv_rtp_sink):
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  (gst_rtp_ssrc_demux_sink_event):
	  Forward FLUSH events instead of leaking them.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
	  (calculate_skew), (rtp_jitter_buffer_insert):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Remove the tail-changed callback in favour of a simple boolean when we
	  insert a buffer in the queue.
	  Add method to peek the tail of the buffer.

2007-10-02 10:27:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Remove some old unused variables.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_flush_start),
	  (gst_rtp_jitter_buffer_flush_stop),
	  (gst_rtp_jitter_buffer_change_state), (apply_offset),
	  (gst_rtp_jitter_buffer_loop):
	  Remove some old unused variables.
	  Don't add the latency to the skew corrected timestamp, latency is only
	  used to sync against the clock.
	  Improve debugging.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
	  (rtp_jitter_buffer_reset_skew), (calculate_skew):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Handle case where server timestamp goes backwards or wildly jumps by
	  temporarily pausing the skew correction.
	  Improve debugging.

2007-09-28 14:51:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Fix crasher in dispose.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (free_client):
	  Fix crasher in dispose.
	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
	  Handle cases where input buffers have no timestamps so that no clock
	  skew can be calculated, in this case interpollate timestamps based on
	  rtp timestamp and assume a 0 clock skew.

2007-09-28 11:17:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Remove jitter correction code, it's now in the lower level object.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c: (apply_latency),
	  (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query):
	  Remove jitter correction code, it's now in the lower level object.
	  Use new -core method for doing a peer query.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
	  (calculate_skew), (rtp_jitter_buffer_insert):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Move jitter correction to the lowlevel jitterbuffer.
	  Increase the max window size.
	  When filling the window, already start estimating the skew using a
	  parabolic weighting factor so that we have a much better startup
	  behaviour that gets more accurate with the more samples we have.
	  Increase the default weighting factor for the steady state to get
	  smoother timestamps.

2007-09-26 20:08:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Fix cleanup crasher.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_dispose),
	  (gst_rtp_bin_finalize):
	  Fix cleanup crasher.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
	  (calculate_skew):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Dynamically adjust the skew calculation window so that we calculate it
	  over a period of around 2 seconds.

2007-09-20 14:34:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. Fixes #478566.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (on_ssrc_active), (create_session),
	  (gst_rtp_bin_class_init):
	  * gst/rtpmanager/gstrtpbin.h:
	  * gst/rtpmanager/gstrtpsession.c: (on_ssrc_active),
	  (gst_rtp_session_class_init), (gst_rtp_session_init),
	  (gst_rtp_session_event_send_rtp_sink):
	  * gst/rtpmanager/gstrtpsession.h:
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (on_ssrc_active), (rtp_session_process_rb):
	  * gst/rtpmanager/rtpsession.h:
	  Add notification of active SSRCs to various RTP elements. Fixes #478566.

2007-09-17 02:01:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
	  Link to the right pads regardless of which one was created first in the
	  ssrc demuxer.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
	  (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
	  * gst/rtpmanager/rtpsource.c: (calculate_jitter):
	  Improve debugging.
	  * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
	  (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
	  (gst_rtp_ssrc_demux_sink_event),
	  (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
	  (gst_rtp_ssrc_demux_rtcp_chain),
	  (gst_rtp_ssrc_demux_internal_links):
	  * gst/rtpmanager/gstrtpssrcdemux.h:
	  Fix race in creating the RTP and RTCP pads when a new SSRC is detected.

2007-09-16 19:40:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property),
	  (gst_rtp_bin_get_property):
	  Use lock to protect variable.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init),
	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
	  (convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop):
	  Reconstruct GST timestamp from RTP timestamps based on measured clock
	  skew and sync offset.
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
	  (rtp_jitter_buffer_set_tail_changed),
	  (rtp_jitter_buffer_set_clock_rate),
	  (rtp_jitter_buffer_get_clock_rate), (calculate_skew),
	  (rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Measure clock skew.
	  Add callback to be notfied when a new packet was inserted at the tail.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
	  (calculate_jitter), (rtp_source_send_rtp):
	  * gst/rtpmanager/rtpsource.h:
	  Remove clock skew detection, it's move to the jitterbuffer now.

2007-09-15 18:48:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_session):
	  Also set NTP base time on new sessions.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query),
	  (gst_rtp_jitter_buffer_set_property),
	  (gst_rtp_jitter_buffer_get_property):
	  Use the right lock to protect our variables.
	  Fix some comment.
	  * gst/rtpmanager/gstrtpsession.c:
	  (gst_rtp_session_getcaps_send_rtp),
	  (gst_rtp_session_chain_send_rtp), (create_send_rtp_sink):
	  Implement getcaps on the sender sinkpad so that payloaders can negotiate
	  the right SSRC.

2007-09-12 21:23:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Various leak fixes.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_session), (free_session),
	  (get_client), (free_client), (gst_rtp_bin_associate),
	  (free_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_dispose),
	  (gst_rtp_bin_finalize):
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init),
	  (gst_rtp_jitter_buffer_finalize):
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_release):
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize),
	  (gst_rtp_session_set_property), (gst_rtp_session_chain_recv_rtp),
	  (gst_rtp_session_chain_send_rtp):
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_dispose):
	  * gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
	  * gst/rtpmanager/rtpsession.h:
	  Various leak fixes.

2007-09-12 18:04:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so that we can generate better
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base),
	  (gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp):
	  Calculate and configure the NTP base time so that we can generate better
	  NTP times in SR packets.
	  Set caps on new ghostpad.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_loop):
	  Clean debug statement.
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (gst_rtp_session_init), (gst_rtp_session_set_property),
	  (gst_rtp_session_get_property), (get_current_ntp_ns_time),
	  (rtcp_thread), (gst_rtp_session_event_recv_rtp_sink),
	  (gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp),
	  (gst_rtp_session_event_send_rtp_sink),
	  (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
	  (create_send_rtp_sink):
	  * gst/rtpmanager/gstrtpsession.h:
	  Add ntp-ns-base property to convert running_time to NTP time.
	  Handle NEWSEGMENT events on send and recv RTP pads so that we can
	  calculate the running time and thus NTP time of the packets.
	  Simplify getting the current NTP time using the pipeline clock.
	  Implement internal links functions.
	  Use the buffer timestamp to calculate the NTP time instead of the clock.
	  * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
	  (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
	  (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain),
	  (gst_rtp_ssrc_demux_internal_links),
	  (gst_rtp_ssrc_demux_src_query):
	  * gst/rtpmanager/gstrtpssrcdemux.h:
	  Implement internal links function.
	  Calculate the diff between different streams, this might be used later
	  to get the inter stream latency.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp):
	  Simple cleanup.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
	  (calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr):
	  Make the clock skew window a little bigger.
	  Apply the clock skew to all buffers, not just one with a new timestamp.
	  Calculate and debug sender clock drift.
	  Use extended last timestamp to interpollate for SR reports.

2007-09-04 15:23:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562).
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c:
	  Make compiler happy: fix compilation with -Wall -Werror
	  (#473562).

2007-09-03 21:19:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Updated example pipelines in docs.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin-marshal.list:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client),
	  (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream),
	  (gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found),
	  (create_recv_rtp), (create_recv_rtcp), (create_send_rtp):
	  * gst/rtpmanager/gstrtpbin.h:
	  Updated example pipelines in docs.
	  Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync.
	  Set the default latency correctly.
	  Add some more points where we can get caps.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init),
	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
	  (gst_rtp_jitter_buffer_query),
	  (gst_rtp_jitter_buffer_set_property),
	  (gst_rtp_jitter_buffer_get_property):
	  Add ts-offset property to control timestamping.
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (gst_rtp_session_init), (gst_rtp_session_set_property),
	  (gst_rtp_session_get_property), (get_current_ntp_ns_time),
	  (rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state),
	  (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp),
	  (gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate),
	  (gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp),
	  (gst_rtp_session_event_send_rtp_sink),
	  (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
	  (create_recv_rtcp_sink), (create_send_rtp_sink),
	  (create_send_rtcp_src):
	  Various cleanups.
	  Feed rtpsession manager with NTP time based on pipeline clock when
	  handling RTP packets and RTCP timeouts.
	  Perform all RTCP with the system clock.
	  Set caps on RTCP outgoing buffers.
	  * gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc),
	  (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
	  (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
	  (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
	  (gst_rtp_ssrc_demux_rtcp_chain):
	  * gst/rtpmanager/gstrtpssrcdemux.h:
	  Also demux RTCP messages.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
	  (update_arrival_stats), (rtp_session_process_rtp),
	  (rtp_session_process_rb), (rtp_session_process_sr),
	  (rtp_session_process_rr), (rtp_session_process_rtcp),
	  (rtp_session_send_rtp), (rtp_session_send_bye),
	  (session_start_rtcp), (session_report_blocks), (session_cleanup),
	  (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsession.h:
	  Remove the get_time callback, the GStreamer part will feed us with
	  enough timing information.
	  Split sync timing and RTCP timing information.
	  Factor out common RB handling for SR and RR.
	  Send out SR RTCP packets for lip-sync.
	  Move SR and RR packet info generation to the source.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
	  (rtp_source_update_caps), (get_clock_rate), (calculate_jitter),
	  (rtp_source_process_rtp), (rtp_source_send_rtp),
	  (rtp_source_process_sr), (rtp_source_process_rb),
	  (rtp_source_get_new_sr), (rtp_source_get_new_rb),
	  (rtp_source_get_last_sr):
	  * gst/rtpmanager/rtpsource.h:
	  * gst/rtpmanager/rtpstats.h:
	  Use caps on incomming buffers to get timing information when they are
	  there.
	  Calculate clock scew of the receiver compared to the sender and adjust
	  the rtp timestamps.
	  Calculate the round trip in sources.
	  Do SR and RR calculations in the source.

2007-08-31 15:26:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Use extended timestamp to release buffers from the jitterbuffer so that we can h...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_flush_stop),
	  (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop):
	  Use extended timestamp to release buffers from the jitterbuffer so that
	  we can handle the rtp wraparound correctly.

2007-08-29 16:56:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_loop):
	  Improve Comments.
	  * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
	  (gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
	  (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
	  (gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
	  (create_send_rtp_sink):
	  Also parse the sink caps for clock-rate instead of only relying on the
	  result of the signal.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
	  Make sure we fetch the clock rate for payloads we are sending out so
	  that we can use it for SR reports.

2007-08-29 01:22:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.*: Distribute synchronisation parameters to the session manager so that it can generate ...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
	  (gst_rtp_session_change_state),
	  (gst_rtp_session_event_send_rtp_sink):
	  * gst/rtpmanager/gstrtpsession.h:
	  Distribute synchronisation parameters to the session manager so that it
	  can generate correct SR packets for lip-sync.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time),
	  (rtp_session_set_timestamp_sync), (session_start_rtcp):
	  * gst/rtpmanager/rtpsession.h:
	  Add methods for setting sync parameters.
	  Set correct RTP time in SR packets using the sync params.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
	  * gst/rtpmanager/rtpsource.h:
	  Record last RTP <-> GST timestamp so that we can use them to convert NTP
	  to RTP timestamps in SR packets.

2007-08-28 20:30:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
	  Add some more advanced example pipelines.
	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
	  (stop_rtcp_thread), (gst_rtp_session_send_rtcp):
	  Add some debug and FIXME.
	  Release LOCK when performing session cleanup.
	  * gst/rtpmanager/rtpsession.c: (session_report_blocks):
	  Add some debug.
	  * gst/rtpmanager/rtpsource.c: (calculate_jitter),
	  (rtp_source_send_rtp):
	  Make sure we always send RTP packets with the session SSRC.

2007-08-27 21:17:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: When synchronizing buffers, take peer latency into account.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop),
	  (gst_rtp_jitter_buffer_query):
	  When synchronizing buffers, take peer latency into account.
	  Don't try to add our latency to invalid peer max latency values.

2007-08-23 21:39:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers a GType that's different than the GstRTPF...
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * gst/rtpmanager/gstrtpbin.c:
	  * gst/rtpmanager/gstrtpbin.h:
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpclient.h:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  * gst/rtpmanager/gstrtpptdemux.c:
	  * gst/rtpmanager/gstrtpptdemux.h:
	  * gst/rtpmanager/gstrtpsession.c:
	  * gst/rtpmanager/gstrtpsession.h:
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  * gst/rtpmanager/gstrtpssrcdemux.h:
	  Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE
	  registers a GType that's different than the GstRTPFoo types that
	  farsight registers (luckily GType names are case sensitive). Should
	  finally fix #430664.

2007-08-21 17:18:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: When drop-on-latency is set but we have no latency configured, just push the buf...
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain),
	  (gst_rtp_jitter_buffer_set_property):
	  When drop-on-latency is set but we have no latency configured, just push
	  the buffer as fast as possible.
	  Fix typo in comment.

2007-08-21 16:04:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/rtpjitterbuffer.*: Fix undefined overflow prone ts_diff handling.
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpjitterbuffer.c:
	  (rtp_jitter_buffer_get_ts_diff):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Fix undefined overflow prone ts_diff handling.

2007-08-16 11:40:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
	  (gst_rtp_jitter_buffer_loop):
	  Fix EOS handling.
	  Convert some DEBUG into WARNINGs.
	  Pause task when flushing.
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
	  Use system clock for RTCP session management timeouts.
	  * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
	  (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
	  Release the session lock when emiting signals.

2007-08-13 06:16:40 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/rtpmanager/rtpjitterbuffer.c: Include stdlib.
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpjitterbuffer.c:
	  Include stdlib.

2007-08-10 17:16:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some...
	  Original commit message from CVS:
	  * gst/rtpmanager/Makefile.am:
	  * gst/rtpmanager/async_jitter_queue.c:
	  * gst/rtpmanager/async_jitter_queue.h:
	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init),
	  (rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize),
	  (rtp_jitter_buffer_new), (compare_seqnum),
	  (rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop),
	  (rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets),
	  (rtp_jitter_buffer_get_ts_diff):
	  * gst/rtpmanager/rtpjitterbuffer.h:
	  Remove complicated async queue and replace with more simple jitterbuffer
	  code while also fixing some bugs.
	  * gst/rtpmanager/gstrtpbin-marshal.list:
	  * gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision),
	  (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout),
	  (create_session), (gst_rtp_bin_class_init), (create_recv_rtp),
	  (create_send_rtp):
	  * gst/rtpmanager/gstrtpbin.h:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose),
	  (gst_jitter_buffer_sink_parse_caps),
	  (gst_rtp_jitter_buffer_flush_start),
	  (gst_rtp_jitter_buffer_flush_stop),
	  (gst_rtp_jitter_buffer_change_state),
	  (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
	  (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property):
	  * gst/rtpmanager/gstrtpsession.c: (on_new_ssrc),
	  (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc),
	  (on_bye_timeout), (on_timeout), (gst_rtp_session_class_init),
	  (gst_rtp_session_init):
	  * gst/rtpmanager/gstrtpsession.h:
	  * gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup):
	  Use new jitterbuffer code.
	  Expose some new signals in preparation for handling EOS.

2007-07-18 07:35:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add stdlib include (free, atoi, exit).
	  Original commit message from CVS:
	  * examples/app/appsrc_ex.c:
	  * examples/switch/switcher.c:
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/timidity/gstwildmidi.c:
	  * ext/x264/gstx264enc.c:
	  * gst/mve/mveaudioenc.c: (mve_compress_audio):
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/spectrum/demo-audiotest.c:
	  * gst/spectrum/demo-osssrc.c:
	  * sys/dvb/gstdvbsrc.c:
	  Add stdlib include (free, atoi, exit).

2007-06-22 20:23:18 +0000  Jens Granseuer <jensgr@gmx.net>

	  gst/: Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
	  Original commit message from CVS:
	  Patch by: Jens Granseuer  <jensgr at gmx net>
	  * gst/equalizer/gstiirequalizer.c:
	  * gst/equalizer/gstiirequalizer10bands.c:
	  * gst/equalizer/gstiirequalizer3bands.c:
	  * gst/equalizer/gstiirequalizernbands.c:
	  * gst/rtpmanager/async_jitter_queue.c:
	  (async_jitter_queue_push_sorted):
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain):
	  * gst/switch/gstswitch.c: (gst_switch_chain):
	  Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
	  Fixes #450185.

2007-05-28 16:37:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream),
	  (gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp),
	  (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad):
	  * gst/rtpmanager/gstrtpclient.c: (create_stream),
	  (gst_rtp_client_request_new_pad):
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
	  * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
	  * gst/rtpmanager/gstrtpptdemux.c:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (gst_rtp_session_request_new_pad):
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  Rename elements to avoid conflict with farsight elements with the same
	  name. Fixes #430664.

2007-05-23 13:08:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Document stuff.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
	  * gst/rtpmanager/gstrtpbin.h:
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init),
	  (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
	  (gst_rtp_pt_demux_clear_pt_map):
	  * gst/rtpmanager/gstrtpptdemux.h:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (rtcp_thread), (gst_rtp_session_clear_pt_map):
	  * gst/rtpmanager/gstrtpsession.h:
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  (gst_rtp_ssrc_demux_class_init):
	  Document stuff.
	  Add clear-pt-map action signal where needed.

2007-05-15 13:29:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpptdemux.c: We always use fixed caps.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
	  We always use fixed caps.

2007-05-15 03:45:45 +0000  David Schleef <ds@schleef.org>

	  gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12.  Work around.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c:
	  g_hash_table_remove_all() only exists in 2.12.  Work around.

2007-05-14 15:28:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/async_jitter_queue.c: Fix leak when flushing.
	  Original commit message from CVS:
	  * gst/rtpmanager/async_jitter_queue.c:
	  (async_jitter_queue_set_flushing_unlocked):
	  Fix leak when flushing.
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map),
	  (gst_rtp_bin_class_init):
	  * gst/rtpmanager/gstrtpbin.h:
	  Add clear-pt-map signal.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_flush_stop),
	  (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_loop):
	  Init clock-rate to -1 to mark unknow clock rate.
	  Fix flushing.

2007-05-10 14:02:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
	  gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
	  gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
	  gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
	  qtdemux_parse_segments, qtdemux_parse_trak):
	  * gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
	  rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
	  rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
	  rtp_session_get_location, rtp_session_get_tool,
	  rtp_session_process_bye, session_report_blocks):
	  * gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
	  rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
	  More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
	  * gst/switch/Makefile.am:
	  Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).

2007-05-10 12:38:49 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	* gst/rtpmanager/async_jitter_queue.c:
	  gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, a...
	  Original commit message from CVS:
	  * gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration,
	  async_jitter_queue_ref, async_jitter_queue_ref_unlocked,
	  async_jitter_queue_set_low_threshold,
	  async_jitter_queue_length_ts_units_unlocked,
	  async_jitter_queue_unref_and_unlock, async_jitter_queue_unref,
	  async_jitter_queue_lock, async_jitter_queue_push,
	  async_jitter_queue_push_unlocked, async_jitter_queue_push_sorted,
	  async_jitter_queue_pop_intern_unlocked, async_jitter_queue_pop,
	  async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked,
	  async_jitter_queue_set_flushing_unlocked,
	  async_jitter_queue_unset_flushing_unlocked):
	  Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)

2007-05-09 11:24:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_query):
	  Pass queries upstream.

2007-05-04 12:32:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug info.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_query):
	  Add some debug info.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_init),
	  (rtp_session_send_rtp):
	  Store real user name in the session.

2007-04-30 13:41:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.
	  Original commit message from CVS:
	  * gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
	  (async_jitter_queue_pop_intern_unlocked):
	  Fix the case where the buffer underruns and does not block.
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
	  (create_recv_rtcp), (create_send_rtp), (create_rtcp),
	  (gst_rtp_bin_request_new_pad):
	  Rename RTCP send pad, like in the session manager.
	  Allow getting an RTCP pad for receiving even if we don't receive RTP.
	  fix handling of send_rtp_src pad.
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
	  When no pt map could be found, fall back to the sinkpad caps.
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
	  (gst_rtp_session_send_rtp), (create_recv_rtp_sink),
	  (create_recv_rtcp_sink), (create_send_rtp_sink),
	  (create_send_rtcp_src):
	  Fix pad names.
	  * gst/rtpmanager/rtpsession.c: (source_push_rtp),
	  (rtp_session_create_source), (rtp_session_process_sr),
	  (rtp_session_send_rtp), (session_start_rtcp):
	  * gst/rtpmanager/rtpsession.h:
	  Unlock session when performing a callback.
	  Add callbacks for the internal session object.
	  Fix sending of RTP packets.
	  first attempt at adding NTP times in the SR packets.
	  Small debug and doc improvements.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
	  Update stats for SR reports.

2007-04-29 14:46:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Remove debug.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):
	  Remove debug.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
	  (rtp_session_process_sdes), (calculate_rtcp_interval),
	  (rtp_session_next_timeout), (session_report_blocks):
	  * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
	  Improve debugging
	  Fix interval for BYE/RTCP packets.

2007-04-27 15:09:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
	  (gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider):
	  Move reconsideration code to the rtpsession object.
	  Simplify timout handling and add reconsideration.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (rtp_session_init), (rtp_session_finalize), (on_bye_ssrc),
	  (on_bye_timeout), (on_timeout), (rtp_session_set_callbacks),
	  (obtain_source), (rtp_session_create_source),
	  (update_arrival_stats), (rtp_session_process_rtp),
	  (rtp_session_process_sr), (rtp_session_process_rr),
	  (rtp_session_process_bye), (rtp_session_process_rtcp),
	  (calculate_rtcp_interval), (rtp_session_send_bye),
	  (rtp_session_next_timeout), (session_start_rtcp),
	  (session_report_blocks), (session_cleanup), (session_sdes),
	  (session_bye), (is_rtcp_time), (rtp_session_on_timeout):
	  * gst/rtpmanager/rtpsession.h:
	  Handle timeout of inactive sources and senders.
	  Implement BYE scheduling.
	  * gst/rtpmanager/rtpsource.c: (calculate_jitter),
	  (rtp_source_process_sr), (rtp_source_get_last_sr),
	  (rtp_source_get_last_rb):
	  * gst/rtpmanager/rtpsource.h:
	  Add members to check for timeouts.
	  * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
	  (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter),
	  (rtp_stats_calculate_bye_interval):
	  * gst/rtpmanager/rtpstats.h:
	  Use RFC algorithm for calculating the reporting interval.

2007-04-25 16:38:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
	  Implement forward and reverse reconsideration.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources),
	  (rtp_session_get_num_active_sources), (rtp_session_process_sr),
	  (session_report_blocks):
	  * gst/rtpmanager/rtpsession.h:
	  Small cleanups.

2007-04-25 15:48:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.*: Make default jitterbuffer latency configurable.
	  Original commit message from CVS:
	  reviewed by: <delete if not using a buddy>
	  * gst/rtpmanager/gstrtpbin.c: (create_stream),
	  (gst_rtp_bin_class_init), (gst_rtp_bin_set_property),
	  (gst_rtp_bin_get_property):
	  * gst/rtpmanager/gstrtpbin.h:
	  Make default jitterbuffer latency configurable.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init),
	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
	  (gst_rtp_jitter_buffer_set_property),
	  (gst_rtp_jitter_buffer_get_property):
	  Debuging cleanups.

2007-04-25 13:19:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_change_state):
	  Report NO_PREROLL when going to PAUSED.
	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
	  Don't send RTCP right before we are shutting down.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
	  (rtp_session_process_sr), (session_report_blocks),
	  (rtp_session_perform_reporting):
	  Improve report blocks.
	  * gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
	  (rtp_source_process_rtp), (rtp_source_process_sr),
	  (rtp_source_process_rb), (rtp_source_get_last_sr),
	  (rtp_source_get_last_rb):
	  * gst/rtpmanager/rtpsource.h:
	  * gst/rtpmanager/rtpstats.h:
	  Cleanups, add methods to access stats.

2007-04-25 08:30:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: fix for pad name change
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_rtcp):
	  fix for pad name change
	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
	  (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
	  Fix for renamed methods.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_init),
	  (rtp_session_finalize), (rtp_session_set_cname),
	  (rtp_session_get_cname), (rtp_session_set_name),
	  (rtp_session_get_name), (rtp_session_set_email),
	  (rtp_session_get_email), (rtp_session_set_phone),
	  (rtp_session_get_phone), (rtp_session_set_location),
	  (rtp_session_get_location), (rtp_session_set_tool),
	  (rtp_session_get_tool), (rtp_session_set_note),
	  (rtp_session_get_note), (source_push_rtp), (obtain_source),
	  (rtp_session_add_source), (rtp_session_get_source_by_ssrc),
	  (rtp_session_create_source), (rtp_session_process_rtp),
	  (rtp_session_process_sr), (rtp_session_process_sdes),
	  (rtp_session_process_rtcp), (rtp_session_send_rtp),
	  (rtp_session_get_reporting_interval), (session_report_blocks),
	  (session_sdes), (rtp_session_perform_reporting):
	  * gst/rtpmanager/rtpsession.h:
	  Prepare for implementing SSRC sampling.
	  Create SSRC for the session.
	  Add methods to set the SDES entries.
	  fix accounting of senders/receivers.
	  Implement SR/RR/SDES RTCP reporting.
	  * gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
	  (rtp_source_process_rtp), (rtp_source_process_sr):
	  * gst/rtpmanager/rtpsource.h:
	  Implement extended sequence number.
	  * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
	  * gst/rtpmanager/rtpstats.h:
	  Rename some fields.

2007-04-21 19:21:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtpmanager/rtpsession.c: Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
	  Original commit message from CVS:
	  * gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
	  Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.

2007-04-18 18:58:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  configure.ac: Disable rtpmanager for now because it depends on CVS -base.
	  Original commit message from CVS:
	  * configure.ac:
	  Disable rtpmanager for now because it depends on CVS -base.
	  * gst/rtpmanager/Makefile.am:
	  Added new files for session manager.
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
	  (create_stream), (pt_map_requested), (new_ssrc_pad_found):
	  Some cleanups.
	  the session manager can now also request a pt-map.
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
	  (gst_rtp_session_class_init), (gst_rtp_session_init),
	  (gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
	  (stop_rtcp_thread), (gst_rtp_session_change_state),
	  (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
	  (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
	  (gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
	  (gst_rtp_session_chain_recv_rtp),
	  (gst_rtp_session_event_recv_rtcp_sink),
	  (gst_rtp_session_chain_recv_rtcp),
	  (gst_rtp_session_event_send_rtp_sink),
	  (gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
	  (gst_rtp_session_request_new_pad):
	  * gst/rtpmanager/gstrtpsession.h:
	  We can ask for pt-map now too when the session manager needs it.
	  Hook up to the new session manager, implement the needed callbacks for
	  pushing data, getting clock time and requesting clock-rates.
	  Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
	  be send to clients.
	  Add code to start and stop the thread that will schedule RTCP through
	  the session manager.
	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
	  (rtp_session_init), (rtp_session_finalize),
	  (rtp_session_set_property), (rtp_session_get_property),
	  (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
	  (on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
	  (rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
	  (rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
	  (source_push_rtp), (source_clock_rate), (check_collision),
	  (obtain_source), (rtp_session_add_source),
	  (rtp_session_get_num_sources),
	  (rtp_session_get_num_active_sources),
	  (rtp_session_get_source_by_ssrc),
	  (rtp_session_get_source_by_cname), (rtp_session_create_source),
	  (update_arrival_stats), (rtp_session_process_rtp),
	  (rtp_session_process_sr), (rtp_session_process_rr),
	  (rtp_session_process_sdes), (rtp_session_process_bye),
	  (rtp_session_process_app), (rtp_session_process_rtcp),
	  (rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
	  (rtp_session_produce_rtcp):
	  * gst/rtpmanager/rtpsession.h:
	  The advanced beginnings of the main session manager that handles the
	  participant database of RTPSources, SSRC probation, SSRC collisions,
	  parse RTCP to update source stats. etc..
	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
	  (rtp_source_init), (rtp_source_finalize), (rtp_source_new),
	  (rtp_source_set_callbacks), (rtp_source_set_as_csrc),
	  (rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
	  (push_packet), (get_clock_rate), (calculate_jitter),
	  (rtp_source_process_rtp), (rtp_source_process_bye),
	  (rtp_source_send_rtp), (rtp_source_process_sr),
	  (rtp_source_process_rb):
	  * gst/rtpmanager/rtpsource.h:
	  Object that encapsulates an SSRC and its state in the database.
	  Calculates the jitter and transit times of data packets.
	  * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
	  (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
	  * gst/rtpmanager/rtpstats.h:
	  Various stats regarding the session and sources.
	  Used to calculate the RTCP interval.

2007-04-13 09:20:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Protect lists and structures with locks.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
	  (gst_rtp_bin_init), (gst_rtp_bin_finalize), (new_ssrc_pad_found),
	  (create_recv_rtp), (gst_rtp_bin_request_new_pad):
	  * gst/rtpmanager/gstrtpbin.h:
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (gst_rtp_session_init), (gst_rtp_session_finalize),
	  (gst_rtp_session_event_recv_rtp_sink),
	  (gst_rtp_session_event_recv_rtcp_sink),
	  (gst_rtp_session_chain_recv_rtcp),
	  (gst_rtp_session_request_new_pad):
	  Protect lists and structures with locks.
	  Return FLOW_OK from RTCP messages for now.

2007-04-12 08:18:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
	  (create_stream), (gst_rtp_bin_class_init), (pt_map_requested):
	  Emit pt map requests and cache results.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init),
	  (gst_jitter_buffer_sink_parse_caps),
	  (gst_jitter_buffer_sink_setcaps),
	  (gst_rtp_jitter_buffer_get_clock_rate),
	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
	  Emit request-pt-map signals.

2007-04-11 13:49:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin-marshal.list:
	  Some more custom marshallers.
	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
	  (clock_rate_request), (create_stream), (gst_rtp_bin_class_init),
	  (pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp):
	  * gst/rtpmanager/gstrtpbin.h:
	  Prepare for caching pt maps.
	  Connect to signals to collect pt maps.
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_class_init),
	  (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop):
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  Add request_clock_rate signal.
	  Use scale insteat of scale_int because the later does not deal with
	  negative numbers.
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
	  (gst_rtp_pt_demux_chain):
	  * gst/rtpmanager/gstrtpptdemux.h:
	  Implement request-pt-map signal.

2007-04-10 09:14:07 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Added custom marshallers for signals.
	  Original commit message from CVS:
	  * gst/rtpmanager/.cvsignore:
	  * gst/rtpmanager/Makefile.am:
	  * gst/rtpmanager/gstrtpbin-marshal.list:
	  Added custom marshallers for signals.
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
	  * gst/rtpmanager/gstrtpbin.h:
	  Prepare for emiting pt map signals.
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init):
	  * gst/rtpmanager/gstrtpssrcdemux.c:
	  (gst_rtp_ssrc_demux_class_init):
	  Fix signals.

2007-04-06 12:28:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.*: Provide a clock.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
	  (gst_rtp_bin_init), (gst_rtp_bin_provide_clock):
	  * gst/rtpmanager/gstrtpbin.h:
	  Provide a clock.

2007-04-06 12:07:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.c: Fix pad template name parsing.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (create_rtcp):
	  Fix pad template name parsing.

2007-04-05 16:10:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug and comments.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain),
	  (gst_rtp_jitter_buffer_loop):
	  Add some debug and comments.
	  Fix double unref() in error cases.

2007-04-05 13:54:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/gstrtpbin.*: Add debugging category.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
	  (create_session), (find_stream_by_ssrc), (create_stream),
	  (gst_rtp_bin_class_init), (new_payload_found),
	  (new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp),
	  (create_send_rtp), (create_rtcp):
	  * gst/rtpmanager/gstrtpbin.h:
	  Add debugging category.
	  Added RTPStream to manage stream per SSRC, each with its own
	  jitterbuffer and ptdemux.
	  Added SSRCDemux.
	  Connect to various SSRC and PT signals and create ghostpads, link stuff.
	  * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
	  Added rtpbin to elements.
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
	  Fix caps and forward GstFlowReturn
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
	  (gst_rtp_session_event_recv_rtp_sink),
	  (gst_rtp_session_chain_recv_rtp),
	  (gst_rtp_session_event_recv_rtcp_sink),
	  (gst_rtp_session_chain_recv_rtcp),
	  (gst_rtp_session_event_send_rtp_sink),
	  (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
	  (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
	  (gst_rtp_session_request_new_pad):
	  Add debug category.
	  Add event handling
	  * gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc),
	  (create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init),
	  (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain),
	  (gst_rtp_ssrc_demux_change_state):
	  * gst/rtpmanager/gstrtpssrcdemux.h:
	  Add debug category.
	  Add new-pt-pad signal.

2007-04-04 10:23:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Added simple SSRC demuxer.
	  Original commit message from CVS:
	  * gst/rtpmanager/Makefile.am:
	  * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
	  * gst/rtpmanager/gstrtpssrcdemux.c: (find_pad_for_ssrc),
	  (create_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
	  (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init),
	  (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event),
	  (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_src_event),
	  (gst_rtp_ssrc_demux_change_state):
	  * gst/rtpmanager/gstrtpssrcdemux.h:
	  Added simple SSRC demuxer.

2007-04-03 11:35:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/: Some more ghostpad magic.
	  Original commit message from CVS:
	  * gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
	  (create_session), (gst_rtp_bin_base_init), (create_recv_rtp),
	  (create_recv_rtcp), (create_send_rtp), (create_rtcp),
	  (gst_rtp_bin_request_new_pad):
	  * gst/rtpmanager/gstrtpbin.h:
	  * gst/rtpmanager/gstrtpclient.c:
	  Some more ghostpad magic.

2007-04-03 09:51:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly.
	  Original commit message from CVS:
	  * gst/rtpmanager/Makefile.am:
	  Add .h file so it can be disted properly.

2007-04-03 09:13:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Add RTP session management elements. Still in progress.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/rtpmanager/Makefile.am:
	  * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
	  (signal_waiting_threads), (async_jitter_queue_ref),
	  (async_jitter_queue_ref_unlocked),
	  (async_jitter_queue_set_low_threshold),
	  (async_jitter_queue_set_high_threshold),
	  (async_jitter_queue_set_max_queue_length),
	  (async_jitter_queue_get_g_queue), (calculate_ts_diff),
	  (async_jitter_queue_length_ts_units_unlocked),
	  (async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
	  (async_jitter_queue_lock), (async_jitter_queue_unlock),
	  (async_jitter_queue_push), (async_jitter_queue_push_unlocked),
	  (async_jitter_queue_push_sorted),
	  (async_jitter_queue_push_sorted_unlocked),
	  (async_jitter_queue_insert_after_unlocked),
	  (async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
	  (async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
	  (async_jitter_queue_length_unlocked),
	  (async_jitter_queue_set_flushing_unlocked),
	  (async_jitter_queue_unset_flushing_unlocked),
	  (async_jitter_queue_set_blocking_unlocked):
	  * gst/rtpmanager/async_jitter_queue.h:
	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
	  (gst_rtp_bin_class_init), (gst_rtp_bin_init),
	  (gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
	  (gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
	  (gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
	  * gst/rtpmanager/gstrtpbin.h:
	  * gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
	  (free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
	  (gst_rtp_client_class_init), (gst_rtp_client_init),
	  (gst_rtp_client_finalize), (gst_rtp_client_set_property),
	  (gst_rtp_client_get_property), (gst_rtp_client_change_state),
	  (gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
	  * gst/rtpmanager/gstrtpclient.h:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_base_init),
	  (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
	  (gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
	  (gst_jitter_buffer_sink_setcaps), (free_func),
	  (gst_rtp_jitter_buffer_flush_start),
	  (gst_rtp_jitter_buffer_flush_stop),
	  (gst_rtp_jitter_buffer_src_activate_push),
	  (gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
	  (compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
	  (gst_rtp_jitter_buffer_query),
	  (gst_rtp_jitter_buffer_set_property),
	  (gst_rtp_jitter_buffer_get_property):
	  * gst/rtpmanager/gstrtpjitterbuffer.h:
	  * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
	  (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
	  (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
	  (gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
	  (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
	  (gst_rtp_pt_demux_change_state):
	  * gst/rtpmanager/gstrtpptdemux.h:
	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
	  (gst_rtp_session_class_init), (gst_rtp_session_init),
	  (gst_rtp_session_finalize), (gst_rtp_session_set_property),
	  (gst_rtp_session_get_property), (gst_rtp_session_change_state),
	  (gst_rtp_session_chain_recv_rtp),
	  (gst_rtp_session_chain_recv_rtcp),
	  (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
	  (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
	  (gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad):
	  * gst/rtpmanager/gstrtpsession.h:
	  Add RTP session management elements. Still in progress.

2009-08-10 13:30:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: push mode; cater for chunk padding

2009-08-04 19:45:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: only use stream's pad after having checked it exists

2009-08-04 13:38:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: sprinkle some more GST_DEBUG_FUNCPTR

2009-08-04 13:36:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: post error message if no pads to push EOS event on

2009-08-04 11:39:59 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix typo in warning message

2009-08-04 11:39:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix some buffer ref handling

2009-08-04 11:37:16 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: do not exceed maximum number of supported streams

2009-08-04 11:35:18 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: prevent double unref; gst_avi_demux_parse_avih already unrefs

2009-08-04 11:32:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: verify size of INFO LIST to satisfy subsequent expectations

2009-07-29 15:25:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: check video stream framerate against avi header frame duration
	  The former might be bogus in silly cases, and the latter seems to
	  carry more weight.

2009-08-04 12:16:13 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: streamline stream duration calculation

2009-07-03 14:04:13 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/raw1394/gstdv1394src.c:
	  dv1394src: Fix element for live usage... which has been broken for 2 years :(
	  This is a live source, therefore:
	  * Use GST_FORMAT_TIME as the default format
	  * set_timestamp to True
	  * properly implement query latency.
	  This allows expected live usage like : playbin2 uri=dv://

2009-08-09 09:43:41 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/raw1394/gstdv1394src.c:
	  raw1394: Remove unneeded variable

2009-08-09 09:43:29 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-demux.c:
	  matroska: remove dead assignments

2009-08-09 09:43:00 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	  rtp: Remove dead assignments and resulting unneeded variables.

2009-08-10 09:53:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/wavpack/Makefile.am:
	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackenc.h:
	* ext/wavpack/md5.c:
	* ext/wavpack/md5.h:
	  wavpack: Use GLib GChecksum instead of our own MD5 implementation
	  This requires GLib 2.16 but that version is already required by core anyway.

2009-08-08 00:47:48 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroska: Adds support to muxing/demuxing WMA
	  Adds support for muxing wma audio family and fixes
	  demuxing of wma family in matroskademux. matroskademux
	  was broken because it missed codec_data.

2009-08-06 20:15:17 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/matroska/matroska-mux.c:
	  matroskamux: adds support for wmv family
	  Adds support to WMV1, WMV2, WMV3 and other family formats that
	  are signaled by the 'format' field in the caps (i.e. WVC1).
	  Partially fixes #576378

2009-08-09 14:19:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: if max == min width/height put an int in the probed caps, not an int range
	  Fixes #560033.

2009-08-09 13:58:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/osxaudio/gstosxaudiosrc.c:
	  osxaudiosrc: if max_channels == min_channels, use an int instead of an int range in the caps

2009-08-09 12:52:17 +0200  LoneStar <lone@auvtech.com>

	* gst/id3demux/id3v2frames.c:
	  id3demux: Try GST_*_TAG_ENCODING and locale encoding if tags are not UTF8
	  Fixes bug #499242.

2009-08-09 01:29:50 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: bump core/base requirements to latest release
	  To avoid confusion.

2009-08-09 01:27:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/flvmux.c:
	  check: fix flvmux unit test on big endian machines
	  flvmux only accepts raw audio in little endian, but audiotestsrc
	  produces audio in the native endianness, which makes linking
	  between audiotestsrc and flvmux fail on big endian machines. Add
	  an audioconvert element in between the two to fix this.

2009-02-15 18:49:44 +0000  Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroska: add kate subtitle support to matroska muxer and demuxer
	  See #525743.

2009-08-07 16:51:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3v2.3.0.html:
	  id3demux: add ID3 v2.3 spec as well

2009-08-07 16:42:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3v2frames.c:
	  id3demux: sizes in ID3 v2.3 are unlikely to be sync-safe integers
	  In ID3 v2.3 compressed frames will have a 4-byte data length indicator
	  after the frame header to indicate the size of the decompressed data.
	  This integer is unlikely to be a sync-safe integer for v2.3 tags,
	  only in v2.4 it's sync-safe.

2009-08-07 16:36:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3tags.c:
	  id3demux: fix typo in debug message

2009-08-07 16:02:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3tags.c:
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c:
	* tests/check/elements/id3demux.c:
	* tests/files/Makefile.am:
	* tests/files/id3-588148-unsynced-v24.tag:
	  id3demux: fix parsing of unsync'ed ID3 v2.4 tags and frames
	  Reversing the unsynchronisation seems to work slightly differently
	  for ID3 v2.3 tags and v2.4 tags: v2.3 tags don't have syncsafe frame
	  sizes in the frame header, so the unsynchronisation is applied to
	  the whole frame data including all the frame headers. v2.4 frames
	  have sync-safe sizes, however, so the unsynchronisation only needs
	  to be applied to the actual frame data, and it seems that's what's
	  being done as well. So we need to undo the unsynchronisation on a
	  per-frame basis for v2.4 tags for things to work properly.
	  Fixes extraction of coverart/images from APIC frames in ID3 v2.4
	  tags (#588148).
	  Add unit test for this as well.

2009-08-06 21:24:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Use SOUP_METHOD_GET instead of "GET" string
	  Fixes bug #590970.

2009-08-06 13:00:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: set the default slave method to skew
	  Set the default slave method to the much better skew algorithm. This is the
	  default in the new base class but we override this here as well for the
	  upcomming release.

2009-08-06 10:20:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: fix compilation with --disable-gst-debug

2009-08-03 18:59:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: use array instead of queue

2009-08-03 18:55:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: push NALs only after SPS/PPS
	  parse complete (bytestream) buffer for SPS/PPS before pushing NALs.
	  Fixes #564501.

2009-08-04 14:44:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/v4l2_calls.h:
	  v4l2: Directly use GST_PTR_FORMAT for printing caps with the LOG_CAPS macro

2009-08-04 11:17:17 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpqdmdepay.c:
	  rtpqdm2depay: Fix debug statement.

2009-08-04 09:32:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2_calls.h:
	  v4l2: Remove some OMAP specific hacks
	  They require special build flags and are not useful in general.

2009-08-04 09:22:29 +0200  Rob Clark <rob@ti.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2sink: change where buffers get dequeued
	  It seems to cause strange occasional high latencies (almost 200ms) when dequeuing buffers from _buffer_alloc().  It is simpler and seems to work much better to dqbuf from the same thread that is queuing the next buffer.

2009-08-04 09:14:20 +0200  Rob Clark <rob@ti.com>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  v4l2: Add v4l2sink element
	  This also does the following changes:
	  (1) pull the bufferpool code out into gstv4l2bufferpool.c, and make a
	  bit more generic so it can be used both for v4l2src and v4l2sink
	  (2) move some of the device probing/configuration/caps stuff into
	  gstv4l2object.c so it does not have to be duplicated between
	  v4l2src and v4l2sink
	  Fixes bug #590280.

2009-08-04 07:07:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/Makefile.am:
	  flvmux: Enable unit test now that it passes

2009-08-03 21:21:39 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	  rtpqdm2depay,rtpsv3vdepay: Add debugging category.

2009-08-03 21:22:48 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpqdmdepay.h:
	  rtpqdm2depay: Handle gaps in incoming packets.
	  Whenever we see a gap, we flush the temporary packets (but not the adapter). If we
	  had some data temporarily stored it will be outputted (the sound will sound a bit
	  garbled... but that's how it sounds on MacOSX :)

2009-08-03 19:01:07 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpqdmdepay.c:
	  rtpqdmdepay: Fix CRC calculation and remove commented code.

2009-08-02 13:42:12 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpqdmdepay.h:
	  rtp: New QDM2 rtp depayloader.
	  Reverse-engineered by comparing:
	  * A rtp hinted file provided by DarwinStreamingServer
	  * The output procued by DSS for that same file
	  Also used various streaming sources available on the internet to fine-tune
	  the code.
	  The header/codec_data extraction methods are from FFMpeg (LGPL).

2009-08-03 21:24:44 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpsv3vdepay.c:
	  rtpsv3vdepay: Properly fill codec_data and cleanup code a bite more.

2009-08-03 19:02:17 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtpsv3vdepay.h:
	  rtpsv3vdepay: Only output buffers once we're configured.

2009-08-03 19:02:00 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpsv3vdepay.c:
	  rtpsv3vdepay: Add more encoding-name variants

2009-08-03 20:08:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/flvmux.c:
	  flvmux: Fix unit test to correctly handle request pads
	  Request pads are removed by the element instance in PAUSED->READY
	  so we need to re-request pads for every run and link them again.
	  Last fix for bug #590447.

2009-08-03 20:08:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix writing of the index for < 128 buffers
	  Partially fixes bug #590447.

2009-08-03 20:07:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix resetting of the element
	  Reset the have_video/have_audio flags and make sure to
	  properly release the request pads.
	  Partially fixes bug #590447.

2009-08-03 18:13:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't add non-utf8 chars to structures

2009-08-03 18:02:31 +0200  Luc Deschenaux <luc.deschenaux at freesurf.ch>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegdepay.h:
	  jpegdepay: use attributes for extra properties
	  Use some of the SDP attributes when they are present to specify the output
	  dimension and framerate. This allows us to receive jpeg frames larger than
	  2040 width/height.
	  Fixes #564437

2009-08-03 18:01:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/README:
	  RTP docs: update with attributes in caps

2009-08-03 17:21:44 +0200  Luc Deschenaux <luc.deschenaux at freesurf.ch>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: put all SDP attributes on caps
	  Put the SDP attributes on the caps too so that they can be used by
	  depayloaders.
	  See #564437

2009-08-03 13:32:12 +0200  Jonathan Tellier <jonathan.tellier at gmail.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: initialize the probe with the server
	  When creating a new probe, pass the server instead of the device string.
	  fixes #590401

2009-08-02 11:44:03 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: don't do things with side-effects inside g_return_val_if_fail()
	  Someone might compile this code with -DG_DISABLE_ASSERT some day.

2009-08-01 21:39:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: don't do logic within g_assert() statements
	  Otherwise that code will just be expanded to nothing when compiled
	  -DG_DISABLE_ASSERT (PS: why is mainloop_start() called in the init
	  function and not when changing state to READY?)

2009-08-01 17:07:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: send newsegment event when operating push-based and unframed
	  For some reason flac doesn't call our metadata callback when we operate
	  in push mode with unframed input, but that's where we set up the
	  newsegment event (since that's where we'd get the duration from the
	  stream info header), so we didn't send a newsegment event at all in this
	  case. Hack around this by storing a generic newsegment event for now
	  which will be used if we don't replace it with a better one that
	  includes the duration.

2009-08-01 16:48:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: small cleanups
	  Remove some callback indirections which are no longer needed because
	  there's only one decoder object type now. Also remove unused variable.

2009-08-01 15:22:49 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: use gst_adapter_copy() to avoid unnecessary buffer merges
	  gst_adapter_peek() will merge buffers as needed, which we can avoid
	  here since we're doing a memcpy anyway and then flush the copied
	  data from the adapter right away.

2009-08-01 00:00:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: repair some broken indenting

2009-08-01 12:19:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/flvmux.c:
	  checks: add basic unit test for flvmux, but disable it for now
	  Basic unit test for flvmux. Fails miserably, hence disabled for now.

2009-07-31 23:28:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/flvdemux.c:
	* tests/files/Makefile.am:
	* tests/files/pcm16sine.flv:
	  check: add basic unit test for flvdemux
	  In particular, test re-use of flvdemux in both pull and push mode
	  (see #583030).

2009-07-31 20:25:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: fix invalid write caused by using sizeof("string") as length
	  sizeof("foo") includes the string's NUL-terminator in the size returned,
	  but we're writing strings here with an explicit size at the beginning
	  and no NUL-terminator. In most cases using sizeof("foo") as length in
	  memcpy is not harmful, but it is where the string goes right at the
	  end of our buffer to write, since we don't allocate space for that
	  NUL terminator.

2009-07-27 18:44:45 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/soup/gstsouphttpsrc.c:
	  soup: Use "GET" instead of SOUP_METHOD_GET. Fixes build with libsoup-2.7.*
	  This is due to a quality API change in libsoup 2.7. SOUP_METHOD_* are now
	  integers and not strings... they could have changed the names.

2009-07-30 17:57:53 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	  jpeg: use longer macro names to not clash with some stupid windows defines
	  libjpeg headers pull some windows system inlcudes (on windows) that contain a
	  define for DEFAULT_QUALITY.

2009-07-29 14:31:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix last commit and improve readability

2009-07-24 19:04:31 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

	* gst/avi/gstavidemux.c:
	  Fixed the fix for TIME->DEFAULT conversion.
	  Fixes bug #578052 again.

2009-07-29 13:38:03 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpsv3vdepay.c:
	  rtpsv3depay: Fix width/height calculation, bring up to marginal rank.
	  Based on documentation found on http://wiki.multimedia.cx/

2009-07-29 12:13:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	  pulse: conditionally compile newer stuff
	  configured_sink/source_usec in the timing_info is only since 0.9.11 so
	  conditionally compile this information.
	  fixes #590038

2009-07-28 18:29:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: cleanups
	  Keep track of the paused state of the source and leave the read function when
	  paused.
	  don't wait for a latency update when the delay is not yet known but simply
	  return 0 instead of blocking.
	  Keep track of the corked state of the stream.
	  Fix the state changes.

2009-07-28 16:11:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: set maxlength always to -1

2009-07-28 15:53:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc; cleanups, report real latency
	  Add some more debug info
	  Avoid some type casts
	  Report the real latency to the application.

2009-07-28 16:11:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: when scanning for 0xff marker ends, ensure desired result
	  Otherwise, any non 0xff byte at end of data would be mistaken for
	  a tag byte, and in case of a frame_len 0 tag subsequently lead to an
	  infinite loop.

2009-07-28 00:30:43 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/avi/gstavimux.c:
	  avimux: adds support to wma

2009-07-28 00:07:15 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/avi/gstavimux.c:
	  avimux: adds support to wmv

2009-07-27 21:34:22 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Downgrade warning message to debug

2009-07-27 11:51:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: avoid using ivalid stream indexes
	  when we get an invalid stream index from pulse because we were just starting,
	  avoid using it for getting and setting the volume.
	  Fixes #589365

2009-07-24 19:38:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	  effectv: Don't allow caps changes for some effectv filters
	  These filters use information from previous frames to
	  generate the current frame and a caps change will make
	  the effect start from the beginning again.

2009-07-24 19:37:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	* gst/effectv/gstwarp.h:
	  warptv: Make the sine table global instead of having it in every instance

2009-07-24 10:47:44 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	  jpeg: make encoder work with libjpeg v7
	  We have to specify do_fancy_downsampling = FALSE in the encoder with did not exist before.

2009-07-24 00:42:33 +0300  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From fedaaee to 94f95e3

2009-07-23 12:06:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: Implement SEEKING query
	  Fixes bug #589423.

2009-07-22 11:16:06 +0100  Colin Guthrie <cguthrie@mandriva.org>

	* ext/pulse/pulsesink.c:
	  pulsesink: Fix a couple error messages that mentioned incorrect function names.
	  Fixes #589459.

2009-07-23 11:50:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvparse.c:
	  flvdemux: Implement SEEKING query
	  Also add some more query types to the answer of the query type function.
	  Fixes bug #589424.

2009-07-21 19:46:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: fix intermittent FLAC__STREAM_DECODER_ABORTED errors when seeking
	  When seeking in a local flac file (ie. operating pull-based), the decoder
	  would often just error out after the loop function sees a DECODER_ABORTED
	  status. This, however, is the read callback's way of telling our loop
	  function that pull_range failed and streaming should stop, in this case
	  because of the flush-start event that the seek handler pushed upstream
	  from the seeking thread. Handle this slightly better by storing the last
	  flow return from pull_range, so the loop function can evaluate it properly
	  when it encounters a DECODER_ABORTED and take the right action.
	  Fixes #578612.

2009-07-21 10:07:00 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/interleave/interleave.c:
	  interleave: fix indenting and upgrade two debugs to warnings.
	  Fix newlines in variable decls. Change two debugs to become warnings as they
	  indicate that things will not work.

2009-07-21 10:04:36 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpeg.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpeg: code cleanups for encoder
	  Remove some disabled code in encoder. Try #if 0'ed code and add comments about
	  why it is disabled. Move idct-method enum to jpeg.c and use in both encoder and
	  decoder. Add idct-method property to encoder.

2009-07-21 07:50:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Answer SEEKING queries in the original format

2009-07-21 01:12:44 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/udp/gstudpnetutils.c:
	  udputils: initialize struct content with 0.
	  Fixes some random crashes.

2009-07-20 19:09:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: set some values to their defaults
	  Set the minreq and maxlength buffer attributes to -1 to let puleseaudio select a
	  sensible value.

2009-07-20 19:04:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: don't wait for posted message
	  We can't wait for the ENTER/LEAVE messages to be be posted because the base
	  class sometimes calls the start method with the object lock, which would block
	  the message posting.
	  Instead, just assume that the message will be posted soon and continue. We'll
	  have to fix this in the base class.

2009-07-20 18:11:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: use relative seeks
	  Use relative seeks because I was told that absolute seeks don't work.

2009-07-20 16:52:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Implement SEEKING query

2009-07-20 08:07:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	  cairorender: Add support for ARGB/BGRA input
	  Note that videotestsrc outputs 100% transparent video
	  which will result in white output from cairorender.

2009-07-17 13:22:57 +0100  Elaine Xiong <Elaine.Xiong@Sun.COM>

	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2: Fix v4l2src on OpenSolaris
	  The v4l2 driver for USB webcams on OpenSolaris does not support select()
	  calls. Detect when select() fails, and skip polling the device afterward,
	  which restores the pre 0.10.14 behaviour on OpenSolaris.
	  Signed-off-by: Jan Schmidt <thaytan@noraisin.net>

2009-07-17 11:22:06 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/elements/.gitignore:
	* tests/examples/v4l2/.gitignore:
	  gitignore: Ignore some new binaries

2009-07-17 13:49:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-cairo.xml:
	* ext/cairo/gstcairorender.c:
	  cairorender: Add to the documentation

2009-07-17 13:42:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	  cairorender: Return not-negotiated if we have no caps

2009-07-17 13:41:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	* ext/cairo/gstcairorender.h:
	  cairorender: Fix caps and colorspace handling

2009-07-17 13:30:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	  cairorender: Use correct mimetypes for PDF and SVG

2009-07-17 13:24:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	  cairorender: Remove pull mode, it only adds complexity but not advantages

2009-07-16 21:55:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	  cairorender: Fix caps negotiation and cairo surface creation

2009-07-16 21:42:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	  cairorender: Correctly set srccaps

2009-07-16 21:31:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	* ext/cairo/gstcairorender.h:
	  cairorender: Move instance/class struct definitions to the header

2009-07-16 21:30:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/cairo/gstcairorender.c:
	* ext/cairo/gstcairorender.h:
	  cairorender: Add Lutz' copyright to the file header

2009-07-16 21:27:45 +0200  Lutz Mueller <lutz@topfrose.de>

	* ext/cairo/Makefile.am:
	* ext/cairo/gstcairo.c:
	* ext/cairo/gstcairorender.c:
	* ext/cairo/gstcairorender.h:
	  cairo: Add cairo-based PDF/PS/SVG encoder element
	  Fixes bug #331420.

2009-07-16 20:44:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flacenc: Optionally write a PADDING block
	  The size of the PADDING block is specified by a new
	  "padding" property.
	  Fixes bug #588483.

2009-07-16 19:35:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Only assume seekability if the server provides Content-Length
	  Previously seekability way always assumed until the first seek actually
	  failed. Now we assume that all servers are not seekable unless they provide
	  a Content-Length header. If a seek fails after that we continue to
	  assume no seekability. Fixes bug #585576.

2009-07-16 15:14:43 +0200  Arnout Vandecappelle <arnout@mind.be>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: don't try to authenticate if no username/password is set.

2009-07-16 17:10:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	  effectv: Chain up finalize to the parent class in warptv
	  Fixes a memory leak.

2009-07-16 12:55:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/pipelines/effectv.c:
	  effectv: Add unit test for all effectv elements

2009-07-16 12:17:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	  effectv: Add new effectv elements to the docs

2009-07-15 14:37:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/Makefile.am:
	* gst/effectv/gsteffectv.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstripple.h:
	  effectv: Add rippletv element
	  This produces a water ripple effect on the video input,
	  based on motion or a rain drop algorithm.
	  Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
	  Fixes bug #588695.

2009-07-12 15:42:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/Makefile.am:
	* gst/effectv/gsteffectv.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gststreak.h:
	  effectv: Add streaktv effect filter element
	  This combines the StreakTV and BaltanTV filters from the
	  effectv project.
	  Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
	  Fixes bug #588368.

2009-07-12 12:31:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstvertigo.c:
	  effectv: Fix processing on big endian architectures

2009-07-12 11:52:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/Makefile.am:
	* gst/effectv/gsteffectv.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstradioac.h:
	  effectv: Add radioactv effect filter
	  This filter adds a radiation-like motion blur effect
	  to the video stream.
	  Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
	  Fixes bug #588359.

2009-07-12 11:26:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstop.c:
	* gst/effectv/gstop.h:
	  effectv: Make the optv threshold property an uint

2009-07-12 10:39:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/Makefile.am:
	* gst/effectv/gsteffectv.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstop.h:
	  effect: Add optv effect filter from the effectv project
	  This filter binarizes input frames and combines them with various
	  optical pattern.
	  Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
	  Fixes bug #588349.

2009-07-03 05:11:26 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Emit stream-status leave message
	  Fixes #587695

2009-07-03 05:06:45 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Emit stream-status enter message
	  Emit stream-status messages for the pulse thread.
	  Don't use our own GCond for signaling but simply use the pulse mainloop
	  mechanisms for synchronisation.
	  See #587695

2009-07-14 18:15:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: debug the latency update values

2009-07-14 16:12:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	  pulsesink: add 24bit sample formats
	  Add check for pulseaudio 0.9.15 and enable 24bits samples in that case.

2009-07-13 12:23:37 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 5845b63 to fedaaee

2009-07-13 17:53:25 +0200  Marc Leeman <marc.leeman at gmail.com>

	* gst/rtp/gstrtpmpvpay.c:
	  mpvpay: Rework the timestamping
	  Rework the timestamping in the mpv payloader so that the timestamps are more
	  accurate.
	  Fixes #587680

2009-07-03 08:47:12 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* configure.ac:
	* tests/examples/Makefile.am:
	* tests/examples/v4l2/Makefile.am:
	* tests/examples/v4l2/probe.c:
	  v4l2src: add a simple test case for device probing

2009-07-03 08:38:43 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

	* configure.ac:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2object.c:
	  v4l2src: optional support for device probing with gudev
	  Enumerate v4l2 devices using gudev if available.
	  Fixes bug #583640.

2009-07-10 19:54:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Random cleanup

2009-07-10 19:54:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Send queries to the master pad by default instead of all pads

2009-07-10 19:34:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend_rgb.c:
	* gst/videomixer/videomixer.c:
	  videomixer: Add RGB, BGR, xRGB, RGBx, xBGR, BGRx support

2009-07-10 17:43:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Clean up debugging a bit

2009-07-10 17:25:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Remove some redundant checks and error out immediately if not negotiated
	  Also stop leaking the output buffer in some error cases.

2009-07-10 17:23:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_i420.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Remove the calculate_frame_size() function and use libgstvideo instead

2009-06-30 15:13:44 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/videomixer.c:
	  videomixer: Remove unused link/unlink pad methods

2009-06-30 12:43:04 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/blend_i420.c:
	  videomixer: I420 mode: Add fast path for 0.0 and 1.0 alpha
	  If the source alpha is 0.0, we take nothing.
	  If the source alpha is 1.0, we overwrite everything.

2009-06-30 12:40:02 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/blend_i420.c:
	  videomixer: I420 blending : Fix main algorithm.
	  When blending a source layer with an alpha of 'a' on top of another
	  destination layer we take the sum of:
	  * 'a' percent of the source layer
	  * (100 - 'a') percent of the destination layer (the remainder)

2009-06-30 12:39:19 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/blend_i420.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	* gst/videomixer/videomixerpad.h:
	  videomixer: Make debugging category global to all the code.

2009-06-29 19:23:41 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videomixer/videomixer.c:
	  videomixer: improve readability of debugging statements.

2009-07-08 13:38:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: do not leak timeout message

2009-07-09 07:14:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avi: Don't forward NEWSEGMENT events from upstream
	  New ones are generated later and simply forwarding them can
	  result in NEWSEGMENT events of different format going downstream.
	  Fixes bug #587983.

2009-07-08 18:19:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_i420.c:
	  videomixer: Make checker pattern lookup table constant

2009-07-08 18:17:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/videomixer.c:
	  videomixer: Add support for ARGB
	  And clean up the caps parsing.

2009-07-08 15:17:41 +0200  Benjamin Gaignard <benjamin@gaignard.net>

	* gst/udp/gstudpnetutils.c:
	  udp: Initialize pointer to NULL
	  Otherwise we're calling free() with some random
	  memory address in error cases.
	  Fixes bug #587982.

2009-07-07 16:35:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: sprinkle some more const

2009-07-07 15:57:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: perform some more (careful) data buffering
	  Once buffering has started (with an mdat atom), continue buffering
	  until moov atom is reached, which handles cases with multiple
	  mdat atoms.  Also keep adapter/offset better in sync with upstream
	  and fix some debug statements.  Fixes #587426.

2009-07-06 10:40:31 +0200  Philip J�genstedt <philipj@opera.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Replace deprecated GST_DISABLE_DEBUG with correct macro. Fixes #587826

2009-07-01 13:07:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: error out instead of dividing by 0
	  Error out if timescale is 0.

2009-07-01 09:32:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  Revert "qtdemux: Make sure we don't blacklist streams by wrongly comparing their"
	  This reverts commit 5503a59a5779b67451d8a271000181790ee76bc7.
	  Reverting this since it causes regressions with a lot of sample files
	  I have, all of which worked fine with the last -good release (#586891).

2009-06-30 15:54:47 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: comment out unused structure

2009-06-30 13:12:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: more size checks, and use g_try_new0() instead of g_new0()
	  Whenever we alloc something based on a user-supplied size, we should
	  really use g_try_new(), otherwise we can easily be made to abort by
	  passing a ridiculously large number to us for allocing. Fixes
	  problems with some fuzzed files.

2009-06-29 18:58:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: guard against bogus atom sizes and short reads
	  Check the possibly 64-bit atom size more carefully before casting it
	  to an int and passing it to gst_pad_pull_range(), otherwise we might
	  end up pulling 0 bytes, getting an empty buffer as requested and
	  dereferencing not available data whilst thinking we actually asked
	  for and got 0x1000000000000 bytes. Similar fix for push mode operation
	  where neededbytes ends up being 0 bytes, which makes us assert. Fixes
	  crash with broken or fuzzed file (NB #122378).

2009-06-29 16:52:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use 0x prefix when logging numbers in hex

2009-07-01 08:40:40 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Don't send empty string tags

2009-06-30 21:35:37 +0400  LRN <lrn1986 at gmail.com>

	* gst/udp/gstmultiudpsink.c:
	  Don't use sendmsg()-dependent code on Windows
	  Fixes #585842

2009-06-30 15:59:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/alaw.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/law/mulaw.c:
	  law: fix caps and negotiation
	  Fix the caps to include the depth (instead of width twice) in the caps of
	  audio/x-raw-int.
	  Fix negotiation to not only copy the rate/channels of the first structure.

2009-06-30 14:48:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: include "1.0=100%" in volume and change upper limit
	  Upper volume limmit was 1000. That appear unneceasrily high. It would also cause
	  sever distortion if accidentialy used. Now its 10 (~ +15db) which is also in
	  sync with volume and playbin2.

2009-06-29 15:39:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulse: some more trivial cleanups

2009-06-29 15:38:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsemixer.c:
	  pulse: trivial cleanups

2009-06-29 15:20:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: clear ringbuffer when asked to
	  Since we map the ringbuffer to the pulseaudio internal ringbuffer, flush the
	  pulseaudio buffer when we are asked to clear the ringbuffer.
	  This avoids some leftover audio after a seek.

2009-06-26 15:00:14 +0100  Jan Schmidt <thaytan@noraisin.net>

	* autogen.sh:
	  autogen.sh: Actually do the 'echo -n' -> printf change.

2009-06-26 14:40:14 +0100  Jan Schmidt <thaytan@noraisin.net>

	* autogen.sh:
	  autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
	  Check for more automake command variants. Use printf instead of 'echo -n'
	  for portability

2009-06-26 13:42:09 +0100  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Automatic update of common submodule
	  From f810030 to 5845b63

2009-06-26 13:19:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: don't process track_num/track_count tags with a 0 value
	  Number/count values of 0 mean they're not set. Don't put those in the
	  taglist.

2009-06-25 18:51:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/waveform/gstwaveformsink.c:
	  waveformsink: use 'guint8' instead of 'byte' to fix compilation with MSVC8
	  We need a cast here for pointer arithmetic to work correctly, but some
	  MSVC versions don't seem to like 'byte', so use guint8 here. Hopefully
	  fixes #585361.

2009-06-25 19:39:37 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: set structs to zero before using them in ioctls
	  This fixes valgrind warnings.

2009-06-25 13:23:40 +0200  Julien Moutte <julien@fluendo.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Make sure we don't blacklist streams by wrongly comparing their duration with entire clip duration.

2009-06-25 13:18:14 +0200  Krzysztof Błaszkowski <kb at sysmikro.com.pl>

	* gst/rtsp/gstrtpdec.c:
	  rtpdec: fix some buffer leaks

2009-06-25 08:11:09 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvparse.c:
	  flvparse: Add missing break in switch/case.

2009-06-25 08:10:38 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Remove unused variable, hint branch likeliness, add comments.

2009-06-25 08:09:57 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Removed unused variable

2009-06-25 07:41:07 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Remove dead assignments and unused variables.
	  Also add branch likeliness macros.

2009-06-25 07:40:26 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix uninitialized variables. Fixes build on macosx

2009-06-24 17:43:25 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: free memory in finalize
	  finalize is called only once. no need to clear pointers there. dispose is for
	  unreffing.

2009-06-24 15:14:14 +0100  Jan Schmidt <jan.schmidt@sun.com>

	* common:
	  Automatic update of common submodule
	  From 6ab11d1 to f810030

2009-06-08 14:46:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: short-circuit gst_avi_demux_src_convert() when parsing the index
	  Don't call gst_avi_demux_src_convert() for each single index entry. Not
	  only do we already have the pointer to the stream context, we also know
	  the formats we want to convert from and to already, so we may just as
	  well use optimised conversion routines that bypass some of the checks
	  and lookups made in gst_avi_demux_src_convert().

2009-06-17 16:39:36 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Another round of G_*LIKELY micro-optimisations.

2009-06-17 16:20:25 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Take last sample duration for dummy segment calculation.
	  This fixes the cases where files without EDL wouldn't output their
	  last buffer.

2009-06-24 12:36:31 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Sprinkle branch likeliness macros over the code.

2009-06-23 16:54:32 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	  raw1394: sprinkle branch likeliness macros accross the code.

2009-06-14 10:36:17 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Add GST_MEMDUMP statements for unknown atoms.
	  This is to help developers track down and implement unhandled atoms faster.

2009-06-23 17:51:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Remove the interlaced field from the output caps if deinterlacing is enabled

2009-06-23 17:48:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/greedyh.c:
	  deinterlace: Copy the correct line from correct place in the history

2009-06-23 16:35:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: use same protocols after redirect
	  After a redirect we want to use the same protocols that we were using for the
	  current url.

2009-06-23 15:35:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: don't leak cover art

2009-06-23 14:10:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstudpnetutils.c:
	  udp: fix compiler warning about EAI_ADDRFAMILY getting redefined in some cases
	  Include the header from where we include all the system headers with the
	  socket stuff before we try to define EAI_ADDRFAMILY ourselves, otherwise
	  we define it ourselves and then get a compiler warning if a system header
	  defines it as well without guarding against it being defined already.

2009-06-23 14:39:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-ids.h:
	  matroska: and the new headers too

2009-06-23 14:32:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroske: fix compiler error
	  change gpointer to guint8 * for codec_state and codec_priv as some
	  functions operate on those types and it avoids breaking strict-aliasing
	  rules.

2009-06-23 12:42:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: avoid leaking buffers
	  Don't leak buffers when resyncing to a keyframe.
	  Avoid leaking buffers when exiting the loop on error conditions.
	  Add some more debug info.
	  Fixes #585911

2009-06-22 15:56:58 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: open/close the device in READY
	  This allows to query the device in READY. Before one need to switch it to PAUSED
	  and that also starts streaming.

2009-06-20 15:41:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_dump.c:
	  qtdemux: use GST_MEMDUMP

2009-06-19 00:16:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/apetag/Makefile.am:
	* gst/apetag/gstapedemux.c:
	  apedemux: add container-format tag
	  Use pbutils here because the string is translated.

2009-06-19 00:15:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/Makefile.am:
	* gst/id3demux/gstid3demux.c:
	  id3demux: add container-format tag
	  Using pbutils here because the string is translated.

2009-06-18 23:51:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	  dvdemux: post container-format tag
	  Also merge the two almost identical _add_*_pad() functions into one.

2009-06-18 23:43:49 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	  dvdemux: don't screw up first audio buffer
	  Query the audio format, esp. dvdemux->num_channels, before we use that
	  variable to allocate the initial buffer. That way we don't accidentally
	  push a zero-sized buffer as first audio buffer.

2009-06-18 23:38:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: post container-format tag

2009-06-18 23:37:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: post container-format tags

2009-06-18 23:36:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: post container-format tag

2009-06-18 23:35:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: post container-format tags

2009-06-21 17:13:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioamplify.c:
	  audioamplify: Fix integer overflows on 32 bit architectures

2009-06-21 09:50:54 +0200  Kipp Cannon <kcannon@ligo.caltech.edu>

	* gst/audiofx/audioamplify.c:
	  audioamplify: Don't declare a loop index static
	  The previous patch to add support for additional sample formats possibly
	  introduced a reentrancy bug:  a variable used for a loop index was declared
	  static.  This patch fixes that, and also adds a "/* *INDENT-ON* */" annotation
	  following the macro block.  (I don't know what the annotation is for, but the
	  adder, where I copied this from, has it).

2009-06-19 22:37:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioamplify.c:
	  audioamplify: Fix off-by-one in wrap-positive mode

2009-06-19 22:20:45 +0200  Kipp Cannon <kcannon@ligo.caltech.edu>

	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audioamplify.h:
	  audioamplify: Add noclip method and support for more formats
	  Fixes bug #585828 and #585831.

2009-06-19 21:46:41 +0200  Koop Mast <kwm@freebsd.org>

	* gst/udp/gstudpnetutils.h:
	  udp: Fix build on FreeBSD
	  Fixes bug #586397.

2009-06-19 18:12:27 +0200  Ognyan Tonchev <ognyan.tonchev at axis.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: add unit tests for buffer-list payloaders
	  See #585559

2009-06-19 18:00:35 +0200  Ognyan Tonchev <ognyan.tonchev at axis.com>

	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	  rtpmp4vpay: add support for buffer-list
	  See #585559

2009-06-19 17:57:12 +0200  Ognyan Tonchev <ognyan.tonchev at axis.com>

	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpjpegpay.h:
	  rtpjpegpay: add support for buffer-lists
	  See #585559

2009-06-19 17:53:32 +0200  Ognyan Tonchev <ognyan.tonchev at axis.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: add support for buffer-lists
	  See #585559

2009-06-18 11:54:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpnetutils.c:
	  udputils: don't free invalid memory
	  As spotted by benjiG in IRC.
	  don't free invalid memory when getaddrinfo failed.

2009-06-17 17:48:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulseink: don't leak device_description
	  don't leak the device_description.
	  some cleanups.

2009-06-19 14:44:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update .po files for sunaudiomixer string changes

2009-06-18 16:58:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: streaming; adjust sizes to cater for padding in chunks

2009-06-17 11:54:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: streaming mode; handle data chunks grouped in rec lists.
	  Fixes #567983.

2009-06-10 12:36:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: map some tags to COMPOSER rather than ARTIST

2009-06-10 12:34:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix some 3GP tag extraction (keywords, genre, location)

2009-06-09 15:36:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: extract pixel-aspect-ratio information

2009-06-17 07:14:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix leaking of the Matroska TITLE element

2009-06-16 20:38:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstaging.h:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstdice.h:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstedge.h:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstquark.h:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstrev.h:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstshagadelic.h:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstvertigo.h:
	* gst/effectv/gstwarp.c:
	* gst/effectv/gstwarp.h:
	  effectv: Add basic documentation for the effectv elements

2009-06-16 20:16:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gsteffectv.h:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstshagadelic.c:
	  effectv: Define the fast PRNG function at a central place

2009-06-16 20:13:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/Makefile.am:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstaging.h:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstdice.h:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstedge.h:
	* gst/effectv/gsteffectv.c:
	* gst/effectv/gsteffectv.h:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstquark.h:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstrev.h:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstshagadelic.h:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstvertigo.h:
	* gst/effectv/gstwarp.c:
	* gst/effectv/gstwarp.h:
	  effectv: Move type definitions into separate headers
	  This is needed for the docs later.

2009-06-16 19:41:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	  effectv: Remove get_unit_size implementations
	  The default on from GstVideoFilter handles this already.

2009-06-16 14:54:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: bump core/base requirements to git
	  Need git core for basesink bufferlist additions; -base requirement
	  bumped gratuitously.

2009-06-16 15:25:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/udpsink.c:
	  tests: add some debug, send newsegment

2009-06-16 15:06:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: add debug line for the socket

2009-06-16 15:06:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/pipelines/flacdec.c:
	  tests: turn g_print into debug

2009-06-16 15:04:15 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/udp/gstmultiudpsink.c:
	* tests/check/Makefile.am:
	* tests/check/elements/udpsink.c:
	  multiudpsink: add support for buffer lists
	  Add support for BufferList and add a unit test.
	  Fixes #585842

2009-06-16 00:02:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: reset session state when stopping
	  Increases the chances that the element is actually reusable.

2009-06-15 23:49:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: log response and request headers and fix some broken indenting

2009-06-15 22:40:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  mp4gdepay: guess constantDuration better
	  Do a better job at guessing the constantDuration parameter when it is not
	  present in the caps.
	  Fixes #585205

2009-06-15 21:09:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstwarp.c:
	  warptv: Clean up warptv element and fix some minor bugs and leaks

2009-06-15 20:53:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstvertigo.c:
	  vertigotv: Clean up vertigotv element and fix some minor bugs and leaks

2009-06-15 20:38:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstdice.c:
	  dicetv: Use guint8 instead of char (which can be signed or unsigned)

2009-06-15 20:36:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstshagadelic.c:
	  shagadelictv: Use guint8/gint8 instead of char (which can be signed or unsigned)

2009-06-15 20:31:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstshagadelic.c:
	  shagadelictv: Clean up element and free all memory in finalize

2009-06-15 20:21:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstrev.c:
	  revtv: Clean up revtv element

2009-06-15 20:07:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstquark.c:
	  quarktv: Simplify some code

2009-06-15 20:07:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstquark.c:
	  quarktv: Use the input data if a NULL buffer is chosen instead of the value 0

2009-06-15 20:00:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstquark.c:
	  quarktv: Fix setting the planes property of quarktv
	  Setting it to a value<16 would cause crashes before because
	  current_plane was set to the old number of planes-1. Also
	  fix calculations for non-2^n planes values.

2009-06-15 17:50:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstquark.c:
	  quarktv: Clean up the quarktv element

2009-06-15 17:39:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gsteffectv.c:
	  effectv: Make elements list constant

2009-06-15 17:37:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstedge.c:
	  edgetv: Clean up edgetv element and fix memory leak

2009-06-15 17:21:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstdice.c:
	  dicetv: Clean up dicetv element and fix some smaller issues
	  This fixes a memory leak (the dice map) and a crash when
	  setting the square-bits property before caps are set.

2009-06-15 17:20:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/Makefile.am:
	* gst/effectv/gstaging.c:
	  agingtv: Actually use GstController for syncing the properties to timestamps

2009-06-15 17:03:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstaging.c:
	  agingtv: Export some more agingtv properties via GObject properties

2009-06-15 15:06:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstaging.c:
	  agingtv: General cleanup and updating of copyright
	  Also make the scratch-lines property exported via a GObject
	  property and initialize/reset the internal state correctly.

2009-06-15 15:05:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/effectv/gstaging.c:
	  agingtv: Store and update state inside the instance struct
	  This makes the coloraging effect and pits effect visible.

2009-06-15 15:51:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: ref custom ring buffer class and type in class_init
	  Hack around thread-safety issues in GObject and our racy _get_type()
	  functions (we could easily fix the _get_type() functions, but we still
	  need to hack around the GObject class races until we require a newer
	  GLib version, I think).

2009-06-14 19:19:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/dv/demo-play.c:
	* tests/old/examples/Makefile.am:
	* tests/old/examples/level/Makefile.am:
	* tests/old/examples/level/README:
	* tests/old/examples/level/demo.c:
	* tests/old/examples/level/plot.c:
	* tests/old/examples/switch/.gitignore:
	* tests/old/examples/switch/Makefile.am:
	* tests/old/examples/switch/switcher.c:
	  Remove a few old example apps from the 0.8 days
	  Some have been replaced by newer ones, others are demoing elements that
	  don't exist any longer (not in -good anyway), and others have not been
	  touched in many years and it seem pointless to keep them around.
	  Removing these files makes sure we don't have any code in our repository
	  that uses Gtk+ symbols which are to be removed for GNOME3, and as such
	  will make some script that greps for this kind of stuff give us a clean
	  bill of code health. Fixes #585757.

2009-06-13 21:02:45 -0400  Olivier Crête <tester@tester.ca>

	* common:
	* gst/rtp/gstrtpsirenpay.c:
	  rtpsirenpay: Remove deprecated symbol
	  Patch by: Luis Menina

2009-06-13 10:43:55 +0200  Marvin Schmidt <marvin_schmidt@gmx.net>

	* tests/check/Makefile.am:
	  tests: Don't run the flacdec test if the plugin isn't built. Fixes #585630

2009-06-12 16:06:28 +0200  Patrick Radizi <patrick.radizi at axis.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add RTP blocksize functionality
	  Add property to make the client suggest a blocksize to the server.
	  Fixes #585549

2009-06-11 22:30:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/README:
	  rtp: update README, fix some typos, mention gstrtpbin

2009-06-11 19:10:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: handle border cases in resampler

2009-06-11 13:32:22 +0100  Jan Schmidt <thaytan@noraisin.net>

	* common:
	* docs/Makefile.am:
	* docs/plugins/Makefile.am:
	* docs/upload.mak:
	  docs: Bump common. Use upload-doc.mak instead of upload.mak
	  Remove the local copy of upload.mak in favour of using the shared
	  upload-doc.make in common/

2009-06-11 11:39:25 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/goom/goom_config_param.h:
	* gst/videomixer/videomixer.c:
	  docs: Quieten a couple more docs warnings

2009-06-11 11:27:26 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/matroska/lzo.c:
	  docs: Remove gtk-doc comment marker
	  These comment blocks aren't gtk-doc comments and cause annoying noise in
	  the docs build.

2009-06-11 10:05:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Implement upstream negotation

2009-06-10 21:47:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Improve debugging and clean up some code

2009-06-10 14:55:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Clip buffers to the current segment if possible

2009-06-10 14:45:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Clean up includes and clean up order of instance struct fields

2009-06-10 16:09:56 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph263pay.h:
	  rtph263pay: Default to doing A, B and C modes, not only A

2009-06-10 09:56:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix QoS calculations
	  The diff is a signed integer, not an unsigned one of course.
	  In modes other than GST_DEINTERLACE_ALL every frame has twice the
	  duration of the field duration.

2009-06-09 14:13:31 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpsirenpay.c:
	  rtpsirenpay: Put the bitrate in the RTP caps
	  The MS code seems to require the bitrate to interoperate and
	  draft-ietf-avt-rtp-g7221-00 also has it.

2009-06-09 19:55:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: Implement basic QoS
	  This change is based on Tim's QoS implementation
	  for jpegdec.

2009-06-09 19:29:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Directly proxy events/queries to the peer pads
	  This removes some overhead introduced by the default handlers
	  that need to iterate over the other pads.

2009-06-09 10:38:52 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: debug_memdump() unknown tags. Refactor junk parsing code.
	  This makes life slightly easier when debugging avi files.

2009-06-08 08:21:43 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/Makefile.am:
	  rtp: Don't forget to dist the headers for the CELT (de)payloaders.

2009-06-07 20:54:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  Revert "Revert "qtdemux: fill timestamp table completely""
	  This reverts commit 9f022c8a8503c2ce0fa617fdb50e41706dd412f5.
	  Sorry, I was thinking about the wrong module.

2009-06-07 20:49:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  Revert "qtdemux: fill timestamp table completely"
	  This reverts commit 790b050fc5302cae89cddcd23b258093967d05a9.
	  I forgot we were frozen.

2009-06-07 20:46:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fill timestamp table completely
	  When there are less timestamps that there are samples, fill up the sample table
	  with the last know timestamp. This situation can happen when the last sample
	  does not decode and doesn't need a timestamp. We however calculate the total
	  track length using the last sample timestamp so we need to have something
	  sensible in there.
	  Fixes #585056

2009-06-07 13:37:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: handle LIST INFO of 0 size
	  Handle LIST INFO chunks of 0 size instead of causing errors.
	  Fixes #584981

2009-06-07 13:24:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  Revert "wavparse: Remove dead assignments, move variable to where it's needed."
	  Reverts commit 44256a78f8dd79a91f3bb2ab7c3aa623c097bb8a and use the result in
	  error reporting so that we can see what's going on.

2009-06-05 18:55:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltdepay.h:
	  celtdepay: add CELT depayloader

2009-06-05 15:30:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpceltpay.h:
	  rtpceltpay: add CELT RTP payloader

2009-06-05 16:54:48 +0100  Jan Schmidt <jan.schmidt@sun.com>

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixeroptions.c:
	* sys/sunaudio/gstsunaudiomixertrack.c:
	  sunaudio: Fix switch setting on some devices. Add debug. Fix a FIXME.
	  Fix the setting of toggle switches on some broken audio drivers which
	  report that no audio ports are settable by ignoring the mod_port field
	  there.
	  Add some debug statements.
	  Fix a FIXME now that Good relies on a new enough gst-plugins-base.

2009-06-04 12:27:19 +0100  Jan Schmidt <jan.schmidt@sun.com>

	* sys/sunaudio/Makefile.am:
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixeroptions.c:
	* sys/sunaudio/gstsunaudiomixeroptions.h:
	* sys/sunaudio/gstsunaudiomixertrack.c:
	* sys/sunaudio/gstsunaudiomixertrack.h:
	  sunaudio: Support new flags for options and actions
	  Use new audio mixer flags added in Base 0.10.23 to expose flags and options
	  on the SunAudio devices.
	  Fixes: #583593
	  Patch By: Brian Cameron <brian.cameron@sun.com>
	  Patch By: Garrett D'Amore <garrett.damore@sun.com>

2009-05-15 11:50:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	  deinterlace: First try to handle DVD still frames correctly
	  This helps a bit with bug #582740 but still doesn't make it work.

2009-06-04 17:37:03 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: only notify if all checks passed
	  Replace goto done: with return, as those are checks when we don't want to flag a
	  pending notify.

2009-06-04 15:19:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: set the right state on rtpbin
	  We need to set the state of gstrtpbin to the same state as our source elements.
	  This fixes fallback to TCP again.

2009-06-03 18:23:53 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: check pointer before accessing
	  Move existing check a few lines up, so that we check before accessing fields.

2009-06-03 18:21:12 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: rename gst_pulse_sink_get_time to gst_pulsesink_get_time
	  Rename internal method for consistency.

2009-06-03 18:19:22 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: use values from pa_stream_get_buffer_attr()
	  We were putting the requested values back into ringbuffer spec, instead of
	  using the queried values.

2009-06-02 19:32:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  vrawpay: trim output buffers
	  Remove the leftover unused bytes in the output buffer.
	  Fixes #584613

2009-06-02 19:30:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  vrawdepay: fix parsing of sampling field
	  commit a12d9a80f225be97b3674b1a0506ac66544dbf49 broke the parsing of the
	  sampling.

2009-05-27 17:06:34 +0100  Jan Schmidt <thaytan@noraisin.net>

	* ext/libpng/gstpngdec.c:
	  pngdec: Avoid possible overflow in calculations
	  A malformed (or simply huge) PNG file can lead to integer overflow in
	  calculating the size of the output buffer, leading to crashes or buffer
	  overflows later. Fixes SA35205 security advisory.

2009-06-02 00:48:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: some more logging - dump header packets
	  Also, the final fixing up of the headers is expected and not something
	  we should warn about.

2009-06-02 00:37:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: never ever pass values >36bits to _set_total_samples_estimate()
	  Let's be paranoid and make sure we never pass a number that takes up
	  more than 36 bits to _set_total_samples_estimate(), since libFLAC
	  expects all the other bits to be zero, and if this is not the case
	  neighbouring fields in the global stream info header may get messed
	  up inadvertently, so that flac -d refuses to decode the stream.
	  See #584455.

2009-06-01 22:33:02 +0200  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* ext/flac/gstflacenc.c:
	  Address bad FLAC sample length encoding of #5844455
	  Commit df707c666433a78d3878af6f055698d5756226c4
	  introduced an obvious bug in the sample length calculation,
	  using the wrong macro for conversion.

2009-06-01 11:58:21 -0700  Brian Cameron <brian.cameron@sun.com>

	* gst/deinterlace/tvtime/mmx.h:
	  deinterlace: Fix spurious colons in asm code
	  Fixes #584174.
	  Signed-off-by: David Schleef <ds@schleef.org>

2009-06-01 00:40:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: skip JUNK chunks in data section in streaming mode
	  Skip JUNK tags in streaming mode as well instead of EOSing
	  prematurely. Fixes #564100.

2009-05-28 14:01:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_i420.c:
	* gst/videomixer/videomixer.c:
	  videomixer: Don't use // comments

2009-05-28 13:56:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_bgra.c:
	  videomixer: Fix background blitting when a color mode is selected with BGRA

2009-05-28 13:54:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_i420.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Some cleanup and fix the calculation of the frame size in bytes

2009-05-28 13:35:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_i420.c:
	  videomixer: Fix I420 blending to actually do something
	  For this we a) implement the checkers filling and b)
	  actually blend the src/dest by using the src alpha value
	  from the pad.

2009-05-28 13:14:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_bgra.c:
	  videomixer: Fix ARGB blending to actually work

2009-05-28 13:04:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend_bgra.c:
	  videomixer: Blend BGRA ourselves instead of using Cairo

2009-05-28 12:55:16 +0200  Alex Ugarte <alexugarte@gmail.com>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_i420.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Add support for blending BGRA and AYUV
	  Fixes bug #577017.

2009-05-28 12:39:46 +0200  Ghislain 'Aus' Lacroix <aus@songbirdnest.com>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Use floating point arithmetic internally for the int16 mode
	  By using int32 arithmetic we will introduce distortions as the
	  IIR filter is very sensitive to rounding errors. Fixes bug #580214.

2009-05-28 10:55:16 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

	* gst-plugins-good.spec.in:
	  Update spec file with latest plugins

2009-05-26 17:19:08 +0100  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Automatic update of common submodule
	  From 888e0a2 to c572721

2009-05-26 16:20:35 +0300  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2: cleanup and commenting
	  Remove newlines inserted by gst-indent once. Remove unused var from instance
	  struct. Add comments. Add another #define for default property value.

2009-05-06 12:43:35 +0300  Stefan Kost <ensonic@users.sf.net>

	* tests/check/Makefile.am:
	  makefile: idea about makeing more sources/sinks testable again

2009-05-25 16:33:35 +0200  John Keeping <john.keeping at lineone.net>

	* ext/libpng/gstpngdec.c:
	  pngdec: match g_malloc() with g_free()
	  Matching g_malloc() with a g_free() is important when a custom allocator is
	  installed.
	  Fixes #583803

2009-05-12 18:39:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	  rtpmp4vpay: don't look for headers in some cases
	  In some streams (starting with 00000100) don't look for the headers but push
	  data as it is.
	  Fixes #582153

2009-05-13 11:50:22 +0200  Patrick Radizi <patrick.radizi at axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix memory leak of messages
	  Free messages correctly.
	  Fixes #577318

2009-05-24 19:32:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: make fakesrc silent
	  Make the fakesrc that is responsible for sending dummy packets silent.

2009-05-24 16:33:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't send teardown before setup
	  Don't send a TEARDOWN request when we did not manage to successfully setup a
	  stream.

2009-05-14 14:46:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Populate a GstIndex that is set on matroskademux

2009-05-14 10:35:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Get the max duration from upstream if there's no duration tag

2009-05-14 10:29:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Write an index table to the end of the file

2009-05-22 01:12:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* autogen.sh:
	* configure.ac:
	  autotools: move the -Wno-portability from autogen.sh to configure.ac
	  If we're lucky it'll get used on automatic rebuilds as well that way.

2009-05-22 01:10:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	* configure.ac:
	* m4/gst-fionread.m4:
	  m4: fix 'suspicious cache id' warnings
	  and update common to pull in a similar fix. Also check in configure
	  whether the compiler supports do while macros (GLib wants this
	  defined and it is needed to avoid warnings with some c++ compilers
	  apparently).

2009-05-22 01:39:33 +0300  Zeeshan Ali (Khattak) <zeeshanak@gnome.org>

	* configure.ac:
	  souphttpsrc: Bump-up libsoup-2.24 dep to >= 2.26
	  The helper function soup_message_headers_get_content_type that we now use
	  was added in 2.26.

2009-05-20 17:57:59 +0300  Zeeshan Ali (Khattak) <zeeshanak@gnome.org>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Set caps for audio/L16 content-type
	  When "Content-Type" header is "audio/L16", we need to set the caps on the
	  outgoing buffers so that downstream elements can have means to detect the
	  stream type and handle it appropriately. Tested with HTTP stream provided
	  by pulse-audio's http module (git master).

2009-05-20 15:06:25 +0300  Zeeshan Ali (Khattak) <zeeshanak@gnome.org>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Rename icy_caps to src_caps

2009-05-21 23:39:13 +0200  Philippe Normand <philippe at fluendo.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: bump max size to 65535x65535
	  Remove artificial jpeg image limits.
	  Fixes #583048.

2009-05-21 21:36:02 +0100  Jan Schmidt <thaytan@noraisin.net>

	* win32/common/config.h:
	  win32: Update the win32 config.h

2009-05-19 15:12:09 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Recognise PGS subpicture streams - the bluray format.
	  Recognise and apply appropriate caps to PGS (Presentation Graphic Stream)
	  subpicture streams.

2009-05-15 10:42:19 +0100  Jan Schmidt <thaytan@noraisin.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: Convert an erroneous assertion
	  Occasionally, we get a change callback for an old stream, triggering
	  the assertion unnecessarily. Just ignore such callbacks.

2009-05-20 16:14:40 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulse: Print a warning on under/overflows

2009-05-20 18:45:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: parse in24 boxes to get endianness
	  in24 samples are normally big-endian but an enda box can change this to
	  little-endian. Recurse into the in24 box and find the enda box so that we get
	  the endianness right.
	  Fixes #582515

2009-05-20 14:14:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: add proper padtemplate

2009-05-20 14:02:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: add more mime types
	  Add mime-type for Panasonic g726 and add more required caps properties for other
	  G726 mime-types.
	  Make mime-types case insensitive.
	  See #582169

2009-05-20 13:47:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: add flow aggregation

2009-05-20 13:29:02 +0200  Arnout Vandecappelle <arnout@mind.be>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: allow content to be empty.
	  gst_adapter_take_buffer doesn't allow buffer to be empty.
	  Simply skip any part where the content is empty.  Don't
	  create a pad for it either.
	  See #582169

2009-05-18 22:19:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpchannels.h:
	  rtp: fix channel positions for mono

2009-05-21 21:02:11 +0100  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	  Back to hacking -> 0.10.15.1

=== release 0.10.15 ===

2009-05-20 22:34:18 +0100  Jan Schmidt <thaytan@noraisin.net>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.15

2009-05-20 22:03:21 +0100  Jan Schmidt <thaytan@noraisin.net>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2009-05-16 02:59:14 +0100  Jan Schmidt <thaytan@noraisin.net>

	* ChangeLog:
	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	* win32/common/config.h:
	  0.10.14.3 pre-release

2009-05-16 02:37:06 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/pipelines/flacdec.c:
	  check: Don't change directory in the test
	  Changing directory invalidates the paths the registry has picked
	  up for our plugins, because the test environment specifies relative
	  paths. Fixing that is a separate problem, in the meantime, build a
	  path to the test files instead of changing directory. Fixes the
	  distcheck.

2009-05-16 01:53:46 +0100  Jan Schmidt <thaytan@noraisin.net>

	* win32/MANIFEST:
	  win32: Remove directdraw project files from the win32 manifest

2009-05-16 01:21:34 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/elements/rganalysis.c:
	  check: Remove assertion that breaks check again git master
	  Remove the assertion that the sender of the tags message is the
	  element until we decide whether that's going to be true or not.

2009-05-16 01:11:33 +0100  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-directdraw.xml:
	* sys/Makefile.am:
	* sys/directdraw/Makefile.am:
	* sys/directdraw/gstdirectdrawplugin.c:
	* sys/directdraw/gstdirectdrawsink.c:
	* sys/directdraw/gstdirectdrawsink.h:
	* win32/vs6/libgstdirectdraw.dsp:
	* win32/vs7/libgstdirectdraw.vcproj:
	* win32/vs8/libgstdirectdraw.vcproj:
	  Moved 'directdraw' from -good to -bad

2009-05-16 00:18:34 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/pipelines/.gitignore:
	  ignores: Ignore the flacdec check binary

2009-05-16 00:17:57 +0100  Jan Schmidt <thaytan@noraisin.net>

	* docs/plugins/inspect/plugin-avi.xml:
	  docs: Update inspection details for the avi plugin

2009-05-16 00:00:07 +0100  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/y4menc.c:
	  Moved 'y4menc' from -bad to -good

2009-05-13 17:55:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] y4menc: change my email
	  change my email to something more current
	  See #580783

2009-05-13 17:54:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] y4menc: don't strip timestamps
	  Fixes #582483

2008-11-04 12:42:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD] Don't install static libs for plugins. Fixes #550851 for -bad.
	  Original commit message from CVS:
	  * ext/alsaspdif/Makefile.am:
	  * ext/amrwb/Makefile.am:
	  * ext/apexsink/Makefile.am:
	  * ext/arts/Makefile.am:
	  * ext/artsd/Makefile.am:
	  * ext/audiofile/Makefile.am:
	  * ext/audioresample/Makefile.am:
	  * ext/bz2/Makefile.am:
	  * ext/cdaudio/Makefile.am:
	  * ext/celt/Makefile.am:
	  * ext/dc1394/Makefile.am:
	  * ext/dirac/Makefile.am:
	  * ext/directfb/Makefile.am:
	  * ext/divx/Makefile.am:
	  * ext/dts/Makefile.am:
	  * ext/faac/Makefile.am:
	  * ext/faad/Makefile.am:
	  * ext/gsm/Makefile.am:
	  * ext/hermes/Makefile.am:
	  * ext/ivorbis/Makefile.am:
	  * ext/jack/Makefile.am:
	  * ext/jp2k/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/lcs/Makefile.am:
	  * ext/libfame/Makefile.am:
	  * ext/libmms/Makefile.am:
	  * ext/metadata/Makefile.am:
	  * ext/mpeg2enc/Makefile.am:
	  * ext/mplex/Makefile.am:
	  * ext/musepack/Makefile.am:
	  * ext/musicbrainz/Makefile.am:
	  * ext/mythtv/Makefile.am:
	  * ext/nas/Makefile.am:
	  * ext/neon/Makefile.am:
	  * ext/ofa/Makefile.am:
	  * ext/polyp/Makefile.am:
	  * ext/resindvd/Makefile.am:
	  * ext/sdl/Makefile.am:
	  * ext/shout/Makefile.am:
	  * ext/snapshot/Makefile.am:
	  * ext/sndfile/Makefile.am:
	  * ext/soundtouch/Makefile.am:
	  * ext/spc/Makefile.am:
	  * ext/swfdec/Makefile.am:
	  * ext/tarkin/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/timidity/Makefile.am:
	  * ext/twolame/Makefile.am:
	  * ext/x264/Makefile.am:
	  * ext/xine/Makefile.am:
	  * ext/xvid/Makefile.am:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/dshow/Makefile.am:
	  * gst/aiffparse/Makefile.am:
	  * gst/app/Makefile.am:
	  * gst/audiobuffer/Makefile.am:
	  * gst/bayer/Makefile.am:
	  * gst/cdxaparse/Makefile.am:
	  * gst/chart/Makefile.am:
	  * gst/colorspace/Makefile.am:
	  * gst/dccp/Makefile.am:
	  * gst/deinterlace/Makefile.am:
	  * gst/deinterlace2/Makefile.am:
	  * gst/dvdspu/Makefile.am:
	  * gst/festival/Makefile.am:
	  * gst/filter/Makefile.am:
	  * gst/flacparse/Makefile.am:
	  * gst/flv/Makefile.am:
	  * gst/games/Makefile.am:
	  * gst/h264parse/Makefile.am:
	  * gst/librfb/Makefile.am:
	  * gst/mixmatrix/Makefile.am:
	  * gst/modplug/Makefile.am:
	  * gst/mpeg1sys/Makefile.am:
	  * gst/mpeg4videoparse/Makefile.am:
	  * gst/mpegdemux/Makefile.am:
	  * gst/mpegtsmux/Makefile.am:
	  * gst/mpegvideoparse/Makefile.am:
	  * gst/mve/Makefile.am:
	  * gst/nsf/Makefile.am:
	  * gst/nuvdemux/Makefile.am:
	  * gst/overlay/Makefile.am:
	  * gst/passthrough/Makefile.am:
	  * gst/pcapparse/Makefile.am:
	  * gst/playondemand/Makefile.am:
	  * gst/rawparse/Makefile.am:
	  * gst/real/Makefile.am:
	  * gst/rtjpeg/Makefile.am:
	  * gst/rtpmanager/Makefile.am:
	  * gst/scaletempo/Makefile.am:
	  * gst/sdp/Makefile.am:
	  * gst/selector/Makefile.am:
	  * gst/smooth/Makefile.am:
	  * gst/smoothwave/Makefile.am:
	  * gst/speed/Makefile.am:
	  * gst/speexresample/Makefile.am:
	  * gst/stereo/Makefile.am:
	  * gst/subenc/Makefile.am:
	  * gst/tta/Makefile.am:
	  * gst/vbidec/Makefile.am:
	  * gst/videodrop/Makefile.am:
	  * gst/videosignal/Makefile.am:
	  * gst/virtualdub/Makefile.am:
	  * gst/vmnc/Makefile.am:
	  * gst/y4m/Makefile.am:
	  * sys/acmenc/Makefile.am:
	  * sys/cdrom/Makefile.am:
	  * sys/dshowdecwrapper/Makefile.am:
	  * sys/dshowsrcwrapper/Makefile.am:
	  * sys/dvb/Makefile.am:
	  * sys/dxr3/Makefile.am:
	  * sys/fbdev/Makefile.am:
	  * sys/oss4/Makefile.am:
	  * sys/qcam/Makefile.am:
	  * sys/qtwrapper/Makefile.am:
	  * sys/vcd/Makefile.am:
	  * sys/wininet/Makefile.am:
	  * win32/common/config.h:
	  Don't install static libs for plugins. Fixes #550851 for -bad.

2008-06-26 15:52:40 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  [MOVED FROM BAD] Add documentation for YUV4MPEG2 encoder element.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * gst/y4m/gsty4mencode.c:
	  Add documentation for YUV4MPEG2 encoder element.

2007-04-24 15:49:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  [MOVED FROM BAD] Plug some leaks; try to make build bot happy again.
	  Original commit message from CVS:
	  * gst/y4m/gsty4mencode.c: (gst_y4m_encode_init),
	  (gst_y4m_encode_setcaps):
	  * tests/check/elements/y4menc.c: (GST_START_TEST):
	  Plug some leaks; try to make build bot happy again.

2006-11-13 18:55:57 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  [MOVED FROM BAD] configure.ac: Enable cdaudio and y4m.
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw at skynet be>
	  * configure.ac:
	  Enable cdaudio and y4m.
	  * gst/y4m/Makefile.am:
	  * gst/y4m/gsty4mencode.c: (gst_y4m_encode_base_init),
	  (gst_y4m_encode_class_init), (gst_y4m_encode_init),
	  (gst_y4m_encode_reset), (gst_y4m_encode_setcaps),
	  (gst_y4m_encode_get_stream_header),
	  (gst_y4m_encode_get_frame_header), (gst_y4m_encode_chain),
	  (gst_y4m_encode_set_property), (gst_y4m_encode_get_property),
	  (gst_y4m_encode_change_state), (plugin_init):
	  * gst/y4m/gsty4mencode.h:
	  Port of y4mencode to 0.10.

2006-04-25 21:56:38 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD] Define GstElementDetails as const and also static (when defined as global)
	  Original commit message from CVS:
	  * ext/amrwb/gstamrwbdec.c:
	  * ext/amrwb/gstamrwbenc.c:
	  * ext/amrwb/gstamrwbparse.c:
	  * ext/arts/gst_arts.c:
	  * ext/artsd/gstartsdsink.c:
	  * ext/audiofile/gstafparse.c:
	  * ext/audiofile/gstafsink.c:
	  * ext/audiofile/gstafsrc.c:
	  * ext/audioresample/gstaudioresample.c:
	  * ext/bz2/gstbz2dec.c:
	  * ext/bz2/gstbz2enc.c:
	  * ext/cdaudio/gstcdaudio.c:
	  * ext/directfb/dfbvideosink.c:
	  * ext/divx/gstdivxdec.c:
	  * ext/divx/gstdivxenc.c:
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
	  * ext/faac/gstfaac.c: (gst_faac_base_init):
	  * ext/faad/gstfaad.c:
	  * ext/gsm/gstgsmdec.c:
	  * ext/gsm/gstgsmenc.c:
	  * ext/hermes/gsthermescolorspace.c:
	  * ext/ivorbis/vorbisfile.c:
	  * ext/lcs/gstcolorspace.c:
	  * ext/libfame/gstlibfame.c:
	  * ext/libmms/gstmms.c: (gst_mms_base_init):
	  * ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
	  * ext/nas/nassink.c: (gst_nassink_base_init):
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/sdl/sdlaudiosink.c:
	  * ext/sdl/sdlvideosink.c:
	  * ext/shout/gstshout.c:
	  * ext/snapshot/gstsnapshot.c:
	  * ext/sndfile/gstsf.c:
	  * ext/swfdec/gstswfdec.c:
	  * ext/tarkin/gsttarkindec.c:
	  * ext/tarkin/gsttarkinenc.c:
	  * ext/theora/theoradec.c:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
	  * ext/xvid/gstxviddec.c:
	  * ext/xvid/gstxvidenc.c:
	  * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init):
	  * gst/chart/gstchart.c:
	  * gst/colorspace/gstcolorspace.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init):
	  * gst/festival/gstfestival.c:
	  * gst/filter/gstbpwsinc.c:
	  * gst/filter/gstiir.c:
	  * gst/filter/gstlpwsinc.c:
	  * gst/freeze/gstfreeze.c:
	  * gst/games/gstpuzzle.c: (gst_puzzle_base_init):
	  * gst/librfb/gstrfbsrc.c:
	  * gst/mixmatrix/mixmatrix.c:
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  * gst/mpeg2sub/gstmpeg2subt.c:
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  * gst/multifilesink/gstmultifilesink.c:
	  * gst/overlay/gstoverlay.c:
	  * gst/passthrough/gstpassthrough.c:
	  * gst/playondemand/gstplayondemand.c:
	  * gst/qtdemux/qtdemux.c:
	  * gst/rtjpeg/gstrtjpegdec.c:
	  * gst/rtjpeg/gstrtjpegenc.c:
	  * gst/smooth/gstsmooth.c:
	  * gst/smoothwave/gstsmoothwave.c:
	  * gst/spectrum/gstspectrum.c:
	  * gst/speed/gstspeed.c:
	  * gst/stereo/gststereo.c:
	  * gst/switch/gstswitch.c:
	  * gst/tta/gstttadec.c: (gst_tta_dec_base_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_base_init):
	  * gst/vbidec/gstvbidec.c:
	  * gst/videocrop/gstvideocrop.c:
	  * gst/videodrop/gstvideodrop.c:
	  * gst/virtualdub/gstxsharpen.c:
	  * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init):
	  * gst/y4m/gsty4mencode.c:
	  * sys/cdrom/gstcdplayer.c:
	  * sys/directdraw/gstdirectdrawsink.c:
	  * sys/directsound/gstdirectsoundsink.c:
	  * sys/glsink/glimagesink.c:
	  * sys/qcam/gstqcamsrc.c:
	  * sys/v4l2/gstv4l2src.c:
	  * sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init):
	  * sys/ximagesrc/ximagesrc.c:
	  Define GstElementDetails as const and also static (when defined as
	  global)

2006-04-08 21:48:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD] Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
	  Original commit message from CVS:
	  * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init):
	  * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init):
	  * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init):
	  * ext/arts/gst_arts.c: (gst_arts_class_init):
	  * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init):
	  * ext/audiofile/gstafsink.c: (gst_afsink_class_init):
	  * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init):
	  * ext/audioresample/gstaudioresample.c:
	  * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init):
	  * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init):
	  * ext/divx/gstdivxdec.c: (gst_divxdec_class_init):
	  * ext/hermes/gsthermescolorspace.c:
	  (gst_hermes_colorspace_class_init):
	  * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init):
	  * ext/jack/gstjack.c: (gst_jack_class_init):
	  * ext/jack/gstjackbin.c: (gst_jack_bin_class_init):
	  * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init):
	  * ext/libfame/gstlibfame.c: (gst_fameenc_class_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init):
	  * ext/nas/nassink.c: (gst_nassink_class_init):
	  * ext/shout/gstshout.c: (gst_icecastsend_class_init):
	  * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
	  * ext/sndfile/gstsf.c: (gst_sf_class_init):
	  * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init),
	  (gst_swfdec_class_init):
	  * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init):
	  * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init):
	  * gst/chart/gstchart.c: (gst_chart_class_init):
	  * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init):
	  * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init):
	  * gst/festival/gstfestival.c: (gst_festival_class_init):
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
	  * gst/filter/gstiir.c: (gst_iir_class_init):
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
	  * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init):
	  * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init):
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  (gst_system_encode_class_init):
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  (gst_mp1videoparse_class_init):
	  * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init):
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  (gst_mp3parse_class_init):
	  * gst/overlay/gstoverlay.c: (gst_overlay_class_init):
	  * gst/passthrough/gstpassthrough.c: (passthrough_class_init):
	  * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init):
	  * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init):
	  * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init):
	  * gst/smooth/gstsmooth.c: (gst_smooth_class_init):
	  * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init):
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
	  * gst/stereo/gststereo.c: (gst_stereo_class_init):
	  * gst/switch/gstswitch.c: (gst_switch_class_init):
	  * gst/tta/gstttadec.c: (gst_tta_dec_class_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_class_init):
	  * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init):
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init):
	  * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init):
	  * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init):
	  * sys/cdrom/gstcdplayer.c: (cdplayer_class_init):
	  * sys/directsound/gstdirectsoundsink.c:
	  (gst_directsoundsink_class_init):
	  * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init):
	  * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init):
	  * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init):
	  * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init):
	  * sys/v4l2/gstv4l2colorbalance.c:
	  (gst_v4l2_color_balance_channel_class_init):
	  * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init),
	  (gst_v4l2_tuner_norm_class_init):
	  * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init):
	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)

2006-04-08 19:04:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD] gst/: Fix more broken GObject macros
	  Original commit message from CVS:
	  * gst/colorspace/gstcolorspace.h:
	  * gst/deinterlace/gstdeinterlace.h:
	  * gst/passthrough/gstpassthrough.h:
	  * gst/y4m/gsty4mencode.h:
	  Fix more broken GObject macros

2006-04-06 11:35:26 +0000  j@bootlab.org <j@bootlab.org>

	  [MOVED FROM BAD] Unify the long descriptions in the plugin details (#337263).
	  Original commit message from CVS:
	  Patch by: j^  <j at bootlab dot org>
	  * ext/amrwb/gstamrwbdec.c:
	  * ext/amrwb/gstamrwbenc.c:
	  * ext/amrwb/gstamrwbparse.c:
	  * ext/arts/gst_arts.c:
	  * ext/artsd/gstartsdsink.c:
	  * ext/audiofile/gstafparse.c:
	  * ext/audiofile/gstafsink.c:
	  * ext/audiofile/gstafsrc.c:
	  * ext/cdaudio/gstcdaudio.c:
	  * ext/directfb/dfbvideosink.c:
	  * ext/divx/gstdivxdec.c:
	  * ext/divx/gstdivxenc.c:
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
	  * ext/faac/gstfaac.c: (gst_faac_base_init):
	  * ext/faad/gstfaad.c:
	  * ext/gsm/gstgsmdec.c:
	  * ext/gsm/gstgsmenc.c:
	  * ext/hermes/gsthermescolorspace.c:
	  * ext/ivorbis/vorbisfile.c:
	  * ext/lcs/gstcolorspace.c:
	  * ext/libfame/gstlibfame.c:
	  * ext/libmms/gstmms.c: (gst_mms_base_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
	  * ext/nas/nassink.c: (gst_nassink_base_init):
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/polyp/polypsink.c: (gst_polypsink_base_init):
	  * ext/sdl/sdlaudiosink.c:
	  * ext/sdl/sdlvideosink.c:
	  * ext/shout/gstshout.c:
	  * ext/snapshot/gstsnapshot.c:
	  * ext/sndfile/gstsf.c:
	  * ext/tarkin/gsttarkindec.c:
	  * ext/tarkin/gsttarkinenc.c:
	  * ext/theora/theoradec.c:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
	  * ext/xvid/gstxviddec.c:
	  * ext/xvid/gstxvidenc.c:
	  * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init):
	  * gst/chart/gstchart.c:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init):
	  * gst/festival/gstfestival.c:
	  * gst/filter/gstiir.c:
	  * gst/filter/gstlpwsinc.c:
	  * gst/freeze/gstfreeze.c:
	  * gst/games/gstpuzzle.c: (gst_puzzle_base_init):
	  * gst/mixmatrix/mixmatrix.c:
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  * gst/mpeg2sub/gstmpeg2subt.c:
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  * gst/multifilesink/gstmultifilesink.c:
	  * gst/overlay/gstoverlay.c:
	  * gst/passthrough/gstpassthrough.c:
	  * gst/playondemand/gstplayondemand.c:
	  * gst/qtdemux/qtdemux.c:
	  * gst/rtjpeg/gstrtjpegdec.c:
	  * gst/rtjpeg/gstrtjpegenc.c:
	  * gst/smooth/gstsmooth.c:
	  * gst/tta/gstttadec.c: (gst_tta_dec_base_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_base_init):
	  * gst/videocrop/gstvideocrop.c:
	  * gst/videodrop/gstvideodrop.c:
	  * gst/virtualdub/gstxsharpen.c:
	  * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init):
	  * gst/y4m/gsty4mencode.c:
	  Unify the long descriptions in the plugin details (#337263).

2006-04-01 10:09:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] rework build; add translations for v4l2
	  Original commit message from CVS:
	  rework build; add translations for v4l2

2005-09-05 17:20:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] Fix up all the state change functions.
	  Original commit message from CVS:
	  Fix up all the state change functions.

2005-07-05 10:51:49 +0000  Andy Wingo <wingo@pobox.com>

	  [MOVED FROM BAD] Way, way, way too many files: Remove crack comment from the 2000 era.
	  Original commit message from CVS:
	  2005-07-05  Andy Wingo  <wingo@pobox.com>
	  * Way, way, way too many files:
	  Remove crack comment from the 2000 era.

2005-01-14 18:36:42 +0000  Stéphane Loeuillet <gstreamer@leroutier.net>

	  [MOVED FROM BAD] I'm a bad boy. using /1001. to force C to do float division and not integer division (as it did in my last commit)
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c:
	  * gst/subparse/gstsubparse.c: (parse_mdvdsub):
	  * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
	  I'm a bad boy. using /1001. to force C to do float division
	  and not integer division (as it did in my last commit)
	  Thanks to David I. Lehn for pointing this mistake.

2005-01-14 12:27:22 +0000  Stéphane Loeuillet <gstreamer@leroutier.net>

	  [MOVED FROM BAD] replace framerate aproximations by their real value (24000/1001, 30000/1001, 60000/1001)
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c:
	  * ext/libfame/gstlibfame.c:
	  * gst/subparse/gstsubparse.c: (parse_mdvdsub):
	  * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
	  replace framerate aproximations by their real value
	  (24000/1001, 30000/1001, 60000/1001)
	  Finish fixing bug #164049

2004-07-27 21:41:30 +0000  Steve Lhomme <steve.lhomme@free.fr>

	* gst/y4m/y4menc.vcproj:
	  [MOVED FROM BAD] more working plugins
	  Original commit message from CVS:
	  more working plugins

2004-07-27 09:57:33 +0000  Steve Lhomme <steve.lhomme@free.fr>

	* gst/y4m/y4menc.vcproj:
	  [MOVED FROM BAD] rename GStreamer-0.8.lib to libgstreamer.lib
	  Original commit message from CVS:
	  rename GStreamer-0.8.lib to libgstreamer.lib

2004-07-27 09:48:51 +0000  Steve Lhomme <steve.lhomme@free.fr>

	* gst/y4m/y4menc.vcproj:
	  [MOVED FROM BAD] avoid problems with math.h, fix release dependancy
	  Original commit message from CVS:
	  avoid problems with math.h, fix release dependancy

2004-07-26 13:20:11 +0000  Steve Lhomme <steve.lhomme@free.fr>

	* gst/y4m/y4menc.vcproj:
	  [MOVED FROM BAD] more plugins supported under windows
	  Original commit message from CVS:
	  more plugins supported under windows

2004-04-01 11:48:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] a52dec:   Use a debug category, Output timestamps correctly
	  Original commit message from CVS:
	  a52dec:   Use a debug category, Output timestamps correctly
	  Emit tag info, Handle events, tell liba52dec about cpu
	  capabilities so it can use MMX etc.
	  dvdec:    Fix a crasher accessing invalid memory
	  dvdnavsrc:Some support for byte-format seeking.
	  Small fixes for still frames and menu button overlays
	  mpeg2dec: Use a debug category. Adjust the report level of several items to
	  LOG. Call mpeg2_custom_fbuf to mark our buffers as 'custom buffers'
	  so it doesn't lose the GstBuffer pointer
	  navseek:  Add the navseek debug element for seeking back and forth in a
	  video stream using arrow keys.
	  mpeg2subt:Pretty much a complete rewrite. Now a loopbased element. May still
	  require work to properly synchronise subtitle buffers.
	  mpegdemux:
	  dvddemux: Don't attempt to create subbuffers of size 0
	  Reduce a couple of error outputs to warnings.
	  y4mencode:Output the y4m frame header correctly

2004-03-15 19:32:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] don't mix tabs and spaces
	  Original commit message from CVS:
	  don't mix tabs and spaces

2004-03-15 16:32:54 +0000  Johan Dahlin <johan@gnome.org>

	  [MOVED FROM BAD] *.h: Revert indenting
	  Original commit message from CVS:
	  * *.h: Revert indenting

2004-03-14 22:34:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  [MOVED FROM BAD] gst-indent
	  Original commit message from CVS:
	  gst-indent

2004-01-12 02:01:52 +0000  Benjamin Otte <otte@gnome.org>

	  [MOVED FROM BAD] gst-libs/gst/video/video.h: Fix caps template names to be understandable.
	  Original commit message from CVS:
	  2004-01-12  Benjamin Otte  <in7y118@public.uni-hamburg.de>
	  * gst-libs/gst/video/video.h:
	  Fix caps template names to be understandable.
	  Prefix everything with GST_VIDEO.
	  * ext/aalib/gstaasink.c:
	  * ext/divx/gstdivxdec.c:
	  * ext/divx/gstdivxenc.c:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c:
	  * ext/hermes/gstcolorspace.c: (gst_colorspace_base_init):
	  * ext/jpeg/gstjpegdec.c: (raw_caps_factory):
	  * ext/jpeg/gstjpegenc.c: (raw_caps_factory):
	  * ext/libcaca/gstcacasink.c:
	  * ext/libpng/gstpngenc.c: (raw_caps_factory):
	  * ext/snapshot/gstsnapshot.c:
	  * ext/swfdec/gstswfdec.c:
	  * ext/xvid/gstxviddec.c:
	  * ext/xvid/gstxvidenc.c:
	  * gst/chart/gstchart.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/effectv/gsteffectv.c:
	  * gst/flx/gstflxdec.c: (gst_flxdec_loop):
	  * gst/goom/gstgoom.c:
	  * gst/median/gstmedian.c:
	  * gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
	  (gst_monoscope_srcconnect), (gst_monoscope_chain):
	  * gst/overlay/gstoverlay.c:
	  * gst/smooth/gstsmooth.c:
	  * gst/smpte/gstsmpte.c:
	  * gst/synaesthesia/gstsynaesthesia.c:
	  * gst/videocrop/gstvideocrop.c:
	  * gst/videodrop/gstvideodrop.c:
	  * gst/y4m/gsty4mencode.c:
	  * sys/qcam/gstqcamsrc.c:
	  * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
	  Make them work with new video.h file.
	  * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
	  (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
	  (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc):
	  Make it work with new buffer allocation system.

2003-12-22 01:47:09 +0000  David Schleef <ds@schleef.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] Merge CAPS branch
	  Original commit message from CVS:
	  Merge CAPS branch

2003-12-04 10:37:38 +0000  Andy Wingo <wingo@pobox.com>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] remove copyright field from plugins
	  Original commit message from CVS:
	  remove copyright field from plugins

2003-11-16 22:02:23 +0000  Leif Johnson <leif@ambient.2y.net>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] + checking in plugin category changes
	  Original commit message from CVS:
	  + checking in plugin category changes

2003-11-07 12:47:02 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* gst/y4m/gsty4mencode.h:
	  [MOVED FROM BAD] Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro...
	  Original commit message from CVS:
	  Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files

2003-11-02 19:17:27 +0000  Benjamin Otte <otte@gnome.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] fix to new plugin system
	  Original commit message from CVS:
	  fix to new plugin system

2003-10-08 16:08:19 +0000  Andy Wingo <wingo@pobox.com>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
	  Original commit message from CVS:
	  /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.

2003-08-10 00:01:58 +0000  David Schleef <ds@schleef.org>

	* gst/y4m/Makefile.am:
	  [MOVED FROM BAD] Remove redundant plugindir definition
	  Original commit message from CVS:
	  Remove redundant plugindir definition

2003-07-06 20:49:52 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  [MOVED FROM BAD] New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri...
	  Original commit message from CVS:
	  New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs

2003-06-29 19:46:13 +0000  Benjamin Otte <otte@gnome.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] compatibility fix for new GST_DEBUG stuff.
	  Original commit message from CVS:
	  compatibility fix for new GST_DEBUG stuff.
	  Includes fixes for missing includes for config.h and unistd.h
	  I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.

2003-01-10 13:38:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] PadConnect -> PadLink
	  Original commit message from CVS:
	  PadConnect -> PadLink

2003-01-10 10:22:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t...
	  Original commit message from CVS:
	  another batch of connect->link fixes
	  please let me know about issues
	  and please refrain of making them yourself, so that I don't spend double
	  the time resolving conflicts

2002-12-08 14:50:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/Makefile.am:
	  [MOVED FROM BAD] parallel install fixes
	  Original commit message from CVS:
	  parallel install fixes

2002-09-18 19:02:52 +0000  Christian Schaller <uraeus@gnome.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] plugins part of license field patch
	  Original commit message from CVS:
	  plugins part of license field patch

2002-06-17 10:29:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/Makefile.am:
	  [MOVED FROM BAD] cosmetic change
	  Original commit message from CVS:
	  cosmetic change

2002-05-03 09:59:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] various name fixes and sundry
	  Original commit message from CVS:
	  various name fixes and sundry

2002-04-20 21:42:51 +0000  Andy Wingo <wingo@pobox.com>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br...
	  Original commit message from CVS:
	  * a hack to work around intltool's brokenness
	  * a current check for mpeg2dec
	  * details->klass reorganizations
	  * an element browser that uses details->klass
	  * separated cdxa parse out from the avi directory

2002-04-11 20:42:26 +0000  Andy Wingo <wingo@pobox.com>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind.
	  Original commit message from CVS:
	  GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE
	  same with *factory and typefind.
	  also, some -Werror fixes.

2002-03-30 17:06:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] Changed to the new props API
	  Original commit message from CVS:
	  Changed to the new props API
	  Other small tuff.

2002-03-20 21:45:04 +0000  Andy Wingo <wingo@pobox.com>

	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  [MOVED FROM BAD] s/Gnome-Streamer/GStreamer/
	  Original commit message from CVS:
	  s/Gnome-Streamer/GStreamer/

2002-03-19 04:10:06 +0000  Andy Wingo <wingo@pobox.com>

	* gst/y4m/Makefile.am:
	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  [MOVED FROM BAD] removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with...
	  Original commit message from CVS:
	  * removal of //-style comments
	  * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
	  and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.

2002-03-19 01:39:43 +0000  Andy Wingo <wingo@pobox.com>

	* gst/y4m/Makefile.am:
	  [MOVED FROM BAD] s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi...
	  Original commit message from CVS:
	  s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/
	  @-substitued variables variables are defined as make variables automagically,
	  and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag

2002-01-18 11:37:19 +0000  Wrobell <wrobell@ite.pl>

	* gst/y4m/Makefile.am:
	  [MOVED FROM BAD] - plugins are built without versioning info
	  Original commit message from CVS:
	  - plugins are built without versioning info

2002-01-13 22:27:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] Bring the plugins in sync with the new core capsnego system.
	  Original commit message from CVS:
	  Bring the plugins in sync with the new core capsnego system.
	  Added some features, enhancements...

2002-01-12 03:34:27 +0000  David I. Lehn <dlehn@users.sourceforge.net>

	* gst/y4m/Makefile.am:
	  [MOVED FROM BAD] s/filter/plugin/ link plugins to GST_LIBS rearrange rules to a common format
	  Original commit message from CVS:
	  * s/filter/plugin/
	  * link plugins to GST_LIBS
	  * rearrange rules to a common format

2001-12-23 20:21:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/Makefile.am:
	* gst/y4m/gsty4mencode.c:
	  [MOVED FROM BAD] more fixes
	  Original commit message from CVS:
	  more fixes

2001-12-23 13:17:36 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/y4m/Makefile.am:
	* gst/y4m/gsty4mencode.c:
	* gst/y4m/gsty4mencode.h:
	  [MOVED FROM BAD] BBB asked me to rename lav to y4m can someone who knows the plugin do this in the source as well ?
	  Original commit message from CVS:
	  BBB asked me to rename lav to y4m
	  can someone who knows the plugin do this in the source as well ?

2009-05-15 18:17:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/Makevars:
	  po: add Makevars magic so we don't get line numbers in *.po files
	  This avoids the number one reason for local modifications in *.po
	  files and and makes things less annoying when working with git (or
	  any other VCS for that matter).

2009-05-15 17:11:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/id3demux.c:
	* tests/check/elements/souphttpsrc.c:
	* tests/check/pipelines/flacdec.c:
	* tests/files/Makefile.am:
	* tests/files/audiotestsrc.flac:
	* tests/files/test-cert.pem:
	* tests/files/test-key.pem:
	  checks: move files required by unit tests into tests/files and make sure they're disted
	  Move unit test data into the directory where it belongs and make in particular
	  the flacdec unit test cd into the directory with the test files instead of making
	  assumptions about the current working directory in that unit test. As a side effect
	  of movng those files, there's only one EXTRA_DIST in tests/check/Makefile.am now,
	  which is likely to work better than having two. Hopefully fixes #582753.

2009-05-14 21:43:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: If the upstream max latency is unbound return unbound max latency
	  Fixes bug #582661.

2009-05-15 08:44:39 +0200  James Andrewartha <trs80@ucc.gu.uwa.edu.au>

	* gst/flv/gstflvmux.c:
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixertrack.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/v4l2/v4l2_calls.c:
	  Fix compiler warnings
	  Fixes bug #582715.

2009-05-14 12:32:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: Improve debugging a bit

2009-05-13 22:46:44 +0200  Josep Torra <n770galaxy@gmail.com>

	* configure.ac:
	  Recovered debugutils line accidentally removed in deinterlace2 move.

2009-05-13 10:46:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* gst/deinterlace/Makefile.am:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.asm:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/greedyhmacros.h:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/mmx.h:
	* gst/deinterlace/tvtime/plugins.h:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/sse.h:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoop0A.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopBottom.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopEdgeA.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopEdgeA8.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddA.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddA2.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddA6.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddAH.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddAH2.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopTop.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopVA.inc:
	* gst/deinterlace/tvtime/tomsmocomp/SearchLoopVAH.inc:
	* gst/deinterlace/tvtime/tomsmocomp/StrangeBob.inc:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll2.inc:
	* gst/deinterlace/tvtime/tomsmocomp/WierdBob.inc:
	* gst/deinterlace/tvtime/tomsmocomp/tomsmocompmacros.h:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	* gst/deinterlace/tvtime/x86-64_macros.inc:
	  Moved 'deinterlace2' from -bad to -good
	  And rename it to deinterlace.

2009-05-08 15:39:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	* gst/deinterlace2/gstdeinterlace2.h:
	  [MOVED FROM BAD 56/56] deinterlace2: Add a disabled mode for passthrough operation
	  Also allow to change the mode in PAUSED and PLAYING by updating
	  the caps if necessary.

2009-04-22 19:43:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	* gst/deinterlace2/gstdeinterlace2.h:
	  [MOVED FROM BAD 55/56] deinterlace2: Add documentation and integrate into the build system

2009-04-19 17:18:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	  [MOVED FROM BAD 54/56] deinterlace2: Make it possible to select interlacing autodetection or to enfore deinterlacing
	  For this add a "mode" property that defaults to "interlaced" for now as
	  most decoders/demuxers don't properly set the "interlaced" field on the
	  caps yet.
	  If this property is set to "auto" the element will work in passthrough
	  mode unless the caps contain the "interlaced" field.

2009-04-17 15:39:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	  [MOVED FROM BAD 53/56] deinterlace2: Use GST_(DEBUG|WARNING|ERROR)_OBJECT instead of the non-OBJECT ones

2009-04-17 15:39:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	  [MOVED FROM BAD 52/56] deinterlace2: Reset history if DISCONT is set on the incoming buffer

2009-04-17 15:39:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	  [MOVED FROM BAD 51/56] deinterlace2: Fix timestamps for buffers with RFF flag set

2009-04-16 17:41:37 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	* gst/deinterlace2/gstdeinterlace2.h:
	* gst/deinterlace2/tvtime/greedy.c:
	* gst/deinterlace2/tvtime/greedyh.c:
	* gst/deinterlace2/tvtime/scalerbob.c:
	* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	* gst/deinterlace2/tvtime/weave.c:
	* gst/deinterlace2/tvtime/weavebff.c:
	* gst/deinterlace2/tvtime/weavetff.c:
	  [MOVED FROM BAD 50/56] deinterlace2: Rename line_length to row_stride and remove output_stride

2009-04-16 15:52:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	  [MOVED FROM BAD 49/56] deinterlace2: Implement support for RFF and ONEFIELD buffer flags

2009-04-15 15:46:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	* gst/deinterlace2/gstdeinterlace2.h:
	* gst/deinterlace2/tvtime/greedy.c:
	* gst/deinterlace2/tvtime/greedyh.c:
	* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  [MOVED FROM BAD 48/56] deinterlace2: Move output buffer from the instance struct to a function parameter

2009-04-15 15:33:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	* gst/deinterlace2/gstdeinterlace2.h:
	  [MOVED FROM BAD 47/56] deinterlace2: Add initial support for automatic detection of the field order

2009-04-15 14:47:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace2/gstdeinterlace2.c:
	  [MOVED FROM BAD 46/56] deinterlace2: Add support for YVYU colorspace
	  This is the same as YUY2 with just Cr and Cb swapped. As
	  we don't make a difference between them when deinterlacing
	  this works.

2008-11-06 14:05:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  [MOVED FROM BAD 45/56] gst/deinterlace2/gstdeinterlace2.c: Bring properties into this century.
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace2_class_init), (gst_deinterlace2_init),
	  (gst_deinterlace2_set_property), (gst_deinterlace2_get_property):
	  Bring properties into this century.

2008-11-04 12:42:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD 44/56] Don't install static libs for plugins. Fixes #550851 for -bad.
	  Original commit message from CVS:
	  * ext/alsaspdif/Makefile.am:
	  * ext/amrwb/Makefile.am:
	  * ext/apexsink/Makefile.am:
	  * ext/arts/Makefile.am:
	  * ext/artsd/Makefile.am:
	  * ext/audiofile/Makefile.am:
	  * ext/audioresample/Makefile.am:
	  * ext/bz2/Makefile.am:
	  * ext/cdaudio/Makefile.am:
	  * ext/celt/Makefile.am:
	  * ext/dc1394/Makefile.am:
	  * ext/dirac/Makefile.am:
	  * ext/directfb/Makefile.am:
	  * ext/divx/Makefile.am:
	  * ext/dts/Makefile.am:
	  * ext/faac/Makefile.am:
	  * ext/faad/Makefile.am:
	  * ext/gsm/Makefile.am:
	  * ext/hermes/Makefile.am:
	  * ext/ivorbis/Makefile.am:
	  * ext/jack/Makefile.am:
	  * ext/jp2k/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/lcs/Makefile.am:
	  * ext/libfame/Makefile.am:
	  * ext/libmms/Makefile.am:
	  * ext/metadata/Makefile.am:
	  * ext/mpeg2enc/Makefile.am:
	  * ext/mplex/Makefile.am:
	  * ext/musepack/Makefile.am:
	  * ext/musicbrainz/Makefile.am:
	  * ext/mythtv/Makefile.am:
	  * ext/nas/Makefile.am:
	  * ext/neon/Makefile.am:
	  * ext/ofa/Makefile.am:
	  * ext/polyp/Makefile.am:
	  * ext/resindvd/Makefile.am:
	  * ext/sdl/Makefile.am:
	  * ext/shout/Makefile.am:
	  * ext/snapshot/Makefile.am:
	  * ext/sndfile/Makefile.am:
	  * ext/soundtouch/Makefile.am:
	  * ext/spc/Makefile.am:
	  * ext/swfdec/Makefile.am:
	  * ext/tarkin/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/timidity/Makefile.am:
	  * ext/twolame/Makefile.am:
	  * ext/x264/Makefile.am:
	  * ext/xine/Makefile.am:
	  * ext/xvid/Makefile.am:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/dshow/Makefile.am:
	  * gst/aiffparse/Makefile.am:
	  * gst/app/Makefile.am:
	  * gst/audiobuffer/Makefile.am:
	  * gst/bayer/Makefile.am:
	  * gst/cdxaparse/Makefile.am:
	  * gst/chart/Makefile.am:
	  * gst/colorspace/Makefile.am:
	  * gst/dccp/Makefile.am:
	  * gst/deinterlace/Makefile.am:
	  * gst/deinterlace2/Makefile.am:
	  * gst/dvdspu/Makefile.am:
	  * gst/festival/Makefile.am:
	  * gst/filter/Makefile.am:
	  * gst/flacparse/Makefile.am:
	  * gst/flv/Makefile.am:
	  * gst/games/Makefile.am:
	  * gst/h264parse/Makefile.am:
	  * gst/librfb/Makefile.am:
	  * gst/mixmatrix/Makefile.am:
	  * gst/modplug/Makefile.am:
	  * gst/mpeg1sys/Makefile.am:
	  * gst/mpeg4videoparse/Makefile.am:
	  * gst/mpegdemux/Makefile.am:
	  * gst/mpegtsmux/Makefile.am:
	  * gst/mpegvideoparse/Makefile.am:
	  * gst/mve/Makefile.am:
	  * gst/nsf/Makefile.am:
	  * gst/nuvdemux/Makefile.am:
	  * gst/overlay/Makefile.am:
	  * gst/passthrough/Makefile.am:
	  * gst/pcapparse/Makefile.am:
	  * gst/playondemand/Makefile.am:
	  * gst/rawparse/Makefile.am:
	  * gst/real/Makefile.am:
	  * gst/rtjpeg/Makefile.am:
	  * gst/rtpmanager/Makefile.am:
	  * gst/scaletempo/Makefile.am:
	  * gst/sdp/Makefile.am:
	  * gst/selector/Makefile.am:
	  * gst/smooth/Makefile.am:
	  * gst/smoothwave/Makefile.am:
	  * gst/speed/Makefile.am:
	  * gst/speexresample/Makefile.am:
	  * gst/stereo/Makefile.am:
	  * gst/subenc/Makefile.am:
	  * gst/tta/Makefile.am:
	  * gst/vbidec/Makefile.am:
	  * gst/videodrop/Makefile.am:
	  * gst/videosignal/Makefile.am:
	  * gst/virtualdub/Makefile.am:
	  * gst/vmnc/Makefile.am:
	  * gst/y4m/Makefile.am:
	  * sys/acmenc/Makefile.am:
	  * sys/cdrom/Makefile.am:
	  * sys/dshowdecwrapper/Makefile.am:
	  * sys/dshowsrcwrapper/Makefile.am:
	  * sys/dvb/Makefile.am:
	  * sys/dxr3/Makefile.am:
	  * sys/fbdev/Makefile.am:
	  * sys/oss4/Makefile.am:
	  * sys/qcam/Makefile.am:
	  * sys/qtwrapper/Makefile.am:
	  * sys/vcd/Makefile.am:
	  * sys/wininet/Makefile.am:
	  * win32/common/config.h:
	  Don't install static libs for plugins. Fixes #550851 for -bad.

2008-10-09 19:38:52 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 43/56] gst/deinterlace2/tvtime/tomsmocomp.c: Fix unused variable compiler warning when not building
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/tomsmocomp.c:
	  (gst_deinterlace_method_tomsmocomp_class_init):
	  Fix unused variable compiler warning when not building
	  X86 assembly.

2008-08-28 17:16:51 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  [MOVED FROM BAD 42/56] gst/dccp/: Fix compilation on Solaris by including filio.h as needed.
	  Original commit message from CVS:
	  * gst/dccp/gstdccp.c:
	  * gst/dccp/gstdccpclientsrc.c:
	  Fix compilation on Solaris by including filio.h as needed.
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
	  Fix compilation with Forte - apparently it hates concatenating a
	  macro argument that starts with an underscore??

2008-08-26 12:33:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 41/56] gst/deinterlace2/tvtime/tomsmocomp/: Unroll the loop to handle two bytes at once. This should give a small speedup an...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
	  Unroll the loop to handle two bytes at once. This should give
	  a small speedup and makes it possible to handle chroma and luma
	  different which is needed later.

2008-08-25 14:37:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 40/56] gst/deinterlace2/: First part of the C implementation of the tomsmocomp deinterlacing algorithm. This only supports s...
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace_method_class_init):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  * gst/deinterlace2/tvtime/tomsmocomp.c:
	  (gst_deinterlace_method_tomsmocomp_class_init):
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
	  First part of the C implementation of the tomsmocomp deinterlacing
	  algorithm. This only supports search-effort=0 currently, is painfully
	  slow and needs some cleanup later when all search-effort settings
	  are implemented in C.

2008-08-02 18:48:17 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 39/56] gst/deinterlace2/: Use oil_memcpy() instead of memcpy() as it's faster for the sizes that are usually used here.
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace_simple_method_interpolate_scanline),
	  (gst_deinterlace_simple_method_copy_scanline),
	  (gst_deinterlace_simple_method_deinterlace_frame):
	  * gst/deinterlace2/tvtime/greedy.c: (deinterlace_frame_di_greedy):
	  * gst/deinterlace2/tvtime/greedyh.c:
	  (deinterlace_frame_di_greedyh):
	  * gst/deinterlace2/tvtime/scalerbob.c:
	  (deinterlace_scanline_scaler_bob):
	  * gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy):
	  * gst/deinterlace2/tvtime/weave.c: (deinterlace_scanline_weave),
	  (copy_scanline):
	  * gst/deinterlace2/tvtime/weavebff.c: (deinterlace_scanline_weave),
	  (copy_scanline):
	  * gst/deinterlace2/tvtime/weavetff.c: (deinterlace_scanline_weave),
	  (copy_scanline):
	  Use oil_memcpy() instead of memcpy() as it's faster for the sizes that
	  are usually used here.

2008-08-02 18:36:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 38/56] gst/deinterlace2/: Add the remaining tvtime deinterlacing methods and fix the deinterlace_frame() implementation of G...
	  Original commit message from CVS:
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace_simple_method_deinterlace_frame),
	  (gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  * gst/deinterlace2/tvtime/linear.c:
	  (deinterlace_scanline_linear_c), (deinterlace_scanline_linear_mmx),
	  (deinterlace_scanline_linear_mmxext),
	  (gst_deinterlace_method_linear_class_init),
	  (gst_deinterlace_method_linear_init):
	  * gst/deinterlace2/tvtime/linearblend.c:
	  (deinterlace_scanline_linear_blend_c),
	  (deinterlace_scanline_linear_blend2_c),
	  (deinterlace_scanline_linear_blend_mmx),
	  (deinterlace_scanline_linear_blend2_mmx),
	  (gst_deinterlace_method_linear_blend_class_init),
	  (gst_deinterlace_method_linear_blend_init):
	  * gst/deinterlace2/tvtime/plugins.h:
	  * gst/deinterlace2/tvtime/scalerbob.c:
	  (deinterlace_scanline_scaler_bob),
	  (gst_deinterlace_method_scaler_bob_class_init),
	  (gst_deinterlace_method_scaler_bob_init):
	  * gst/deinterlace2/tvtime/weave.c: (deinterlace_scanline_weave),
	  (copy_scanline), (gst_deinterlace_method_weave_class_init),
	  (gst_deinterlace_method_weave_init):
	  * gst/deinterlace2/tvtime/weavebff.c: (deinterlace_scanline_weave),
	  (copy_scanline), (gst_deinterlace_method_weave_bff_class_init),
	  (gst_deinterlace_method_weave_bff_init):
	  * gst/deinterlace2/tvtime/weavetff.c: (deinterlace_scanline_weave),
	  (copy_scanline), (gst_deinterlace_method_weave_tff_class_init),
	  (gst_deinterlace_method_weave_tff_init):
	  Add the remaining tvtime deinterlacing methods and fix the
	  deinterlace_frame() implementation of GstDeinterlaceSimpleMethod.

2008-08-02 18:30:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 37/56] gst/deinterlace2/tvtime/vfir.c: Implement the VFIR deinterlacing method as simple method.
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c),
	  (deinterlace_line_mmx), (gst_deinterlace_method_vfir_class_init):
	  Implement the VFIR deinterlacing method as simple method.

2008-08-02 18:18:54 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 36/56] gst/deinterlace2/gstdeinterlace2.*: Add a GstDeinterlaceSimpleMethod subclass of GstDeinterlaceMethod that can be use...
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace_simple_method_interpolate_scanline),
	  (gst_deinterlace_simple_method_copy_scanline),
	  (gst_deinterlace_simple_method_deinterlace_frame),
	  (gst_deinterlace_simple_method_class_init),
	  (gst_deinterlace_simple_method_init):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  Add a GstDeinterlaceSimpleMethod subclass of GstDeinterlaceMethod that
	  can be used by simple deinterlacing methods. They only have to provide
	  a function for interpolating a scanline or copying a scanline.

2008-08-02 18:15:49 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 35/56] gst/deinterlace2/gstdeinterlace2.c: Respect the latency of the deinterlacing algorithm for the timestamps of every bu...
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_chain):
	  Respect the latency of the deinterlacing algorithm for the timestamps
	  of every buffer.

2008-08-02 18:13:20 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 34/56] gst/deinterlace2/tvtime/: Add the MMX registers to the clobbered registers only if __MMX__ is defined.
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
	  Add the MMX registers to the clobbered registers only if __MMX__ is
	  defined.

2008-08-02 18:09:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 33/56] gst/deinterlace2/: Enable tomsmocomp again as the C port will be ready for the next release.
	  Original commit message from CVS:
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method),
	  (gst_deinterlace2_class_init):
	  Enable tomsmocomp again as the C port will be ready for the next
	  release.

2008-08-02 18:02:44 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 32/56] gst/deinterlace2/gstdeinterlace2.c: Don't use proxy_getcaps() but implement our own getcaps() function that doubles/h...
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_init),
	  (gst_greatest_common_divisor), (gst_fraction_double),
	  (gst_deinterlace2_getcaps), (gst_deinterlace2_setcaps):
	  Don't use proxy_getcaps() but implement our own getcaps() function
	  that doubles/halfs the framerate if all fields should be sent out.

2008-07-18 08:34:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 31/56] Disable the tomsmocomp algorithm for this release as it's buggy and has no C implementation yet.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method),
	  (gst_deinterlace2_class_init), (gst_deinterlace2_init):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (gst_deinterlace_method_greedy_l_class_init):
	  * gst/deinterlace2/tvtime/greedyh.c:
	  (gst_deinterlace_method_greedy_h_class_init):
	  * gst/deinterlace2/tvtime/vfir.c:
	  (gst_deinterlace_method_vfir_class_init):
	  Disable the tomsmocomp algorithm for this release as it's buggy
	  and has no C implementation yet.
	  Build the deinterlace2 plugin on all architectures but still mark it
	  as experimental.
	  Build the x86 inline assembly only if GCC inline assembly is supported
	  and only on x86 or amd64. Fixes bug #543286.

2008-07-14 14:13:54 +0000  Edward Hervey <bilboed@bilboed.com>

	  [MOVED FROM BAD 30/56] gst/deinterlace2/tvtime/: Fix build on x86_64
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (gst_deinterlace_method_greedy_l_class_init):
	  * gst/deinterlace2/tvtime/greedyh.c:
	  (gst_deinterlace_method_greedy_h_class_init):
	  * gst/deinterlace2/tvtime/vfir.c:
	  (gst_deinterlace_method_vfir_class_init):
	  Fix build on x86_64

2008-07-13 10:56:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 29/56] gst/deinterlace2/tvtime/greedyh.asm: Always use the C implementation if width is not a multiple of 4. The assembly op...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  Always use the C implementation if width is not a multiple of 4. The
	  assembly optimized version only handle this and calling the C
	  implementation for the remaining part doesn't work because it needs
	  previous calculations.

2008-07-13 10:52:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 28/56] gst/deinterlace2/tvtime/: Some cleanup, use 3DNOW instead of TDNOW in macros.
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  * gst/deinterlace2/tvtime/greedyh.c:
	  * gst/deinterlace2/tvtime/greedyhmacros.h:
	  Some cleanup, use 3DNOW instead of TDNOW in macros.
	  * gst/deinterlace2/tvtime/tomsmocomp.c:
	  (gst_deinterlace_method_tomsmocomp_class_init):
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
	  The SSE method in fact only needs MMXEXT, declare it as such.

2008-07-08 13:31:37 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 27/56] Don't use declarations after statements in the remaining code.
	  Original commit message from CVS:
	  * ext/spc/gstspc.c: (spc_setup):
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
	  Don't use declarations after statements in the remaining code.

2008-07-06 20:43:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 26/56] gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc: Mark internal processing functions as static inline for quite ...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
	  Mark internal processing functions as static inline for quite some
	  speedup as they're used only once and need to get many local variables
	  passed as parameter.

2008-07-05 19:20:30 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 25/56] gst/deinterlace2/gstdeinterlace2.*: Call the current instance "self" instead of "object".
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace_method_deinterlace_frame),
	  (gst_deinterlace2_set_method), (gst_deinterlace2_init),
	  (gst_deinterlace2_reset_history), (gst_deinterlace2_reset),
	  (gst_deinterlace2_set_property), (gst_deinterlace2_get_property),
	  (gst_deinterlace2_pop_history), (gst_deinterlace2_head_history),
	  (gst_deinterlace2_push_history), (gst_deinterlace2_chain),
	  (gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event),
	  (gst_deinterlace2_change_state), (gst_deinterlace2_src_event),
	  (gst_deinterlace2_src_query):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  Call the current instance "self" instead of "object".

2008-07-05 19:11:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 24/56] gst/deinterlace2/gstdeinterlace2.*: Include latency of the method in the returned latency.
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace_method_get_latency),
	  (gst_deinterlace2_set_method), (gst_deinterlace2_class_init),
	  (gst_deinterlace2_push_history), (gst_deinterlace2_chain),
	  (gst_deinterlace2_setcaps), (gst_deinterlace2_src_query):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  Include latency of the method in the returned latency.
	  Fix outputting of all fields, i.e. doubling of the framerate.

2008-07-05 16:47:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 23/56] gst/deinterlace2/: Use a GstObject subtype for the deinterlacing methods and export the different settings for each d...
	  Original commit message from CVS:
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace_method_class_init), (gst_deinterlace_method_init),
	  (gst_deinterlace_method_deinterlace_frame),
	  (gst_deinterlace_method_get_fields_required),
	  (gst_deinterlace2_methods_get_type), (_do_init),
	  (gst_deinterlace2_set_method), (gst_deinterlace2_class_init),
	  (gst_deinterlace2_child_proxy_get_child_by_index),
	  (gst_deinterlace2_child_proxy_get_children_count),
	  (gst_deinterlace2_child_proxy_interface_init),
	  (gst_deinterlace2_init), (gst_deinterlace2_finalize),
	  (gst_deinterlace2_chain), (gst_deinterlace2_src_query):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (deinterlace_greedy_packed422_scanline_c),
	  (deinterlace_greedy_packed422_scanline_mmx),
	  (deinterlace_greedy_packed422_scanline_mmxext),
	  (deinterlace_frame_di_greedy),
	  (gst_deinterlace_method_greedy_l_set_property),
	  (gst_deinterlace_method_greedy_l_get_property),
	  (gst_deinterlace_method_greedy_l_class_init),
	  (gst_deinterlace_method_greedy_l_init):
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  * gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C),
	  (deinterlace_frame_di_greedyh),
	  (gst_deinterlace_method_greedy_h_set_property),
	  (gst_deinterlace_method_greedy_h_get_property),
	  (gst_deinterlace_method_greedy_h_class_init),
	  (gst_deinterlace_method_greedy_h_init):
	  * gst/deinterlace2/tvtime/greedyh.h:
	  * gst/deinterlace2/tvtime/plugins.h:
	  * gst/deinterlace2/tvtime/tomsmocomp.c:
	  (gst_deinterlace_method_tomsmocomp_set_property),
	  (gst_deinterlace_method_tomsmocomp_get_property),
	  (gst_deinterlace_method_tomsmocomp_class_init),
	  (gst_deinterlace_method_tomsmocomp_init):
	  * gst/deinterlace2/tvtime/tomsmocomp.h:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir),
	  (gst_deinterlace_method_vfir_class_init),
	  (gst_deinterlace_method_vfir_init):
	  Use a GstObject subtype for the deinterlacing methods and export
	  the different settings for each deinterlacing method via GObject
	  properties.
	  Implement GstChildProxy interface to allow access to the used
	  deinterlacing method and to allow adjusting the different settings.
	  Move global variables of the tomsmocomp deinterlacing method into
	  function local variables to make it possible to use this deinterlacing
	  method from different instances.

2008-07-05 12:22:37 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 22/56] gst/deinterlace2/tvtime/greedyh.asm: Support widths that are not a multiply of 4 when using the assembly optimized gr...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  Support widths that are not a multiply of 4 when using the assembly
	  optimized greedyh implementations.

2008-07-04 18:54:15 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 21/56] gst/deinterlace2/tvtime/greedyh.c: Only build the assembly optimized implementations on x86.
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedyh.c:
	  (deinterlace_frame_di_greedyh):
	  Only build the assembly optimized implementations on x86.

2008-06-30 07:51:07 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 20/56] gst/deinterlace2/: Remove useless file and mark everything possible as static.
	  Original commit message from CVS:
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/tvtime/tomsmocomp.c: (tomsmocomp_init),
	  (tomsmocomp_filter_mmx), (tomsmocomp_filter_3dnow),
	  (tomsmocomp_filter_sse), (deinterlace_frame_di_tomsmocomp):
	  * gst/deinterlace2/tvtime/tomsmocomp.h:
	  Remove useless file and mark everything possible as static.
	  * gst/deinterlace2/tvtime/greedy.c:
	  * gst/deinterlace2/tvtime/greedyh.c:
	  Use "_stdint.h" instead of <stdint.h>.

2008-06-29 10:56:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 19/56] gst/deinterlace2/: Get rid of speedy.[ch] as we don't use most of it's code anyway and it doesn't seem to be relicens...
	  Original commit message from CVS:
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_init):
	  * gst/deinterlace2/tvtime/greedy.c: (deinterlace_frame_di_greedy):
	  * gst/deinterlace2/tvtime/greedyh.c:
	  (deinterlace_frame_di_greedyh):
	  * gst/deinterlace2/tvtime/speedtools.h:
	  * gst/deinterlace2/tvtime/speedy.c:
	  * gst/deinterlace2/tvtime/speedy.h:
	  * gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy):
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir):
	  Get rid of speedy.[ch] as we don't use most of it's code anyway
	  and it doesn't seem to be relicensed to LGPL. Use memcpy() instead
	  of the speedy memcpy everywhere instead.
	  * gst/deinterlace2/gstdeinterlace2.h:
	  Remove many unused declarations.

2008-06-28 18:13:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 18/56] gst/deinterlace2/gstdeinterlace2.c: Divide latency be 2 to convert from fields to frames.
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_src_query):
	  Divide latency be 2 to convert from fields to frames.

2008-06-28 18:10:52 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 17/56] gst/deinterlace2/tvtime/greedy.c: Don't use scanlines function from gstdeinterlace2 as it's not appropiate for this m...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (deinterlace_greedy_packed422_scanline_c),
	  (deinterlace_greedy_packed422_scanline_mmx),
	  (deinterlace_greedy_packed422_scanline_mmxext),
	  (deinterlace_frame_di_greedy):
	  Don't use scanlines function from gstdeinterlace2 as it's
	  not appropiate for this method. Instead implement deinterlace_frame
	  function by taking the one from greedyh.
	  * gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C):
	  Small fix for the C implementation.
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir):
	  Don't use the scanlines function from gstdeinterlace2 as it's only
	  used for this method and will be removed. Instead implement
	  deinterlace_frame function and make it a bit more efficient.
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace2_class_init), (gst_deinterlace2_set_method),
	  (gst_deinterlace2_push_history), (gst_deinterlace2_chain),
	  (gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event),
	  (gst_deinterlace2_change_state), (gst_deinterlace2_src_event),
	  (gst_deinterlace2_src_query):
	  Fix coding style and remove scanlines function as it's unused now.

2008-06-28 17:25:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 16/56] gst/deinterlace2/tvtime/: Add a C implementation for the greedyh deinterlacing method, clean up the code a bit and ma...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  * gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C),
	  (deinterlace_frame_di_greedyh), (dscaler_greedyh_get_method):
	  * gst/deinterlace2/tvtime/greedyhmacros.h:
	  Add a C implementation for the greedyh deinterlacing method, clean
	  up the code a bit and mark the SSE version as MMXEXT as it doesn't
	  require any SSE instructions.

2008-06-27 13:22:34 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 15/56] gst/deinterlace2/gstdeinterlace2.c: If we're outputting all fields the framerate has to be doubled.
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace2_set_property), (gst_deinterlace2_chain),
	  (gst_deinterlace2_setcaps):
	  If we're outputting all fields the framerate has to be doubled.
	  Set duration on the outgoing buffers.

2008-06-25 16:05:08 +0000  Edward Hervey <bilboed@bilboed.com>

	  [MOVED FROM BAD 14/56] gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h: Remove unneeded macros that break build on macosx.
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
	  Remove unneeded macros that break build on macosx.

2008-06-24 12:08:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 13/56] gst/deinterlace2/tvtime/greedy.c: Optimize MMX/MMXEXT implementations a bit by requiring two less memory accesses and...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (deinterlace_greedy_packed422_scanline_mmx),
	  (deinterlace_greedy_packed422_scanline_mmxext):
	  Optimize MMX/MMXEXT implementations a bit by requiring two less
	  memory accesses and fix the workaround for the missing right shift
	  on bytes to unset the highest bit of every byte.

2008-06-24 10:15:41 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 12/56] gst/deinterlace2/tvtime/greedy.c: Remove sfence instruction as it's not needed and actually is an SSE instruction.
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (deinterlace_greedy_packed422_scanline_mmxext):
	  Remove sfence instruction as it's not needed and actually is an SSE
	  instruction.

2008-06-24 10:12:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 11/56] gst/deinterlace2/tvtime/greedy.c: Add plain MMX implementation for the greedyl method.
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (deinterlace_greedy_packed422_scanline_mmx),
	  (deinterlace_greedy_packed422_scanline):
	  Add plain MMX implementation for the greedyl method.

2008-06-24 09:40:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 10/56] gst/deinterlace2/Makefile.am: Move the assembly includes to noinst_HEADERS where they belong.
	  Original commit message from CVS:
	  * gst/deinterlace2/Makefile.am:
	  Move the assembly includes to noinst_HEADERS where they belong.
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c),
	  (deinterlace_line_mmx):
	  Fix C and MMX implementations a bit more.

2008-06-24 09:10:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 09/56] gst/deinterlace2/tvtime/greedy.c: Fix the C implementation to produce correct results and optimize the
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (deinterlace_greedy_packed422_scanline_c),
	  (deinterlace_greedy_packed422_scanline_mmxext),
	  (deinterlace_greedy_packed422_scanline):
	  Fix the C implementation to produce correct results and optimize the
	  MMXEXT implementation.
	  Handle odd widths and don't read over array boundaries in the MMXEXT
	  implementation.
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c),
	  (deinterlace_line_mmx), (deinterlace_scanline_vfir):
	  Fix a small rounding bug in the MMX implementation, the MMX
	  implementation doesn't actually need MMXEXT instructions so don't mark
	  it as such.
	  Handle odd widths in both implementations.

2008-06-21 09:05:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 08/56] gst/deinterlace2/tvtime/greedy.c: Implement a C version of the greedy low motion algorithm and mark the assembly opti...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/greedy.c:
	  (deinterlace_greedy_packed422_scanline_sse),
	  (deinterlace_greedy_packed422_scanline_c),
	  (deinterlace_greedy_packed422_scanline):
	  Implement a C version of the greedy low motion algorithm and mark the
	  assembly optimized version as SSE as it uses SSE instructions
	  additional to MMX instructions.

2008-06-20 14:48:40 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 07/56] gst/deinterlace2/tvtime/vfir.c: Make it possible to use the vfir method on X86 CPUs without MMXEXT too but use the MM...
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_mmxext),
	  (deinterlace_line_c), (deinterlace_scanline_vfir):
	  Make it possible to use the vfir method on X86 CPUs without MMXEXT too
	  but use the MMXEXT optimized code whenever possible.

2008-06-20 14:35:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 06/56] gst/deinterlace2/gstdeinterlace2.*: Reset element state on PAUSED->READY properly, don't leak any buffers when finali...
	  Original commit message from CVS:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace2_class_init), (gst_deinterlace2_init),
	  (gst_deinterlace2_reset_history), (gst_deinterlace2_reset),
	  (gst_deinterlace2_finalize), (gst_deinterlace2_chain),
	  (gst_deinterlace2_sink_event), (gst_deinterlace2_change_state),
	  (gst_deinterlace2_src_query):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  Reset element state on PAUSED->READY properly, don't leak any buffers
	  when finalizing, allocate buffers with gst_pad_alloc_buffer() and
	  properly return flow returns from gst_pad_push() instead of ignoring them.

2008-06-20 13:45:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 05/56] gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h: Add missing header.
	  Original commit message from CVS:
	  * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
	  Add missing header.

2008-06-20 13:24:29 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 04/56] Fix compilation on generic x86/amd64 and include deinterlace2 in the build system. Because of several bugs it's still...
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
	  Fix compilation on generic x86/amd64 and include deinterlace2 in the
	  build system. Because of several bugs it's still enabled only
	  by --enable-experimental.

2008-06-18 06:31:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD 03/56] Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * examples/app/appsrc-ra.c:
	  * examples/app/appsrc-seekable.c:
	  * examples/app/appsrc-stream.c:
	  * examples/app/appsrc-stream2.c:
	  * ext/directfb/dfbvideosink.h:
	  * ext/metadata/gstbasemetadata.c:
	  * ext/metadata/gstbasemetadata.h:
	  * ext/metadata/metadata.c:
	  * ext/metadata/metadataexif.c:
	  * ext/theora/theoradec.h:
	  * gst/deinterlace2/gstdeinterlace2.h:
	  * gst/deinterlace2/tvtime/speedy.c:
	  * gst/deinterlace2/tvtime/speedy.h:
	  * gst/deinterlace2/tvtime/vfir.c:
	  Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
	  comments.

2008-06-11 11:12:49 +0000  Martin Eikermann <meiker@upb.de>

	  [MOVED FROM BAD 02/56] gst/deinterlace2/: Add a deinterlacer plugin based on the tvtime/DScaler deinterlacer, which was relicensed to LGPL f...
	  Original commit message from CVS:
	  Based on a patch by: Martin Eikermann <meiker at upb dot de>
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace2_method_get_type),
	  (gst_deinterlace2_fields_get_type),
	  (gst_deinterlace2_field_layout_get_type),
	  (gst_deinterlace2_base_init), (gst_deinterlace2_class_init),
	  (gst_deinterlace2_init), (gst_deinterlace2_set_method),
	  (gst_deinterlace2_set_property), (gst_deinterlace2_get_property),
	  (gst_deinterlace2_finalize), (gst_deinterlace2_pop_history),
	  (gst_deinterlace2_head_history), (gst_deinterlace2_push_history),
	  (gst_deinterlace2_deinterlace_scanlines), (gst_deinterlace2_chain),
	  (gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event),
	  (gst_deinterlace2_change_state), (gst_deinterlace2_src_event),
	  (gst_deinterlace2_src_query), (gst_deinterlace2_src_query_types),
	  (plugin_init):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  * gst/deinterlace2/tvtime/greedy.c: (copy_scanline),
	  (deinterlace_greedy_packed422_scanline_mmxext),
	  (dscaler_greedyl_get_method):
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  * gst/deinterlace2/tvtime/greedyh.c:
	  (deinterlace_frame_di_greedyh), (dscaler_greedyh_get_method),
	  (greedyh_init), (greedyh_filter_mmx), (greedyh_filter_3dnow),
	  (greedyh_filter_sse):
	  * gst/deinterlace2/tvtime/greedyh.h:
	  * gst/deinterlace2/tvtime/greedyhmacros.h:
	  * gst/deinterlace2/tvtime/mmx.h:
	  * gst/deinterlace2/tvtime/plugins.h:
	  * gst/deinterlace2/tvtime/speedtools.h:
	  * gst/deinterlace2/tvtime/speedy.c: (multiply_alpha), (clip255),
	  (comb_factor_packed422_scanline_mmx),
	  (diff_factor_packed422_scanline_c),
	  (diff_factor_packed422_scanline_mmx),
	  (diff_packed422_block8x8_mmx), (diff_packed422_block8x8_c),
	  (packed444_to_packed422_scanline_c),
	  (packed422_to_packed444_scanline_c),
	  (packed422_to_packed444_rec601_scanline_c),
	  (vfilter_chroma_121_packed422_scanline_mmx),
	  (vfilter_chroma_121_packed422_scanline_c),
	  (vfilter_chroma_332_packed422_scanline_mmx),
	  (vfilter_chroma_332_packed422_scanline_c),
	  (kill_chroma_packed422_inplace_scanline_mmx),
	  (kill_chroma_packed422_inplace_scanline_c),
	  (invert_colour_packed422_inplace_scanline_mmx),
	  (invert_colour_packed422_inplace_scanline_c),
	  (mirror_packed422_inplace_scanline_c),
	  (interpolate_packed422_scanline_c),
	  (convert_uyvy_to_yuyv_scanline_mmx),
	  (convert_uyvy_to_yuyv_scanline_c),
	  (interpolate_packed422_scanline_mmx),
	  (interpolate_packed422_scanline_mmxext),
	  (blit_colour_packed422_scanline_c),
	  (blit_colour_packed422_scanline_mmx),
	  (blit_colour_packed422_scanline_mmxext),
	  (blit_colour_packed4444_scanline_c),
	  (blit_colour_packed4444_scanline_mmx),
	  (blit_colour_packed4444_scanline_mmxext), (small_memcpy),
	  (speedy_memcpy_c), (speedy_memcpy_mmx), (speedy_memcpy_mmxext),
	  (blit_packed422_scanline_c), (blit_packed422_scanline_mmx),
	  (blit_packed422_scanline_mmxext),
	  (composite_colour4444_alpha_to_packed422_scanline_c),
	  (composite_colour4444_alpha_to_packed422_scanline_mmxext),
	  (composite_packed4444_alpha_to_packed422_scanline_c),
	  (composite_packed4444_alpha_to_packed422_scanline_mmxext),
	  (composite_packed4444_to_packed422_scanline_c),
	  (composite_packed4444_to_packed422_scanline_mmxext),
	  (composite_alphamask_to_packed4444_scanline_c),
	  (composite_alphamask_to_packed4444_scanline_mmxext),
	  (composite_alphamask_alpha_to_packed4444_scanline_c),
	  (premultiply_packed4444_scanline_c),
	  (premultiply_packed4444_scanline_mmxext),
	  (blend_packed422_scanline_c), (blend_packed422_scanline_mmxext),
	  (quarter_blit_vertical_packed422_scanline_mmxext),
	  (quarter_blit_vertical_packed422_scanline_c),
	  (subpix_blit_vertical_packed422_scanline_c),
	  (a8_subpix_blit_scanline_c), (myround), (init_RGB_to_YCbCr_tables),
	  (init_YCbCr_to_RGB_tables), (rgb24_to_packed444_rec601_scanline_c),
	  (rgba32_to_packed4444_rec601_scanline_c),
	  (packed444_to_rgb24_rec601_scanline_c),
	  (packed444_to_nonpremultiplied_packed4444_scanline_c),
	  (aspect_adjust_packed4444_scanline_c), (setup_speedy_calls),
	  (speedy_get_accel):
	  * gst/deinterlace2/tvtime/speedy.h:
	  * gst/deinterlace2/tvtime/sse.h:
	  * gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy),
	  (deinterlace_frame_di_tomsmocomp), (dscaler_tomsmocomp_get_method),
	  (tomsmocomp_init), (tomsmocomp_filter_mmx),
	  (tomsmocomp_filter_3dnow), (tomsmocomp_filter_sse):
	  * gst/deinterlace2/tvtime/tomsmocomp.h:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoop0A.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA8.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA2.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA6.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH2.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVA.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVAH.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line),
	  (deinterlace_scanline_vfir), (copy_scanline),
	  (dscaler_vfir_get_method):
	  * gst/deinterlace2/tvtime/x86-64_macros.inc:
	  Add a deinterlacer plugin based on the tvtime/DScaler deinterlacer,
	  which was relicensed to LGPL for GStreamer and in theory provides
	  better and faster results than the simple deinterlace element.
	  Fixes bug #163578.
	  Ported to GStreamer 0.10 but still not enabled or included in the
	  build system by default because of bad artefacts caused by a bug
	  somewhere and as it can be only build on x86/amd64 ATM and requires
	  special CFLAGS. Will be fixed soon.

2008-06-11 11:12:14 +0000  Martin Eikermann <meiker@upb.de>

	  [MOVED FROM BAD 01/56] gst/deinterlace2/: Add a deinterlacer plugin based on the tvtime/DScaler deinterlacer, which was relicensed to LGPL f...
	  Original commit message from CVS:
	  Based on a patch by: Martin Eikermann <meiker at upb dot de>
	  * gst/deinterlace2/Makefile.am:
	  * gst/deinterlace2/gstdeinterlace2.c:
	  (gst_deinterlace2_method_get_type),
	  (gst_deinterlace2_fields_get_type),
	  (gst_deinterlace2_field_layout_get_type),
	  (gst_deinterlace2_base_init), (gst_deinterlace2_class_init),
	  (gst_deinterlace2_init), (gst_deinterlace2_set_method),
	  (gst_deinterlace2_set_property), (gst_deinterlace2_get_property),
	  (gst_deinterlace2_finalize), (gst_deinterlace2_pop_history),
	  (gst_deinterlace2_head_history), (gst_deinterlace2_push_history),
	  (gst_deinterlace2_deinterlace_scanlines), (gst_deinterlace2_chain),
	  (gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event),
	  (gst_deinterlace2_change_state), (gst_deinterlace2_src_event),
	  (gst_deinterlace2_src_query), (gst_deinterlace2_src_query_types),
	  (plugin_init):
	  * gst/deinterlace2/gstdeinterlace2.h:
	  * gst/deinterlace2/tvtime/greedy.c: (copy_scanline),
	  (deinterlace_greedy_packed422_scanline_mmxext),
	  (dscaler_greedyl_get_method):
	  * gst/deinterlace2/tvtime/greedyh.asm:
	  * gst/deinterlace2/tvtime/greedyh.c:
	  (deinterlace_frame_di_greedyh), (dscaler_greedyh_get_method),
	  (greedyh_init), (greedyh_filter_mmx), (greedyh_filter_3dnow),
	  (greedyh_filter_sse):
	  * gst/deinterlace2/tvtime/greedyh.h:
	  * gst/deinterlace2/tvtime/greedyhmacros.h:
	  * gst/deinterlace2/tvtime/mmx.h:
	  * gst/deinterlace2/tvtime/plugins.h:
	  * gst/deinterlace2/tvtime/speedtools.h:
	  * gst/deinterlace2/tvtime/speedy.c: (multiply_alpha), (clip255),
	  (comb_factor_packed422_scanline_mmx),
	  (diff_factor_packed422_scanline_c),
	  (diff_factor_packed422_scanline_mmx),
	  (diff_packed422_block8x8_mmx), (diff_packed422_block8x8_c),
	  (packed444_to_packed422_scanline_c),
	  (packed422_to_packed444_scanline_c),
	  (packed422_to_packed444_rec601_scanline_c),
	  (vfilter_chroma_121_packed422_scanline_mmx),
	  (vfilter_chroma_121_packed422_scanline_c),
	  (vfilter_chroma_332_packed422_scanline_mmx),
	  (vfilter_chroma_332_packed422_scanline_c),
	  (kill_chroma_packed422_inplace_scanline_mmx),
	  (kill_chroma_packed422_inplace_scanline_c),
	  (invert_colour_packed422_inplace_scanline_mmx),
	  (invert_colour_packed422_inplace_scanline_c),
	  (mirror_packed422_inplace_scanline_c),
	  (interpolate_packed422_scanline_c),
	  (convert_uyvy_to_yuyv_scanline_mmx),
	  (convert_uyvy_to_yuyv_scanline_c),
	  (interpolate_packed422_scanline_mmx),
	  (interpolate_packed422_scanline_mmxext),
	  (blit_colour_packed422_scanline_c),
	  (blit_colour_packed422_scanline_mmx),
	  (blit_colour_packed422_scanline_mmxext),
	  (blit_colour_packed4444_scanline_c),
	  (blit_colour_packed4444_scanline_mmx),
	  (blit_colour_packed4444_scanline_mmxext), (small_memcpy),
	  (speedy_memcpy_c), (speedy_memcpy_mmx), (speedy_memcpy_mmxext),
	  (blit_packed422_scanline_c), (blit_packed422_scanline_mmx),
	  (blit_packed422_scanline_mmxext),
	  (composite_colour4444_alpha_to_packed422_scanline_c),
	  (composite_colour4444_alpha_to_packed422_scanline_mmxext),
	  (composite_packed4444_alpha_to_packed422_scanline_c),
	  (composite_packed4444_alpha_to_packed422_scanline_mmxext),
	  (composite_packed4444_to_packed422_scanline_c),
	  (composite_packed4444_to_packed422_scanline_mmxext),
	  (composite_alphamask_to_packed4444_scanline_c),
	  (composite_alphamask_to_packed4444_scanline_mmxext),
	  (composite_alphamask_alpha_to_packed4444_scanline_c),
	  (premultiply_packed4444_scanline_c),
	  (premultiply_packed4444_scanline_mmxext),
	  (blend_packed422_scanline_c), (blend_packed422_scanline_mmxext),
	  (quarter_blit_vertical_packed422_scanline_mmxext),
	  (quarter_blit_vertical_packed422_scanline_c),
	  (subpix_blit_vertical_packed422_scanline_c),
	  (a8_subpix_blit_scanline_c), (myround), (init_RGB_to_YCbCr_tables),
	  (init_YCbCr_to_RGB_tables), (rgb24_to_packed444_rec601_scanline_c),
	  (rgba32_to_packed4444_rec601_scanline_c),
	  (packed444_to_rgb24_rec601_scanline_c),
	  (packed444_to_nonpremultiplied_packed4444_scanline_c),
	  (aspect_adjust_packed4444_scanline_c), (setup_speedy_calls),
	  (speedy_get_accel):
	  * gst/deinterlace2/tvtime/speedy.h:
	  * gst/deinterlace2/tvtime/sse.h:
	  * gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy),
	  (deinterlace_frame_di_tomsmocomp), (dscaler_tomsmocomp_get_method),
	  (tomsmocomp_init), (tomsmocomp_filter_mmx),
	  (tomsmocomp_filter_3dnow), (tomsmocomp_filter_sse):
	  * gst/deinterlace2/tvtime/tomsmocomp.h:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoop0A.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA8.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA2.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA6.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH2.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVA.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVAH.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
	  * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
	  * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line),
	  (deinterlace_scanline_vfir), (copy_scanline),
	  (dscaler_vfir_get_method):
	  * gst/deinterlace2/tvtime/x86-64_macros.inc:
	  Add a deinterlacer plugin based on the tvtime/DScaler deinterlacer,
	  which was relicensed to LGPL for GStreamer and in theory provides
	  better and faster results than the simple deinterlace element.
	  Fixes bug #163578.
	  Ported to GStreamer 0.10 but still not enabled or included in the
	  build system by default because of bad artefacts caused by a bug
	  somewhere and as it can be only build on x86/amd64 ATM and requires
	  special CFLAGS. Will be fixed soon.

2009-05-13 10:30:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  flv: Actually add the flv plugin to configure.ac

2009-05-13 09:24:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/pipelines/flacdec.c:
	  checks: fix flacdec unit tests on big-endian machines and under valgrind
	  Flacdec outputs 16-bit samples, so let's check if the value of the first
	  sample is what we expect rather than just the first byte, which may be
	  different from what we expect depending on the host's endianness. Fixes
	  the flacdec unit tests on PPC. Also fix a bunch of leaks in the unit
	  tests to make valgrind happy. Fixes #582420.

2009-05-13 09:18:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix buffer leak
	  gst_buffer_replace() will take its own ref, so we still have
	  to unref the buffer if we don't need it any longer.

2009-05-12 21:20:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix pointer arithmetic
	  This fixes a seeking regression, bug #134522.

2009-05-12 19:22:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: add Since tag to gtk-doc chunk

2009-05-12 21:36:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Moved 'flv' from -bad to -good

2009-05-07 17:53:42 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  [MOVED FROM BAD 57/57] Add ranks to various muxers and encoders in -bad

2009-04-29 18:52:20 +0100  Tristan Matthews <le.businessman@gmail.com>

	* gst/flv/gstflvmux.c:
	  [MOVED FROM BAD 56/57] flvmux: init variable to NULL to fix compiler warning
	  Fixes #580786.

2009-04-29 13:56:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvparse.c:
	  [MOVED FROM BAD 55/57] flv: Set/require the framed/parsed fields of the audio/mpeg caps to TRUE

2009-04-29 13:16:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  [MOVED FROM BAD 54/57] flv: Always write at least the minimal tags and write the PAR as tags

2009-04-29 13:03:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  [MOVED FROM BAD 53/57] flv: Add support for muxing some tags

2009-04-29 13:03:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  [MOVED FROM BAD 52/57] flv: Add support for title tag

2009-04-29 09:40:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvparse.c:
	  [MOVED FROM BAD 51/57] flv: Fix parsing of tags and add new mappings
	  We shouldn't register a new GstTag for every unknown tag
	  we find as this might lead to conflicts and also those
	  tags are essentially unknown.
	  Add mappings for some known tags and also convert string
	  dates to GDate, as found in many FLV files.

2009-04-22 19:52:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  [MOVED FROM BAD 50/57] flv: Add documentation to flvmux and flvdemux
	  Partially fixes bug #573737.

2009-01-22 13:39:34 +0100  Jan Urbanski <j.urbanski@students.mimuw.edu.pl>

	* gst/flv/gstflvparse.c:
	  [MOVED FROM BAD 49/57] Add support for ECMA arrays in script tags. Fixes bug #567965.
	  Add support for ECMA arrays in script tags. This fixes
	  seeking on some files that have the seek table stored
	  inside an ECMA array instead of the normal array.

2008-12-03 11:43:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 48/57] gst/flv/gstflvparse.c: Check if strings are valid UTF8 before using them.
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (FLV_GET_STRING):
	  Check if strings are valid UTF8 before using them.

2008-11-24 11:17:19 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 47/57] gst/flv/gstflvdemux.c: Fix non key unit seeking by always going to the previous keyframe. Mark the discont flag when ...
	  Original commit message from CVS:
	  2008-11-24  Julien Moutte  <julien@fluendo.com>
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_find_offset),
	  (gst_flv_demux_handle_seek_push),
	  (gst_flv_demux_handle_seek_pull):
	  Fix non key unit seeking by always going to the previous
	  keyframe. Mark
	  the discont flag when we've moved in the file.
	  * gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate): MP3
	  streams
	  are parsed already, makes autoplugged pipelines shorter.

2008-11-04 12:42:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD 46/57] Don't install static libs for plugins. Fixes #550851 for -bad.
	  Original commit message from CVS:
	  * ext/alsaspdif/Makefile.am:
	  * ext/amrwb/Makefile.am:
	  * ext/apexsink/Makefile.am:
	  * ext/arts/Makefile.am:
	  * ext/artsd/Makefile.am:
	  * ext/audiofile/Makefile.am:
	  * ext/audioresample/Makefile.am:
	  * ext/bz2/Makefile.am:
	  * ext/cdaudio/Makefile.am:
	  * ext/celt/Makefile.am:
	  * ext/dc1394/Makefile.am:
	  * ext/dirac/Makefile.am:
	  * ext/directfb/Makefile.am:
	  * ext/divx/Makefile.am:
	  * ext/dts/Makefile.am:
	  * ext/faac/Makefile.am:
	  * ext/faad/Makefile.am:
	  * ext/gsm/Makefile.am:
	  * ext/hermes/Makefile.am:
	  * ext/ivorbis/Makefile.am:
	  * ext/jack/Makefile.am:
	  * ext/jp2k/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/lcs/Makefile.am:
	  * ext/libfame/Makefile.am:
	  * ext/libmms/Makefile.am:
	  * ext/metadata/Makefile.am:
	  * ext/mpeg2enc/Makefile.am:
	  * ext/mplex/Makefile.am:
	  * ext/musepack/Makefile.am:
	  * ext/musicbrainz/Makefile.am:
	  * ext/mythtv/Makefile.am:
	  * ext/nas/Makefile.am:
	  * ext/neon/Makefile.am:
	  * ext/ofa/Makefile.am:
	  * ext/polyp/Makefile.am:
	  * ext/resindvd/Makefile.am:
	  * ext/sdl/Makefile.am:
	  * ext/shout/Makefile.am:
	  * ext/snapshot/Makefile.am:
	  * ext/sndfile/Makefile.am:
	  * ext/soundtouch/Makefile.am:
	  * ext/spc/Makefile.am:
	  * ext/swfdec/Makefile.am:
	  * ext/tarkin/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/timidity/Makefile.am:
	  * ext/twolame/Makefile.am:
	  * ext/x264/Makefile.am:
	  * ext/xine/Makefile.am:
	  * ext/xvid/Makefile.am:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/dshow/Makefile.am:
	  * gst/aiffparse/Makefile.am:
	  * gst/app/Makefile.am:
	  * gst/audiobuffer/Makefile.am:
	  * gst/bayer/Makefile.am:
	  * gst/cdxaparse/Makefile.am:
	  * gst/chart/Makefile.am:
	  * gst/colorspace/Makefile.am:
	  * gst/dccp/Makefile.am:
	  * gst/deinterlace/Makefile.am:
	  * gst/deinterlace2/Makefile.am:
	  * gst/dvdspu/Makefile.am:
	  * gst/festival/Makefile.am:
	  * gst/filter/Makefile.am:
	  * gst/flacparse/Makefile.am:
	  * gst/flv/Makefile.am:
	  * gst/games/Makefile.am:
	  * gst/h264parse/Makefile.am:
	  * gst/librfb/Makefile.am:
	  * gst/mixmatrix/Makefile.am:
	  * gst/modplug/Makefile.am:
	  * gst/mpeg1sys/Makefile.am:
	  * gst/mpeg4videoparse/Makefile.am:
	  * gst/mpegdemux/Makefile.am:
	  * gst/mpegtsmux/Makefile.am:
	  * gst/mpegvideoparse/Makefile.am:
	  * gst/mve/Makefile.am:
	  * gst/nsf/Makefile.am:
	  * gst/nuvdemux/Makefile.am:
	  * gst/overlay/Makefile.am:
	  * gst/passthrough/Makefile.am:
	  * gst/pcapparse/Makefile.am:
	  * gst/playondemand/Makefile.am:
	  * gst/rawparse/Makefile.am:
	  * gst/real/Makefile.am:
	  * gst/rtjpeg/Makefile.am:
	  * gst/rtpmanager/Makefile.am:
	  * gst/scaletempo/Makefile.am:
	  * gst/sdp/Makefile.am:
	  * gst/selector/Makefile.am:
	  * gst/smooth/Makefile.am:
	  * gst/smoothwave/Makefile.am:
	  * gst/speed/Makefile.am:
	  * gst/speexresample/Makefile.am:
	  * gst/stereo/Makefile.am:
	  * gst/subenc/Makefile.am:
	  * gst/tta/Makefile.am:
	  * gst/vbidec/Makefile.am:
	  * gst/videodrop/Makefile.am:
	  * gst/videosignal/Makefile.am:
	  * gst/virtualdub/Makefile.am:
	  * gst/vmnc/Makefile.am:
	  * gst/y4m/Makefile.am:
	  * sys/acmenc/Makefile.am:
	  * sys/cdrom/Makefile.am:
	  * sys/dshowdecwrapper/Makefile.am:
	  * sys/dshowsrcwrapper/Makefile.am:
	  * sys/dvb/Makefile.am:
	  * sys/dxr3/Makefile.am:
	  * sys/fbdev/Makefile.am:
	  * sys/oss4/Makefile.am:
	  * sys/qcam/Makefile.am:
	  * sys/qtwrapper/Makefile.am:
	  * sys/vcd/Makefile.am:
	  * sys/wininet/Makefile.am:
	  * win32/common/config.h:
	  Don't install static libs for plugins. Fixes #550851 for -bad.

2008-10-28 18:44:44 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 45/57] gst/flv/gstflvdemux.c: Implement position query in time format.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_query):
	  Implement position query in time format.

2008-10-28 18:41:19 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 44/57] gst/flv/: Put the GstSegment directly into the instance struct instead of allocating and free'ing it again.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
	  (gst_flv_demux_loop), (gst_flv_demux_handle_seek_push),
	  (gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event),
	  (gst_flv_demux_dispose), (gst_flv_demux_init):
	  * gst/flv/gstflvdemux.h:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video), (gst_flv_parse_tag_timestamp):
	  Put the GstSegment directly into the instance struct instead of
	  allocating and free'ing it again.
	  Push tags already if only one pad was added, no need to wait for
	  the second one.
	  When generating our index set has_video and has_audio if we find
	  video or audio in case the FLV header has incorrect data.

2008-10-27 09:45:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 43/57] gst/flv/: Don't memcpy() all data we want to push downstream, instead just create subbuffers and push them downstream.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_chain),
	  (gst_flv_demux_pull_tag), (gst_flv_demux_pull_header),
	  (gst_flv_demux_create_index):
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_script),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
	  (gst_flv_parse_tag_timestamp), (gst_flv_parse_tag_type),
	  (gst_flv_parse_header):
	  * gst/flv/gstflvparse.h:
	  Don't memcpy() all data we want to push downstream, instead just
	  create subbuffers and push them downstream.
	  Fix some minor memory leaks.

2008-10-27 09:41:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 42/57] gst/flv/Makefile.am: Fix (non-critical) syntax error and add all required CFLAGS and LIBS.
	  Original commit message from CVS:
	  * gst/flv/Makefile.am:
	  Fix (non-critical) syntax error and add all required CFLAGS and LIBS.
	  * gst/flv/gstflvparse.c: (FLV_GET_STRING),
	  (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
	  (gst_flv_parse_tag_timestamp), (gst_flv_parse_tag_type):
	  Rewrite the script tag parsing to make sure we don't try to read
	  more data than we have. Also use GST_READ_UINT24_BE directly and
	  fix some minor memory leaks.
	  This should make all crashes on fuzzed FLV files disappear.

2008-10-27 09:37:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 41/57] gst/flv/gstflvparse.c: Properly check everywhere that we have enough data to parse and don't read outside the allocat...
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (FLV_GET_STRING),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
	  (gst_flv_parse_tag_type), (gst_flv_parse_header):
	  Properly check everywhere that we have enough data to parse and
	  don't read outside the allocated memory region.

2008-10-27 09:35:34 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 40/57] gst/flv/gstflvparse.c: If the caps change during playback and negotiation fails error out instead of trying to continue.
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video):
	  If the caps change during playback and negotiation fails error out
	  instead of trying to continue.

2008-10-27 09:33:40 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 39/57] gst/flv/: Add support for Speex audio and allow buffers without valid timestamp in the muxer.
	  Original commit message from CVS:
	  * gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps),
	  (gst_flv_mux_request_new_pad), (gst_flv_mux_write_buffer),
	  (gst_flv_mux_collected):
	  * gst/flv/gstflvmux.h:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate):
	  Add support for Speex audio and allow buffers without valid
	  timestamp in the muxer.

2008-10-27 09:32:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 38/57] gst/flv/gstflvdemux.c: Don't post an error message on the bus if sending EOS downstream didn't work. Fixes bug #550454.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_loop),
	  (gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push),
	  (gst_flv_demux_handle_seek_pull):
	  Don't post an error message on the bus if sending EOS downstream
	  didn't work. Fixes bug #550454.
	  Fix seek event handling to look at the flags of the seek event
	  instead of assuming some random flags, don't send segment-start
	  messages when operating in push mode and push seek events upstream
	  if we couldn't handle them.

2008-10-27 09:27:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 37/57] gst/flv/gstflvdemux.c: Error out early if pulling a tag failed.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag):
	  Error out early if pulling a tag failed.

2008-10-27 09:25:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 36/57] gst/flv/: In pull mode we create our own index before doing anything else and don't use the index provided by some fi...
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_create_index),
	  (gst_flv_demux_loop):
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_script),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
	  (gst_flv_parse_tag_timestamp):
	  * gst/flv/gstflvparse.h:
	  In pull mode we create our own index before doing anything else
	  and don't use the index provided by some files (which are more than
	  often incorrect and cause failed seeks).
	  For push mode we still use the index provided by the file and extend it
	  while doing the playback.

2008-10-27 09:20:01 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 35/57] gst/flv/gstflvdemux.c: Instead of using gst_pad_event_default() use a small gst_pad_push_event() wrapper that only do...
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_push_src_event),
	  (gst_flv_demux_loop), (gst_flv_demux_handle_seek_pull),
	  (gst_flv_demux_sink_event):
	  Instead of using gst_pad_event_default() use a small
	  gst_pad_push_event() wrapper that only does what we want and is much
	  more simple.

2008-10-27 09:14:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 34/57] gst/flv/gstflvdemux.*: If our index was created by the element and not provided from the outside we should destroy it...
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_change_state),
	  (gst_flv_demux_set_index), (gst_flv_demux_init):
	  * gst/flv/gstflvdemux.h:
	  If our index was created by the element and not provided from the
	  outside we should destroy it when starting a new stream to get
	  all old entries removed.

2008-10-27 09:12:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 33/57] gst/flv/gstflvdemux.c: Improve debugging a bit when pulling a buffer from upstream fails.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range):
	  Improve debugging a bit when pulling a buffer from upstream fails.

2008-10-27 09:10:54 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 32/57] gst/flv/: Close the currently playing segment from the streaming thread instead of the thread where the seek event is...
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
	  (gst_flv_demux_handle_seek_pull), (gst_flv_demux_dispose):
	  * gst/flv/gstflvdemux.h:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video):
	  Close the currently playing segment from the streaming thread
	  instead of the thread where the seek event is handled.

2008-10-16 15:21:15 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 31/57] gst/flv/gstflvmux.c: Don't set video_codec to the value that actually should go into audio codec, otherwise we create...
	  Original commit message from CVS:
	  * gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps),
	  (gst_flv_mux_write_buffer):
	  Don't set video_codec to the value that actually should go
	  into audio codec, otherwise we create invalid files.
	  Fixes bug #556564.

2008-10-12 17:08:10 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 30/57] gst/flv/gstflvdemux.c: Fix regression of handling flow returns in pull mode.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag),
	  (gst_flv_demux_pull_header):
	  Fix regression of handling flow returns in pull mode.
	  Fixes bug #556003.

2008-10-10 16:33:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 29/57] gst/flv/gstflvparse.c: Use gst_pad_alloc_buffer_and_set_caps() to make sure we get a buffer with caps that we can wor...
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video):
	  Use gst_pad_alloc_buffer_and_set_caps() to make sure we get
	  a buffer with caps that we can work with (i.e. the pad's caps).
	  Add non-keyframe video frames to the index too but without the
	  keyframe flag.
	  Add audio frames to the index only if we have no video stream.

2008-10-10 16:15:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 28/57] gst/flv/gstflvparse.c: Create pads from the pad templates, use fixed caps on them and only activate them after the ca...
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video):
	  Create pads from the pad templates, use fixed caps on them
	  and only activate them after the caps are set.

2008-10-09 16:20:26 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 27/57] gst/flv/: Get an approximate duration of the file by looking at the timestamp of the last tag in pull mode. If we get...
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_loop):
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_timestamp):
	  * gst/flv/gstflvparse.h:
	  Get an approximate duration of the file by looking at the timestamp
	  of the last tag in pull mode. If we get (maybe better) duration from
	  metadata later we'll use that instead.

2008-10-09 15:43:02 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 26/57] gst/flv/gstflvdemux.c: Refactor _pull_range() logic with checks into a seperate function to make things a bit more re...
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range),
	  (gst_flv_demux_pull_tag), (gst_flv_demux_pull_header):
	  Refactor _pull_range() logic with checks into a seperate function
	  to make things a bit more readable.

2008-10-09 15:26:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 25/57] gst/flv/gstflvdemux.c: Use gst_element_class_set_details_simple().
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_chain),
	  (gst_flv_demux_base_init):
	  Use gst_element_class_set_details_simple().
	  If we get GST_FLOW_NOT_LINKED in the parse loop but at least
	  one of the pads is linked continue the loop.

2008-10-09 10:00:51 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 24/57] gst/flv/gstflvparse.c: Correct caps for video codec id 5: It's On2 VP6 with alpha channel which needs a different dec...
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_video_negotiate):
	  Correct caps for video codec id 5: It's On2 VP6 with alpha channel
	  which needs a different decoder and has different caps.
	  Add support for audio codec id 14, which is MP3 with 8kHz sampling
	  rate.
	  Fix endianness and signedness for raw audio codec ids.
	  Add support for alaw and mulaw audio.

2008-10-09 09:48:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 23/57] gst/flv/gstflvdemux.c: Go out of the parse loop as soon as we get an error instead of parsing until the GstAdapter is...
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_chain):
	  Go out of the parse loop as soon as we get an error instead
	  of parsing until the GstAdapter is empty.
	  Add some explanations about the header and tag size.
	  Don't print synchronizing message if everything is fine.

2008-10-09 09:26:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD 22/57] gst/flv/: Add first version of a FLV muxer. The only missing feature is writing of stream metadata.
	  Original commit message from CVS:
	  * gst/flv/Makefile.am:
	  * gst/flv/gstflvdemux.c: (plugin_init):
	  * gst/flv/gstflvmux.c: (gst_flv_mux_base_init),
	  (gst_flv_mux_class_init), (gst_flv_mux_init),
	  (gst_flv_mux_finalize), (gst_flv_mux_reset),
	  (gst_flv_mux_handle_src_event), (gst_flv_mux_handle_sink_event),
	  (gst_flv_mux_video_pad_setcaps), (gst_flv_mux_audio_pad_setcaps),
	  (gst_flv_mux_request_new_pad), (gst_flv_mux_release_pad),
	  (gst_flv_mux_write_header), (gst_flv_mux_write_buffer),
	  (gst_flv_mux_collected), (gst_flv_mux_change_state):
	  * gst/flv/gstflvmux.h:
	  Add first version of a FLV muxer. The only missing feature is writing
	  of stream metadata.

2008-06-13 22:46:43 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 21/57] gst/flv/: Introduce demuxing support for AAC and
	  Original commit message from CVS:
	  2008-06-14  Julien Moutte  <julien@fluendo.com>
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
	  (gst_flv_demux_dispose):
	  * gst/flv/gstflvdemux.h:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_video_negotiate),
	  (gst_flv_parse_tag_video): Introduce demuxing support for AAC
	  and
	  H.264/AVC inside FLV.
	  * sys/dshowdecwrapper/gstdshowaudiodec.c:
	  (gst_dshowaudiodec_init),
	  (gst_dshowaudiodec_chain), (gst_dshowaudiodec_push_buffer),
	  (gst_dshowaudiodec_sink_event), (gst_dshowaudiodec_setup_graph):
	  * sys/dshowdecwrapper/gstdshowaudiodec.h:
	  * sys/dshowdecwrapper/gstdshowvideodec.c:
	  (gst_dshowvideodec_init),
	  (gst_dshowvideodec_sink_event), (gst_dshowvideodec_chain),
	  (gst_dshowvideodec_push_buffer),
	  (gst_dshowvideodec_src_getcaps):
	  * sys/dshowdecwrapper/gstdshowvideodec.h: Lot of random fixes
	  to improve stability (ref counting, safety checks...)

2008-04-25 08:07:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  [MOVED FROM BAD 20/57] gst/flv/gstflvdemux.c: Forward unknown queries upstream instead of returning FALSE on them.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_query):
	  Forward unknown queries upstream instead of returning FALSE on them.

2008-04-11 23:19:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  [MOVED FROM BAD 19/57] gst/flv/gstflvparse.c: Handle NULL returns from FLV_GET_STRING() more gracefully. Fixes crash caused by a strlen on a...
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
	  (gst_flv_parse_tag_script):
	  Handle NULL returns from FLV_GET_STRING() more gracefully. Fixes
	  crash caused by a strlen on a NULL string (#527622).

2007-12-11 11:54:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  [MOVED FROM BAD 18/57] gst/flv/gstflvparse.c: Don't strdup (and thus leak) codec name strings when passing them to gst_tag_list_add().
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video):
	  Don't strdup (and thus leak) codec name strings when passing
	  them to gst_tag_list_add().

2007-12-09 19:37:53 +0000  Edward Hervey <bilboed@bilboed.com>

	  [MOVED FROM BAD 17/57] gst/flv/gstflvparse.c: Fix list of supported and known codecs.
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video):
	  Fix list of supported and known codecs.
	  Emit tag with the codec name so it gets properly reported in totem and
	  other applications.

2007-11-25 10:45:09 +0000  Edward Hervey <bilboed@bilboed.com>

	  [MOVED FROM BAD 16/57] gst/flv/gstflvparse.c: Output segment with proper 'stop' value, makes flvdemux 100% compatible with gnonlin.
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video):
	  Output segment with proper 'stop' value, makes flvdemux 100% compatible
	  with gnonlin.

2007-11-12 19:22:24 +0000  Edward Hervey <bilboed@bilboed.com>

	  [MOVED FROM BAD 15/57] gst/flv/gstflvparse.c: Add mapping for Nellymoser ASAO audio codec.
	  Original commit message from CVS:
	  * gst/flv/gstflvparse.c:
	  Add mapping for Nellymoser ASAO audio codec.
	  (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Make sure we
	  actually have data to read at the end of the tag. This avoids trying
	  to allocate negative buffers.

2007-10-22 15:45:49 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 14/57] gst/flv/gstflvparse.c: Don't emit no-more-pads for single pad scenarios as the header is definitely not reliable. We ...
	  Original commit message from CVS:
	  2007-10-22  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video), (gst_flv_parse_tag_type): Don't
	  emit no-more-pads for single pad scenarios as the header
	  is definitely not reliable. We emit them for 2 pads scenarios
	  though to speed up media discovery.

2007-09-27 10:06:23 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 13/57] gst/flv/gstflvparse.c: I got it wrong again, audio rate was not detected correctly in all cases.
	  Original commit message from CVS:
	  2007-09-27  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video): I got it wrong again, audio rate
	  was not detected correctly in all cases.

2007-09-26 16:30:50 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 12/57] gst/flv/gstflvparse.c: codec_data is needed for every tag not just the first one. (Fix a stupid bug i introduced with...
	  Original commit message from CVS:
	  2007-09-26  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video): codec_data is needed for every tag
	  not just the first one. (Fix a stupid bug i introduced without
	  testing)

2007-09-26 11:17:08 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 11/57] gst/flv/gstflvparse.c: Fix bit masks operations to be sure we detect the codec_tags and sample rates correctly.
	  Original commit message from CVS:
	  2007-09-26  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video): Fix bit masks operations to be
	  sure we detect the codec_tags and sample rates correctly.
	  Fix raw audio caps generation.

2007-09-12 08:38:22 +0000  Peter Kjellerstedt <pkj@axis.com>

	  [MOVED FROM BAD 10/57] gst/: Printf format fixes (#476128).
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst-libs/gst/app/gstappsink.c:
	  * gst/flv/gstflvdemux.c:
	  * gst/flv/gstflvparse.c:
	  * gst/interleave/deinterleave.c:
	  * gst/switch/gstswitch.c:
	  Printf format fixes (#476128).

2007-08-27 14:56:05 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 09/57] gst/flv/gstflvdemux.c: Make sure we initialize the seek result.
	  Original commit message from CVS:
	  2007-08-27  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_handle_seek_pull):
	  Make sure we initialize the seek result.

2007-08-24 17:03:15 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 08/57] gst/flv/gstflvdemux.c: Remove some useless ifdef.
	  Original commit message from CVS:
	  2007-08-24  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_flush),
	  (gst_flv_demux_chain), (gst_flv_demux_pull_tag),
	  (gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push),
	  (gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event),
	  (gst_flv_demux_src_event): Remove some useless ifdef.

2007-08-24 15:31:26 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 07/57] gst/flv/gstflvdemux.c: Implement seeking in push mode.
	  Original commit message from CVS:
	  2007-08-24  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_flush),
	  (gst_flv_demux_cleanup), (gst_flv_demux_chain),
	  (gst_flv_demux_pull_tag), (gst_flv_demux_find_offset),
	  (gst_flv_demux_handle_seek_push),
	  (gst_flv_demux_handle_seek_pull),
	  (gst_flv_demux_sink_event), (gst_flv_demux_src_event): Implement
	  seeking in push mode.
	  * gst/flv/gstflvdemux.h:

2007-08-22 14:50:51 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 06/57] gst/flv/: Handle pixel aspect ratio through metadata tags like ASF does. Fluendo muxer supports this and
	  Original commit message from CVS:
	  2007-08-22  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
	  (gst_flv_demux_pull_tag):
	  * gst/flv/gstflvdemux.h:
	  * gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
	  (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video): Handle pixel aspect ratio through
	  metadata tags like ASF does. Fluendo muxer supports this and
	  Flash players can support it as well this way.

2007-08-22 14:03:42 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 05/57] gst/flv/: Make sure we don't try filling up the index if no times object was parsed. Fix the way we decide to push ta...
	  Original commit message from CVS:
	  2007-08-22  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag):
	  * gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
	  (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio),
	  (gst_flv_parse_tag_video): Make sure we don't try filling up the
	  index if no times object was parsed. Fix the way we decide to
	  push
	  tags and emit no-more-pads. Fix some printf typing in debugging.

2007-08-14 14:56:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  [MOVED FROM BAD 04/57] gst/flv/gstflvdemux.c: Fix locking and refcounting on the index.
	  Original commit message from CVS:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_set_index),
	  (gst_flv_demux_get_index):
	  Fix locking and refcounting on the index.

2007-08-14 14:22:09 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 03/57] gst/flv/gstflvdemux.c: First method for seeking in pull mode using the index built step by step or coming from metadata.
	  Original commit message from CVS:
	  2007-08-14  Julien MOUTTE  <julien@moutte.net>
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
	  (gst_flv_demux_adapter_flush), (gst_flv_demux_chain),
	  (gst_flv_demux_pull_tag), (gst_flv_demux_do_seek),
	  (gst_flv_demux_handle_seek), (gst_flv_demux_sink_event),
	  (gst_flv_demux_src_event), (gst_flv_demux_query),
	  (gst_flv_demux_change_state), (gst_flv_demux_set_index),
	  (gst_flv_demux_get_index), (gst_flv_demux_dispose),
	  (gst_flv_demux_class_init): First method for seeking in pull
	  mode using the index built step by step or coming from metadata.
	  * gst/flv/gstflvdemux.h:
	  * gst/flv/gstflvparse.c: (FLV_GET_STRING),
	  (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Parse
	  more metadata types and keyframes index.

2007-07-25 13:29:04 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 02/57] gst/flv/: Handle not linked pads, try to make it reusable, more safety checks.
	  Original commit message from CVS:
	  2007-07-25  Julien MOUTTE  <julien@moutte.net>
	  (gst_flv_demux_chain), (gst_flv_demux_pull_tag),
	  (gst_flv_demux_change_state), (gst_flv_demux_dispose),
	  (gst_flv_demux_init):
	  * gst/flv/gstflvdemux.h:
	  * gst/flv/gstflvparse.c: (FLV_GET_STRING),
	  (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
	  (gst_flv_parse_header):
	  * gst/flv/gstflvparse.h: Handle not linked pads, try to make it
	  reusable, more safety checks.

2007-07-19 15:05:30 +0000  Julien Moutte <julien@moutte.net>

	  [MOVED FROM BAD 01/57] Adds a first draft of an FLV demuxer.
	  Original commit message from CVS:
	  2007-07-19  Julien MOUTTE  <julien@moutte.net>
	  * configure.ac:
	  * gst/flv/Makefile.am:
	  * gst/flv/gstflvdemux.c: (gst_flv_demux_flush),
	  (gst_flv_demux_cleanup), (gst_flv_demux_chain),
	  (gst_flv_demux_pull_tag), (gst_flv_demux_pull_header),
	  (gst_flv_demux_seek_to_prev_keyframe), (gst_flv_demux_loop),
	  (gst_flv_demux_sink_activate),
	  (gst_flv_demux_sink_activate_push),
	  (gst_flv_demux_sink_activate_pull), (gst_flv_demux_sink_event),
	  (gst_flv_demux_change_state), (gst_flv_demux_dispose),
	  (gst_flv_demux_base_init), (gst_flv_demux_class_init),
	  (gst_flv_demux_init), (plugin_init):
	  * gst/flv/gstflvdemux.h:
	  * gst/flv/gstflvparse.c: (FLV_GET_BEUI24), (FLV_GET_STRING),
	  (gst_flv_demux_query_types), (gst_flv_demux_query),
	  (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
	  (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
	  (gst_flv_parse_tag_type), (gst_flv_parse_header):
	  * gst/flv/gstflvparse.h: Adds a first draft of an FLV demuxer.
	  It does not do seeking yet, it supports pull and push mode so
	  YES
	  you can use it to play youtube videos directly from an HTTP uri.
	  Not so much testing done yet but it parses metadata, reply to
	  duration queries, etc...

2009-05-12 13:00:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/Makefile.am:
	  rtp: Link to -lm
	  Fixes bug #582281.

2009-05-12 11:16:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/rganalysis.c:
	  rganalysis: Remove invalid unit test
	  The test creates buffers with non-silence, sets the GAP
	  flag on it and expects rganalysis to ignore the content and assume silence.
	  That's not the way how GAP buffers should be used, if the GAP flag is set
	  elements *can* assume that they only contain silence but they're not *required*
	  to assume that. The GAP flag must only be set on silence buffers.
	  Fixes bug #582252.

2009-05-12 00:48:49 +0100  Jan Schmidt <thaytan@noraisin.net>

	* ChangeLog:
	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	* win32/common/config.h:
	  0.10.14.2 pre-release

2009-05-11 23:13:20 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/files/Makefile.am:
	  checks: dist id3-577468-unsynced-tag.tag test file

2009-05-11 21:02:27 +0200  Tristan Matthews <le.businessman at gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: initialize variable to 0
	  Fixes #582218.

2009-05-11 18:21:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Only search for the index entry once

2009-05-11 18:18:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Use the first index entry if it's after the seek position

2009-05-11 18:15:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Use the first entry for a given stream if the first entry is after the seek position

2009-05-11 16:50:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Use binary search for finding the requested index entry when seeking

2009-05-11 15:36:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Improve/optimize seeking
	  First of all a keyframe seek should be done to the
	  keyframe right before the requested position and not
	  to the keyframe that is nearest to the requested position.
	  Use per track index arrays and use our new binary search function
	  from core to speed up the search.

2009-05-11 15:36:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  Require released versions of core/base

2009-05-11 10:15:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/Makefile.am:
	  gdkpixbuf: Use the libs and cflags of gdk pixbuf instead of gtk
	  This fixes the build if gdk-pixbuf is found but gtk isn't

2009-05-11 09:58:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  Always define the conditional HAVE_GTK to fix configure in some cases

2009-05-10 16:53:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: Don't write a Xing header

2009-05-10 11:17:23 +0200  Marc-Andre Lureau <marcandre.lureau@gmail.com>

	* autogen.sh:
	  Run libtoolize before aclocal
	  This unbreaks the build in some cases. Fixes bug #582021

2009-05-09 10:50:45 -0700  David Schleef <ds@schleef.org>

	* gst/matroska/matroska-demux.c:
	  matroska: fix printf format to agree with argument

2009-05-08 19:42:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	  raw1394: include stdlib.h for strtol()
	  Fixes compiler warning when compiling with xml stuff in core disabled.

2009-05-08 16:40:57 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Actually output the pending buffer.. and not a blank one.
	  It was previously sending the bogus buffer which was returned from
	  the bufferalloc (required for reverse negotiation apparently) instead
	  of the pending buffer.

2009-05-08 14:24:47 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

	* ext/twolame/gsttwolame.c:
	  Switch twolame to primary rank

2009-05-08 12:00:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Allow non-string fields in the extra-headers property

2009-05-08 11:35:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kdepay.h:
	  rtj2kdepay: add basic JPEG 2000 depayloader

2009-05-08 11:31:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: set marker bit correctly

2009-05-08 11:29:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Add support for extra-headers appended to the HTTP request
	  This allows to set the Referer header among other things by
	  adding a "extra-headers" property that takes a GstStructure
	  with field=string pairs.
	  Fixes bug #581806.

2009-05-08 10:38:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpj2kpay.h:
	  rtpj2kpay: add a simple JPEG 2000 payloader

2009-05-08 10:31:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: we only need to swap bits on LE

2009-05-07 18:10:08 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

	* ext/flac/gstflac.c:
	* ext/jpeg/gstjpeg.c:
	* ext/libpng/gstpng.c:
	* ext/speex/gstspeex.c:
	* gst/avi/gstavi.c:
	* gst/matroska/matroska-mux.c:
	  Add RANKS for various encoders and muxers

2009-05-07 17:59:52 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  Add ranks to mp3 encoders

2009-05-07 17:59:52 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

	* ext/twolame/gsttwolame.c:
	  Add ranks to mp3 encoders

2009-05-07 17:09:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: add some debugging

2009-05-07 15:58:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: parse xiph headers length correctly
	  See #580980

2009-05-07 16:25:41 +0200  Gabriel Bouvigne <bouvigne@mp3-tech.org>

	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrganalysis.h:
	* gst/replaygain/rganalysis.c:
	* gst/replaygain/rganalysis.h:
	  rganalysis: Add ability to post level messages
	  Fixes bug #581568.

2009-05-07 10:10:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: Fixup the bitrate only for CBR
	  Additionally clarify some property descriptions.

2009-05-06 23:56:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: refuse some unsupported jpeg formats

2009-05-06 21:47:17 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* ext/lame/gstlamemp3enc.c:
	  lame: fix format string in debug statement

2009-05-06 18:06:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: fix description

2009-05-06 16:09:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: rewrite quant table handling
	  Rewrite the quant table parsing to also handle multiple tables in one JPEG HDQ
	  segment.
	  Handle more jpeg types by keeping track of the tables used per component and
	  putting the used ones in the quant headers.

2009-04-18 17:23:51 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/elements/id3v2mux.c:
	  id3v2mux: Make the test failure slightly more informative

2009-04-20 18:33:09 +0100  Jan Schmidt <thaytan@noraisin.net>

	* ext/flac/gstflacdec.c:
	  flac: Make buffers created during seek act like normal buffers.
	  Store the offset and caps when allocating a buffer during seeking, and then
	  allocate a new buffer with buffer_alloc before we push it out. This ensures
	  that in all respects the first buffer decoded during seeking behaves like
	  all other buffers, including allowing downstream re-negotiation.

2009-04-18 18:00:54 +0200  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/flac/gstflacdec.c:
	  flacdec: don't use pad_alloc when decoding while seeking. Fixes #579422

2009-05-06 13:22:51 +0200  Arnout Vandecappelle <arnout@mind.be>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: refactored gst_jpeg_dec_parse_image_data
	  Fixes #579808

2009-05-06 13:11:53 +0200  Arnout Vandecappelle <arnout@mind.be>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: support additional 0xff before end marker.
	  JPEG markers may be preceded by additional 0xff.  jpegdec should
	  skip over these, even before the end marker.
	  See #579808

2009-05-06 12:54:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: handle input with 1 quant table
	  Also handle input with just one quant table, simply duplicate the quant table.
	  Handle invalid SOF correctly and some small cleanups.
	  Fixes #578257

2009-04-29 15:58:10 +0300  Marco Ballesio <marco.ballesio@nokia.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix byte order swapping in 3GPP classification entity tag
	  Fixes #580746.

2009-05-05 16:38:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lame: fix compilation with LAME versions < 3.98
	  lame_set_VBR_quality(), which takes a floating point value for the
	  quality, has been added only in v3.98. Use lame_set_VBR_q(), which
	  takes quality as an integer, for older LAME versions.
	  Fixes #581341.

2009-05-05 17:07:13 +0200  Arnout Vandecappelle <arnout@mind.be>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: avoid reading from inavlid memory
	  Read the timestamp of the incomming buffer before we push it in the adapter and
	  flush it out again as the buffer might be unreffed then and we read from invalid
	  memory.
	  Fixes #581444.

2009-05-05 17:03:29 +0200  Arnout Vandecappelle <arnout@mind.be>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: don't leak dynamic pads
	  Free the dynamic pads data in finalize.
	  Fixes #581432

2009-05-05 16:32:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpjpegpay.h:
	  rtpjpegpay: correctly set the type header
	  Don't require width/height on the caps. Use the SOF header to find width/height
	  and fall back to the caps if there is no SOF. Also use the SOF info to find the
	  subsampling and quantization tables used. This allows us to set the right type
	  value in the JPEG rtp header.
	  Deprecate the quality property, it's unused now and it was used wrongly before.
	  Always send full quant tables for now until we have some code to detect default
	  ones.
	  Fixes #580880

2009-05-05 16:28:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegdepay.h:
	  rtpjpegdepay: use width/height from payload
	  Use the width and the height from the payload headers and set them on the
	  output caps for added awesomeness.
	  Fix quant parsing, we need to check the type in the lower 6 bits.
	  Add first bits of caching quantization tables.

2009-05-05 16:24:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: set colorspace before _set_defaults()
	  The libjpeg api says that we need to set the colorspace before we call
	  _set_defaults(). Indeed, if we don't do that we end up with some very freaky
	  non-standard quant table and huffman table indexes.

2009-05-05 13:19:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/Makefile.am:
	  tests: don't build examples if --disable-examples was passed to configure

2009-05-05 12:33:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: clean up mess around gtk+ checking
	  And don't check for gtk+ when it's not needed (ie. if examples are disabled)

2009-05-05 12:27:21 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/pixbufscale.h:
	  configure: make gdk-pixbuf plugin depend only on gdk-pixbuf, not gtk+

2009-05-04 18:55:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix find_stream_by_* functions
	  Fix various version of find_stream_by_* by not trying to convert an int to a
	  pointer and vice versa, for portability reasons.
	  Fixes #581333

2009-05-04 18:32:05 +0200  Chris Winter <elwintro at gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix dummy nat packet logic
	  Fix a typo in the dummy NAT packet sending code.
	  Fixes #581329

2009-04-30 10:24:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid errors after server eof
	  Server eof (e.g. connection closed) is announced as connection closed,
	  so better record state and act accordingly to prevent (read/write)
	  errors during subsequent teardown/cleanup sequences.  #Fixes 580851.(c).

2009-04-30 10:19:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: also set base_time on src after flush
	  timestamps following flush/seek should be consistent between
	  UDP and TCP interleaved case.  Fixes #580851.(b).

2009-04-30 10:17:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: sanity checks on range info
	  A max range that overflows should not be trusted,
	  nor should a max range that equals the min range.
	  Fixes #580851.(a).

2009-05-04 16:16:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: use SKIP flag to use SCALE headers
	  We can use the SKIP seek flag to instruct the server to send data faster then
	  normal but with the same bandwidth.
	  Fixes #537609

2009-05-04 14:19:22 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* ext/speex/gstspeexdec.c:
	  speexdec: make speex_dec_convert work with same-format values when no data has been decoded.

2009-05-04 12:51:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lamemp3enc: Add a note to the encoding-engine-quality property
	  that says, that this does not affect the bitrate at all.

2009-05-04 12:48:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  lame: Implement preset interface

2009-05-04 12:47:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/twolame/gsttwolame.c:
	  twolame: Implement preset interface

2009-05-04 12:43:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flac: Implement preset interface

2009-05-04 12:41:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speex: Implement preset interface

2009-05-04 12:40:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/wavpack/gstwavpackenc.c:
	  wavpack: Implement preset interface

2009-05-04 12:35:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use binary search for index
	  Use the new binary search method for finding the right index entry faster.

2009-05-04 11:26:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: draw the complete U and V planes
	  Round up the scaled U and V width and height so that we always draw the correct
	  amount of pixels to fill the complete image.
	  Fixes #569611

2009-04-30 10:21:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	* ext/lame/gstlamemp3enc.h:
	  lamemp3enc: Remove fast-vbr property and rename vbr-quality to quality

2009-04-30 10:16:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	  lame/lamemp3enc: Fix memory leak on FLUSH_STOP

2009-04-30 10:14:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlame.c:
	  lame: Deprecate the lame element

2009-04-30 10:10:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/gstlamemp3enc.c:
	  lame: Update example pipelines with the new properties

2009-04-29 19:01:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/lame/Makefile.am:
	* ext/lame/gstlame.c:
	* ext/lame/gstlamemp3enc.c:
	* ext/lame/gstlamemp3enc.h:
	* ext/lame/plugin.c:
	  lame: Add lamemp3enc element with much simplified interface
	  This deprecates the lame element and fixes bug #494528.

2009-05-01 19:35:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: add some more micro optimisations

2009-04-30 18:41:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_dump.c:
	* gst/qtdemux/qtdemux_types.c:
	  qtdemux: micro optimize qtdemux a little
	  Sprinkle some G_LIKELY around.
	  Avoid traversing and dumping the tree when debugging is not activated.

2009-04-30 14:22:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: add support for subtitle pictures
	  Add support for subtitle pictures.
	  Fixes #568278.

2009-04-30 10:32:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: make sure we always signal waiters
	  Always signal the waiters in the async callbacks. Especially for the volume
	  callbacks since this might cause deadlocks.

2009-04-29 18:09:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: release state lock before stopping task
	  We need to release the state lock before trying to wait for the task to end
	  because the task might also take the lock.
	  Fixes #577671

2009-04-29 12:19:27 +0200  Hans de Goede <jwrdegoede at fedoraproject.org>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: handle ac-3 audio
	  fix demuxing of m4v streams with ac-3 audio
	  Fixes #580554

2009-04-29 11:12:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: Use the tag merge mode that was set on the interface for merging tag events

2009-04-25 09:43:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix getaddrinfo error reporting
	  getaddrinfo errors should be reported with gai_strerror instead of errno as
	  spotted by MikeS.

2009-04-27 10:08:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg726pay.c:
	  g726pay: fix compilation

2009-04-27 10:02:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg726pay.h:
	  g726pay: add RFC compliant packetizing
	  Shuffle the input bits according to RFC 3551 for G726 payloads.
	  Add option to force the previous behaviour.
	  Fixes #567140

2009-04-27 09:59:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg726depay.c:
	  g726depay: add debug category
	  Add a debugging category, add some comments and remove _peek_parent().

2009-04-26 15:59:50 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  id3v2mux: we need taglib 1.5 for ID3v2::RelativeVolumeFrame::setIdentification
	  Bump taglib requirement.

2009-04-24 02:11:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/id3demux.c:
	* tests/files/id3-577468-unsynced-tag.tag:
	  id3demux: add unit test file for unsynced id3 tags

2009-04-24 01:51:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3tags.c:
	  id3demux: parse unsynchronised tags properly
	  We didn't handle unsynchronization at all up to now, which might have
	  caused frames to not be extracted - esp. frames after an APIC picture
	  frame. Fixes #577468.

2009-04-24 01:01:53 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3tags.c:
	  id3demux: pass the right size value for size of all frames to the parser
	  Frame data size is tag size adjusted for size of the tag header and
	  footer, not tag size including header and footer.

2009-04-22 15:24:55 +0200  Patrick Radizi <patrick.radizi at axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix some more pad leaks
	  Fix some pad leaks.
	  See #577318.

2009-04-21 22:12:45 +0100  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Automatic update of common submodule
	  From b3941ea to 6ab11d1

2009-04-21 14:02:01 -0700  Michael Smith <msmith@songbirdnest.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: override caps based on data from ESDS atoms in mpeg4.
	  If the codec is actually something else (e.g. mjpeg) change the caps to
	  match when parsing the ESDS atom.
	  Also, for AAC, override rate and channels with correct values read from
	  ESDS, since the rate/channels values elsewhere are often wrong.

2009-04-20 19:32:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix warning for still images by not trying to divide by 0
	  Don't pass a 0 divisor to gst_util_uint64_scale(), or it will complain
	  in the single image case where fps=0/1 (are we supposed to differentiate
	  between no fps=still image and fps=0/1=variable rate here btw?)

2009-04-20 17:25:34 +0100  Jan Schmidt <thaytan@noraisin.net>

	* gst/udp/gstudpnetutils.c:
	  udp: Fix a simple typo in the previous commit
	  Use #ifdef instead of #if, to fix the build

2009-04-20 15:48:21 +0200  Andy Wingo <wingo@wingomac.bcn.oblong.net>

	  fix format string in pngdec
	  * ext/libpng/gstpngdec.c: Fix size_t vs unsigned int format in error message.

2009-04-20 15:46:03 +0200  Andy Wingo <wingo@wingomac.bcn.oblong.net>

	  only use struct ip_mreqn if it is detected
	  * configure.ac: Make an explicit check for struct ip_mreqn.
	  * gst/udp/gstudpnetutils.c: Use HAVE_IP_MREQN instead of the ad-hoc checks.

2009-04-20 13:45:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  Fix push mode buffering sanity check to actually fit the description.

2009-04-19 14:03:38 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/twolame/gsttwolame.c:
	  twolame: Remove unneeded variable, value assigned was never read.

2009-04-19 14:02:03 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/lame/gstlame.c:
	  lame: Remove unneeded variable, it's assigned a value never read.

2009-04-18 19:11:06 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: And let's not forget to remove the unused variable.

2009-04-18 18:50:32 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: Remove dead assignments, the variables are never read after.

2009-04-18 18:49:49 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpmp4vpay.c:
	  rtpmp4vpay: Remove dead assignment. The value is never read after.

2009-04-18 18:48:55 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Remove dead assignment.
	  t is being overwritten after, before it's used.

2009-04-18 18:48:06 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpamrdepay.c:
	  rtpamrdepay: Remove unneeded variable, the value is only read once.

2009-04-18 18:47:05 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpamrpay.c:
	  rtpamrpay: Remove unneeded variable, the value is only read once.

2009-04-18 18:46:12 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/goom/filters.c:
	  goom/filters: Remove dead assignment. Value overwritten just after.

2009-04-18 18:45:32 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: Remove dead assignment. Value never read after.

2009-04-18 18:45:07 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: Remove dead assignment. Value never read after.

2009-04-18 18:43:31 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: Remove unused variable, it's never being read.

2009-04-18 18:42:45 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Remove dead assignment. 'res' isn't read after.

2009-04-18 18:41:58 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Remove unused variable. 'res' is never read.

2009-04-18 18:40:48 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Remove dead variable. 'stream' is never read after.

2009-04-18 18:39:48 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/videobox/gstvideobox.c:
	  videbox: Remove dead assignments.
	  These variables are never read after this point.

2009-04-18 18:38:29 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/goom/convolve_fx.c:
	  goom: ff and iff are only used in a '#ifdef DRAW_MOTIF' block.

2009-04-18 18:34:11 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Remove dead assignment.
	  res isn't read after this.

2009-04-18 18:32:03 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Remove dead assignments, move variable to where it's needed.
	  The header_read_error label will return GST_FLOW_ERROR

2009-04-18 18:21:22 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtpvrawdepay: Remove dead assignment.
	  The value of 'str' will never be used in these cases.

2009-04-18 18:19:12 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Remove useless variable.
	  iret was never read outside of that loop, and is always being exited if
	  iret was != GST_FLOW_OK anyway.

2009-04-18 18:17:35 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Move 'res' to where it's actually being used.
	  res was never used outside of that block except for a dead assignment.

2009-04-18 18:16:33 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	  audiofx: Remove unused variable.
	  rz is never used in these methods.

2009-04-18 18:15:39 +0200  Edward Hervey <bilboed@bilboed.com>

	* sys/osxaudio/gstosxringbuffer.c:
	  osxringbuffer: Run gst-indent.

2009-04-18 18:14:49 +0200  Edward Hervey <bilboed@bilboed.com>

	* sys/ximage/gstximagesrc.c:
	  ximage: Remove dead assignments.
	  Those variables are not read after that point.

2009-04-18 18:11:00 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/dv/gstdvdemux.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libpng/gstpngdec.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/speex/gstspeexenc.c:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/auparse/gstauparse.c:
	* gst/effectv/gstquark.c:
	* gst/flx/gstflxdec.c:
	* gst/icydemux/gsticydemux.c:
	* gst/interleave/interleave.c:
	* gst/matroska/matroska-mux.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/qtdemux/gstrtpxqtdepay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/smpte/paint.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videomixer/videomixer.c:
	* gst/wavparse/gstwavparse.c:
	* sys/ximage/gstximagesrc.c:
	  Remove trivial unused variables detected by CLang static analyzer.

2009-04-18 17:52:00 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/gconf/gstswitchsink.c:
	* gst/qtdemux/gstrtpxqtdepay.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  Remove blank {set|get}_property/change_state/finalize methods.

2009-04-18 17:42:55 +0200  Edward Hervey <bilboed@bilboed.com>

	* ext/cairo/gsttimeoverlay.c:
	* ext/esd/esdsink.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/pulse/pulsesink.c:
	* gst/alpha/gstalphacolor.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/efence.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gsttaginject.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/udp/gstudpsink.c:
	* gst/videofilter/gstvideobalance.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	  Remove unused variables in _class_init
	  Detected by LLVM's CLang static analyzer

2009-04-18 13:54:08 +0100  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/elements/souphttpsrc.c:
	  check: Check whether threads are already initialised before g_thread_init()

2009-04-18 14:32:40 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: mark discont on the streams as was said the debug line
	  After a seek mark all streams with discont as it was said in the debug line.
	  Fixes that buffers after a seek are generated without a valid timestamp.

2009-04-18 08:45:18 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: map GST_RTSP_EEOF to EOS on server requests
	  Permit properly handle the EOS condition when server report it in a request.

2009-04-18 08:39:57 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: Fix build on macosx.
	  Use G_GSIZE_FORMAT instead of u.

2009-04-16 22:50:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix sample offset calculation again

2009-04-15 19:32:18 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	  sunaudio: fix broken indentation of variable declarations

2009-04-15 19:28:53 +0100  James Andrewartha <trs80@ucc.gu.uwa.edu.au>

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiosink.c:
	  sunaudio: remove some unused variables and goto labels
	  Fixes #579070.

2009-04-15 19:24:49 +0200  James Andrewartha <trs80 at ucc.gu.uwa.edu.au>

	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	  rtph263pay: fix compilation on big-endian
	  Some semicolons were missing from the big-endian structs in gstrtph263pay.h.
	  A GST_DEBUG call was missing a format specifier.
	  Fixes #579069

2009-04-15 20:10:04 +0300  Marco Ballesio <marco.ballesio@nokia.com>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	* gst/qtdemux/qtdemux_fourcc.h:
	* gst/qtdemux/qtdemux_types.c:
	* gst/qtdemux/quicktime.c:
	  qtdemux: implement 3GPP (TS 26.244 V8.0.0) Asset metadata handling, Fixes #132193
	  Implements 3gpp iso metadata tags which are different from mov udta atoms.

2009-04-15 15:51:24 +0200  Peter Kjellerstedt <pkj@axis.com>

	* gst/debugutils/efence.h:
	  debugutils: Use G_BEGIN_DECLS/G_END_DECLS.
	  Use G_BEGIN_DECLS/G_END_DECLS to avoid gst-indent messing up the
	  indentation due to extern "C" { }.

2009-04-15 16:03:27 +0300  Stefan Kost <ensonic@users.sf.net>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* gst/debugutils/Makefile.am:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/debug.vcproj:
	* gst/debugutils/efence.c:
	* gst/debugutils/efence.h:
	* gst/debugutils/efence.vcproj:
	* gst/debugutils/gstdebug.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavigationtest.h:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstnavseek.h:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gstpushfilesrc.h:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/gsttaginject.h:
	* gst/debugutils/navigationtest.vcproj:
	* gst/debugutils/negotiation.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/progressreport.h:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	* gst/debugutils/tests.c:
	* gst/debugutils/tests.h:
	  debug: rename debug to debugutils to avoid clash with --disable-debug. Fixes #562168

2009-04-15 15:43:04 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/debug/efence.c:
	* gst/debug/efence.h:
	* gst/debug/gstnavigationtest.h:
	* gst/debug/gstnavseek.h:
	* gst/debug/gstpushfilesrc.h:
	* gst/debug/gsttaginject.h:
	* gst/debug/progressreport.h:
	* gst/debug/tests.h:
	  debug: indent before renaming

2009-04-15 14:07:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg726depay.c:
	  g726depay: add property for aal2 force

2009-04-15 13:56:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726depay.h:
	  g726depay: implement RFC3551 packing
	  We implemented the AAL2 packing, add the encoding-name for those to the caps and
	  a property to force AAL2 decoding (always TRUE for now).
	  Implement RFC3551 unpacking for regular G726.
	  See #567140.

2009-04-15 00:22:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263pay.h:
	  rtph263pay: fix build

2009-04-14 18:52:48 +0200  Youness Alaoui <youness.alaoui at collabora.co.uk>

	* gst/rtp/gstrtph263pay.c:
	  h263pay: various fixes
	  Re-enable mode A support and a property to control it.
	  Fix memory leak of GstRtpH263PayBoundry objects.
	  Fix marker.
	  Fixes #509311

2009-04-14 18:44:51 +0200  Janin Kolenc <janin.kolenc at marand.si>

	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	  h263pay: Fix the payloader
	  Fix the H263 payloader to be more RFC 2190 compliant.
	  See #509311

2009-04-14 17:27:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: don't push EOS in streaming mode
	  In streaming mode, avidemux is not supposed to send an EOS event downstream but
	  it is supposed to return UNEXPECTED from the chain function instead so that
	  upstream can do the right EOS handling.

2009-04-13 14:03:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  Add initial support for muxing/demuxing Speex audio
	  Note: This is not in the Matroska spec yet
	  Fixes bug #578310.

2009-04-10 21:31:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: handle NULL timing info
	  Don't crash when the timing info is not yet available.

2009-04-10 21:42:13 +0300  Stefan Kost <ensonic@users.sf.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulse: make it work on 0.9.12
	  First we ignore request to fill the ringbuffer which are less then a segment.
	  The small request where causing stutter.
	  Then we disable flushing the stream when running against pa 0.9.12 as this
	  triggers an assertiong in the sound server and terminates it. It does not happen
	  with 0.9.10 and 0.9.14.

2009-04-10 14:18:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: handle server disconnect in get_time
	  When the server is disconnected or when we are shut down, make our clock return
	  an invalid time instead of erroring out.

2009-04-10 12:01:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: bps is signed int to avoid overflow
	  Keep bps as gint instead of guint because we will be doing signed math with it
	  later on and we don't want weird results.

2009-04-10 00:26:44 +0200  LRN <lrn1986 at gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: add convert query, fix duration query
	  Fix the duration query so that it also works with formats other than
	  TIME, such as DEFAULT to get the number of frames.
	  Add a convert function.
	  Fixes #578052.

2009-04-09 23:43:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: check for a stream
	  Don't try to change the stream volume (and other things) when we don't have a
	  stream yet. Just store the values for later.

2009-04-09 18:07:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix compilation for newer pulseaudio

2009-04-09 17:18:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: uncork fixes and use prebuf = 0
	  We can use prebuf = 0 to instruct pulse to not pause the stream on underflows.
	  This way we can remove the underflow callback. We however have to manually
	  uncork the stream now when we have no available space in the buffer or when we
	  are writing too far away from the current read_index.

2009-04-09 14:38:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: handle write errors

2009-04-09 14:16:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: write silence on underflow
	  Start filling up the buffer with empty samples when an underflow happens. We
	  need to do this to keep pulseaudio reporting the right time for us.

2009-04-09 13:14:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: handle pull-based scheduling
	  Use the default basesink methods for implementing pull based scheduling, it
	  works fine for us.

2009-04-09 12:13:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: add beginnings of pull-based scheduling

2009-04-08 18:17:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: keep track of clock reset
	  when we switch streams, the clock will reset to 0. Make sure that the provided
	  clock doesn't get stuck when this happens by keeping an initial offset. We also
	  need to make sure that we subtract this offset in samples when writing to the
	  ringbuffer.

2009-04-08 13:52:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: rewrite pulsesink
	  Derive from BaseAudioSink and implement our custom ringbuffer that maps to the
	  internal pulseaudio ringbuffer.

2009-04-08 13:52:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseutil.c:
	  pulse: remove some stray debug lines

2009-04-09 11:30:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: use slightly more adaptive formula for QoS
	  Should work at least a tad better if the decoder can't keep up, and
	  should also spread dropped frames a bit more evenly over time.

2009-04-07 22:35:31 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  wavparse: don't leak pad-template
	  gst_element_class_add_pad_template() does not take ownership.

2009-04-04 21:18:55 +0300  Felipe Contreras <felipe.contreras@gmail.com>

	* common:
	  Automatic update of common submodule
	  From d0ea89e to b3941ea

2009-04-01 01:15:31 +0200  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  add pending_samples so that we only update segment's last stop after really sending the samples

2009-03-15 21:31:49 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* tests/check/pipelines/flacdec.c:
	  add debug and an assert

2009-03-15 21:30:32 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* ext/flac/gstflacdec.c:
	  add debugging

2009-03-03 10:14:02 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* tests/check/Makefile.am:
	* tests/check/audiotestsrc.flac:
	* tests/check/pipelines/flacdec.c:
	  add a test to check that we get all decoded bytes from a 10-buffer audiotestsrc flac, in the case of:  - a full decode  - a decode of a seek for the full file  - a decode of a seek for a small part, smaller than the first buffer
	  The test fails because flacdec drops the first outgoing buffer on a seek

2009-03-03 10:06:52 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* ext/flac/gstflacdec.c:
	  clipping should also work if it's done on the first buffer starting at 0

2009-04-04 14:54:01 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From f8b3d91 to d0ea89e

2009-04-03 09:57:15 +0100  Zaheer Merali <zaheerabbas@merali.org>

	* gst/qtdemux/LEGAL:
	  Fix grammar.

2009-04-02 22:41:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: allow http:// on the proxy setting
	  Allow and ignore http:// at the start of the proxy setting, like
	  souphttpsrc.
	  Fixes #573173

2009-04-02 21:08:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't leak the udpsrc pad
	  Fix memory leak in rtspsrc because we didn't unref the udpsrc pad.
	  See #577318

2009-04-01 17:31:18 -0700  Michael Smith <msmith@songbirdnest.com>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: fix length encoding in packed headers.
	  As for vorbis payloader; this by inspection had the same bug.

2009-04-01 17:23:33 -0700  Michael Smith <msmith@songbirdnest.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: in packed headers, properly flag multibyte lengths.
	  In the sequence of header lengths, for headers >127 bytes, we use
	  multiple bytes to encode the length. Bytes other than the last must have
	  the top (flag) bit set.

2009-04-02 00:20:02 +0100  Jonathan Matthew <jonathan@d14n.org>

	* ext/taglib/gstid3v2mux.cc:
	* tests/check/elements/id3v2mux.c:
	  id3v2mux: write RVA2 frames containing peak/gain volume data

2009-04-02 00:05:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: demote some log message from DEBUG to LOG
	  And log decoder object.

2009-04-01 21:15:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: implement basic QoS
	  Don't decode frames that are going to be too late anyway.

2009-04-01 12:26:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't emit ugly warnings with older rtpjitterbuffer versions
	  The on-npt-stop signals was added only recently to rtpjitterbuffer in
	  -bad, so check if the signal exists before g_signal_connect()ing to
	  it, to avoid warnings.

2009-03-31 19:08:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add proxy support

2009-03-31 17:16:04 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-mux.c:
	  matroska: don't leak serialized values when writing tags

2009-03-31 17:06:50 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-demux.c:
	  matroska: don't alter passed data and especialy don't leak.
	  If we need different size, Make a copy, work with that and free it.

2009-03-31 16:42:15 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/goom/plugin_info.c:
	  goom: the structure is not fully initialized, but the copied.
	  Set to fully to 0 to avoid creep of uninitialized values.

2009-03-31 16:25:58 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-mux.c:
	  matroska: init endianess as such and signedness as boolean.

2009-03-31 16:22:42 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: don't use ininitialized var in debug log statement
	  Also make the log statement useful by printing the human readable format name.

2009-03-31 12:01:21 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: don't leak atom data in case of a wrong fourcc

2009-03-31 11:57:36 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/matroska/matroska-demux.c:
	  matroska: don't leak read data in demuxer

2009-03-31 11:50:41 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	  udp: don't use protocol in debug message after freeing

2009-03-30 14:10:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	  rtpmp4adepay: output should be framed already

2009-03-27 21:17:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flac: require a 'newer' flac and remove support for the legacy flac API

2009-03-27 17:48:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: link to the on_npt_stop signal to EOS
	  Connect to the on_npt_stop signal of the session manager to schedule the EOS
	  actions.

2009-03-26 14:39:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: some stream synchronization to aid seeking in unbalanced clips
	  Some clips (trailers) may have (length-wise) unbalanced streams,
	  which stalls the pipeline if seeking into that region.
	  Additional stream synchronization can handle this, as well as
	  sparse (subtitle) streams (at some later time ?)

2009-03-26 10:31:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: additional safety and sanity checks (push based mode)

2009-03-26 10:18:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: some more indent fixes

2009-03-24 16:00:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: fix gst-indent screwup

2009-03-25 17:54:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtsp.c:
	* gst/rtsp/gstrtspsrc.c:
	* po/POTFILES.in:
	  rtspsrc: better error message when the RTSP extension for Real streams is missing
	  Try to post a decent error message when it looks like we're failing
	  because the Real RTSP extension plugin is missing. Also add i18n
	  bits for rtspsrc so our error messages get translated.

2009-03-25 15:42:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavi.c:
	* gst/qtdemux/quicktime.c:
	  i18n: make sure gettext gives us UTF-8 at all times

2009-03-25 01:28:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	  rtpmp4apay,rtpmp4depay: fix buffer leaks in AAC payloader and depayloader

2009-03-25 01:22:17 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpmp4apay.c:
	  rtpmp4apay: warn if input is unframed

2009-03-22 21:20:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: put GstSegment inside the element struct instead of allocating it separately

2009-03-25 10:08:41 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: move duplicated timestamping and buffer metadata code to _create()
	  This will include the latency changes also in the mmap case.

2009-03-25 10:06:48 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: remove win32 ifdefs introduced by commit cff3f46760eac74c9bbd7a36aca44fedf327424b
	  V4l2src is under sys and does not exists/run under windows anyway.

2009-03-24 15:44:42 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: handle FLUSH_STOP event
	  Clean up some state (most notably pad flow returns) to resume
	  proper streaming following flushing seek.

2009-03-24 12:42:13 +0100  Alessandro Decina <alessandro.decina@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: don't post an error if EOS can't be pushed downstream.
	  This aligns avidemux with other demuxers and fixes a bug using avidemux
	  with a recent gnonlin.

2009-03-23 11:22:08 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: clean up the state change function
	  Make the state change function a bit more readable and only pause after the
	  parent had a change to pause first.

2009-03-09 23:43:55 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/dtmf/Makefile.am:
	  Makefile.am: no static libs for plugins

2009-03-20 17:22:32 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: support seeking in push based mode

2009-03-20 17:11:39 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: align push based behaviour more with pull based
	  Cater for DELTA_UNIT flag on buffers, keep track of current
	  position, remove and warn about edit lists if any (as those
	  as are de facto discarded anyway), add some debug statements
	  and indent fixes.

2009-03-20 17:03:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix mem leaks and prevent excessive buffering in push based mode

2009-03-20 13:27:59 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Track the corked/uncorked state ourselves
	  Use an instance variable to track whether the stream is corked or not,
	  instead of using PA API that was only introduced in 0.9.11

2009-03-19 18:39:04 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ext/pulse/pulsesink.c:
	  pulse: Make sure the stream is uncorked in the write function
	  If the caps changes, the sink is reset without transitioning through
	  a PAUSED->PLAYING state change, resulting in a corked stream. This avoids
	  the problem by checking that the stream is uncorked when writing samples
	  to it.

2009-03-20 01:02:26 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: fix direction of latency query and other upstream queries
	  Don't send queries back to the element they just came from by sending
	  them to the peer of the wrong pad.

2009-03-19 11:10:40 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* .gitignore:
	* tests/check/elements/.gitignore:
	  .gitignore: ignore more

2009-03-18 16:55:27 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	  rtpmp4adepay: don't append an extra 0 byte to the codec data
	  The audioMuxVersion structure is packed in such a way that the codec
	  data does not start byte-aligned, which means there's an extra bit of
	  padding at the end. We don't want that bit in the codec data, since
	  some decoders seem get confused when they're fed with an extra codec
	  data byte (also it's just not right of course).

2009-03-19 13:25:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: fix base64 decoding
	  We can't pass -1 to _decode_step, that functions returns 0 right away instead of
	  decoding up to the string end.

2009-03-19 13:24:02 +0100  David Adam <zanchey at ucc.gu.uwa.edu.au>

	* gst/udp/gstudpnetutils.c:
	  udp: Fix build if on Solaris
	  This patch checks for Solaris and uses ip_mreq instead of ip_mreqn if on this
	  platform.
	  Fixes #575937.

2009-03-18 14:50:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp: Use GLib functions for encoding/decoding base64

2009-03-16 19:17:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add some debug for the timestamps
	  When timestamping in TCP mode, log the first timestamp we put on the buffers.

2009-03-15 23:26:56 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: log details if we have them, needed for #575391

2009-03-13 18:32:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: convert _ in properties to -
	  --

2009-03-13 18:28:59 +0100  Edgar E. Iglesias <edgar.iglesias@gmail.com>

	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpnetutils.c:
	* gst/udp/gstudpnetutils.h:
	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: Add network interface selection
	  Add network interface selection when joining multicast groups.
	  Useful when using the udpsrc on multihomed hosts.
	  Fixes #575234.
	  API: GstUDPSrc::multicast-iface

2009-03-13 15:43:52 +0000  Jan Schmidt <thaytan@noraisin.net>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: Prepend to lists and reverse them at the end.
	  Gratuitous micro-optimisation - prepend to lists and reverse them, rather
	  than appending to them each time.

2009-03-13 15:40:50 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ext/pulse/pulsesink.c:
	  pulsesink: Wait until there is enough room to write an entire segment
	  When trying to write out a segment, wait until there is enough free space
	  for the entire segment. This helps to reduce ripple in the clock reporting,
	  where the app might query the playback position while only half a segment
	  has been written (and is therefore reported by _delay(), even though
	  the ring buffer has not yet been advanced)

2009-03-12 20:38:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: don't send PAUSE when not connected
	  don't send a PAUSE request when we are no longer connected.

2009-03-12 16:10:25 +0100  Laszlo Pandy <laszlok2@gmail.com>

	* ext/flac/gstflacdec.c:
	  Don't call FLAC__ methods before it's initialized. Fixes #516031
	  In the event handler, gst_flac_dec_sink_event(), two functions are called on
	  the FLAC stream without checking if it has been initialized:
	  FLAC__stream_decoder_flush()
	  FLAC__stream_decoder_process_until_end_of_stream()
	  Both these FLAC__*() functions modify the internal state of the FLAC stream.
	  Later, when the buffers start flowing, gst_flac_dec_chain() tries to initialize
	  the stream. the FLAC__stream_decoder_init_stream() call will fail because the
	  previous calls to FLAC__*() changed the stream state so it is no longer in the
	  initialized state.

2009-03-11 17:59:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix timeout check
	  ---

2009-03-11 12:48:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* win32/MANIFEST:
	  win32: update MANIFEST, fixing 'make dist'
	  config.h.in no longer exists.

2009-03-10 21:14:43 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/multipart/Makefile.am:
	  makefile: fix typo in no-static plugins rule

2009-03-10 11:01:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/libpng/gstpngdec.c:
	  pngdec: various cleanups.
	  Make some code more readable.
	  Fix a leak when pull range returns a shot buffer.
	  Push EOS after posting the error.

2009-03-10 10:16:27 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/rtp/gstrtpvorbisdepay.c:
	  gstrtpvorbisdepay: Fix build on macosx

2009-03-01 17:37:56 +0100  Edward Hervey <bilboed@bilboed.com>

	* .gitignore:
	  .gitignore: Ignore m4 directory

2008-11-04 12:42:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  [MOVED FROM BAD] Don't install static libs for plugins. Fixes #550851 for -bad.
	  Original commit message from CVS:
	  * ext/alsaspdif/Makefile.am:
	  * ext/amrwb/Makefile.am:
	  * ext/apexsink/Makefile.am:
	  * ext/arts/Makefile.am:
	  * ext/artsd/Makefile.am:
	  * ext/audiofile/Makefile.am:
	  * ext/audioresample/Makefile.am:
	  * ext/bz2/Makefile.am:
	  * ext/cdaudio/Makefile.am:
	  * ext/celt/Makefile.am:
	  * ext/dc1394/Makefile.am:
	  * ext/dirac/Makefile.am:
	  * ext/directfb/Makefile.am:
	  * ext/divx/Makefile.am:
	  * ext/dts/Makefile.am:
	  * ext/faac/Makefile.am:
	  * ext/faad/Makefile.am:
	  * ext/gsm/Makefile.am:
	  * ext/hermes/Makefile.am:
	  * ext/ivorbis/Makefile.am:
	  * ext/jack/Makefile.am:
	  * ext/jp2k/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/lcs/Makefile.am:
	  * ext/libfame/Makefile.am:
	  * ext/libmms/Makefile.am:
	  * ext/metadata/Makefile.am:
	  * ext/mpeg2enc/Makefile.am:
	  * ext/mplex/Makefile.am:
	  * ext/musepack/Makefile.am:
	  * ext/musicbrainz/Makefile.am:
	  * ext/mythtv/Makefile.am:
	  * ext/nas/Makefile.am:
	  * ext/neon/Makefile.am:
	  * ext/ofa/Makefile.am:
	  * ext/polyp/Makefile.am:
	  * ext/resindvd/Makefile.am:
	  * ext/sdl/Makefile.am:
	  * ext/shout/Makefile.am:
	  * ext/snapshot/Makefile.am:
	  * ext/sndfile/Makefile.am:
	  * ext/soundtouch/Makefile.am:
	  * ext/spc/Makefile.am:
	  * ext/swfdec/Makefile.am:
	  * ext/tarkin/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/timidity/Makefile.am:
	  * ext/twolame/Makefile.am:
	  * ext/x264/Makefile.am:
	  * ext/xine/Makefile.am:
	  * ext/xvid/Makefile.am:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/dshow/Makefile.am:
	  * gst/aiffparse/Makefile.am:
	  * gst/app/Makefile.am:
	  * gst/audiobuffer/Makefile.am:
	  * gst/bayer/Makefile.am:
	  * gst/cdxaparse/Makefile.am:
	  * gst/chart/Makefile.am:
	  * gst/colorspace/Makefile.am:
	  * gst/dccp/Makefile.am:
	  * gst/deinterlace/Makefile.am:
	  * gst/deinterlace2/Makefile.am:
	  * gst/dvdspu/Makefile.am:
	  * gst/festival/Makefile.am:
	  * gst/filter/Makefile.am:
	  * gst/flacparse/Makefile.am:
	  * gst/flv/Makefile.am:
	  * gst/games/Makefile.am:
	  * gst/h264parse/Makefile.am:
	  * gst/librfb/Makefile.am:
	  * gst/mixmatrix/Makefile.am:
	  * gst/modplug/Makefile.am:
	  * gst/mpeg1sys/Makefile.am:
	  * gst/mpeg4videoparse/Makefile.am:
	  * gst/mpegdemux/Makefile.am:
	  * gst/mpegtsmux/Makefile.am:
	  * gst/mpegvideoparse/Makefile.am:
	  * gst/mve/Makefile.am:
	  * gst/nsf/Makefile.am:
	  * gst/nuvdemux/Makefile.am:
	  * gst/overlay/Makefile.am:
	  * gst/passthrough/Makefile.am:
	  * gst/pcapparse/Makefile.am:
	  * gst/playondemand/Makefile.am:
	  * gst/rawparse/Makefile.am:
	  * gst/real/Makefile.am:
	  * gst/rtjpeg/Makefile.am:
	  * gst/rtpmanager/Makefile.am:
	  * gst/scaletempo/Makefile.am:
	  * gst/sdp/Makefile.am:
	  * gst/selector/Makefile.am:
	  * gst/smooth/Makefile.am:
	  * gst/smoothwave/Makefile.am:
	  * gst/speed/Makefile.am:
	  * gst/speexresample/Makefile.am:
	  * gst/stereo/Makefile.am:
	  * gst/subenc/Makefile.am:
	  * gst/tta/Makefile.am:
	  * gst/vbidec/Makefile.am:
	  * gst/videodrop/Makefile.am:
	  * gst/videosignal/Makefile.am:
	  * gst/virtualdub/Makefile.am:
	  * gst/vmnc/Makefile.am:
	  * gst/y4m/Makefile.am:
	  * sys/acmenc/Makefile.am:
	  * sys/cdrom/Makefile.am:
	  * sys/dshowdecwrapper/Makefile.am:
	  * sys/dshowsrcwrapper/Makefile.am:
	  * sys/dvb/Makefile.am:
	  * sys/dxr3/Makefile.am:
	  * sys/fbdev/Makefile.am:
	  * sys/oss4/Makefile.am:
	  * sys/qcam/Makefile.am:
	  * sys/qtwrapper/Makefile.am:
	  * sys/vcd/Makefile.am:
	  * sys/wininet/Makefile.am:
	  * win32/common/config.h:
	  Don't install static libs for plugins. Fixes #550851 for -bad.

2008-09-02 09:56:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  [MOVED FROM BAD] Enable/fix up translations for these plugins.
	  Original commit message from CVS:
	  * ext/resindvd/plugin.c: (plugin_init):
	  * ext/resindvd/resindvdsrc.c:
	  * ext/twolame/gsttwolame.c: (plugin_init):
	  * gst/aiffparse/aiffparse.c: (plugin_init):
	  Enable/fix up translations for these plugins.
	  * po/LINGUAS:
	  Add 'ca' to LINGUAS.
	  * po/POTFILES.in:
	  * po/POTFILES.skip:
	  Add more files for translation and more files which tools
	  should skip.

2008-08-07 14:34:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD] ext/twolame/gsttwolame.*: Allow raw float samples as input for encoding.
	  Original commit message from CVS:
	  * ext/twolame/gsttwolame.c: (gst_two_lame_sink_setcaps),
	  (gst_two_lame_chain):
	  * ext/twolame/gsttwolame.h:
	  Allow raw float samples as input for encoding.

2008-08-02 17:39:13 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  [MOVED FROM BAD] Add TwoLAME MP2 encoding element, based on the LAME element.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/Makefile.am:
	  * ext/twolame/Makefile.am:
	  * ext/twolame/gsttwolame.c: (gst_two_lame_mode_get_type),
	  (gst_two_lame_padding_get_type), (gst_two_lame_emphasis_get_type),
	  (gst_two_lame_release_memory), (gst_two_lame_finalize),
	  (gst_two_lame_base_init), (gst_two_lame_class_init),
	  (gst_two_lame_src_setcaps), (gst_two_lame_sink_setcaps),
	  (gst_two_lame_init), (gst_two_lame_set_property),
	  (gst_two_lame_get_property), (gst_two_lame_sink_event),
	  (gst_two_lame_chain), (gst_two_lame_setup),
	  (gst_two_lame_change_state), (gst_two_lame_get_default_settings),
	  (plugin_init):
	  * ext/twolame/gsttwolame.h:
	  Add TwoLAME MP2 encoding element, based on the LAME element.

2009-03-09 23:12:33 +0000  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Automatic update of common submodule
	  From 7032163 to f8b3d91

2009-03-09 18:07:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbisdepay.c:
	  vorbisdepay: fix some leaks
	  And leak the codebooks.
	  Use glib base64 decoders.
	  Use subbuffers to avoid a memcpy of the headers.

2009-03-09 17:14:12 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: don't lose the first buffer after a seek
	  The flacdec API calls the write callback when performing a seek. We cannot yet
	  push out a buffer at that time so we must keep it and push it out later.
	  Flush out the upstream part of the pipeline when doing a seek.
	  Fixes #574275.

2009-03-09 15:20:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: sanitize tag names
	  Sanitize the tag names before turning them into a structure name. We can only
	  add alphanumeric values as the structure name.

2009-03-08 12:04:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From ffa738d to 7032163

2009-03-08 11:19:56 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 3f13e4e to ffa738d

2009-03-07 11:45:35 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 3c7456b to 3f13e4e

2009-03-07 10:45:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 57c83f2 to 3c7456b

2009-03-06 21:56:26 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: fix pads, so that they are subset of template caps
	  Do not add w=0 | h=0. When we can't get a framerate add fraction range.

2009-03-05 14:08:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: fix range parsing
	  Fix parsing of the range headers.

2009-02-10 17:20:57 +0000  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirendepay.h:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpsirenpay.h:
	  Move siren rtp pay/depay from gst-plugins-farsight

2009-03-04 16:25:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix memory leak in close
	  Close the connection even when we fail to send the teardown message.
	  Use the connection url (which is a copy of the src url).

2009-03-04 16:15:05 +0100  Peter Kjellerstedt <pkj@axis.com>

	* tests/check/Makefile.am:
	  check: gst-plugins-good.supp needs to be distributed.

2009-03-04 12:29:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix do-rtcp property description
	  ---

2009-03-03 12:20:27 +0100  Edward Hervey <bilboed@bilboed.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Expose the SoupSession 'timeout' property.

2009-03-02 15:07:24 +0100  Edward Hervey <bilboed@bilboed.com>

	* .gitignore:
	  .gitignore: Ignore the m4/ directory

2009-03-02 17:18:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmp4vpay.c:
	  rtpmp4vpay: Add support for more formats
	  Hack around short header mpeg4 video files and put the short header as the
	  config string.
	  Fixes #572551.

2009-03-02 16:08:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add support for http tunneling
	  Add support for http tunneling and a new rtsph:// uri for it.
	  See #573173.

2009-03-02 09:43:30 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	  Merge branch 'master' of ssh://thomasvs@git.freedesktop.org/git/gstreamer/gst-plugins-good

2009-03-02 08:41:15 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* ext/flac/gstflacdec.c:
	  Add/clarify/fix some logging.

2009-03-01 12:47:37 -0800  David Schleef <ds@hutch-2.local>

	* sys/osxvideo/Makefile.am:
	  Remove hardcoded definition of OBJC

2009-03-01 19:55:26 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	  Wait for a frame to become available before capturing it
	  Use GstPoll to wait for the fd of the video device to become readable before
	  trying to capture a frame. This speeds up stopping v4l2src a lot as it no
	  longer has to wait for the next frame, especially when capturing with low
	  framerates or when the video device just never generates a frame (which seems a
	  common issue for uvcvideo devices)
	  Fixes bug #563574.

2009-02-14 17:56:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/law/alaw-decode.c:
	* gst/law/mulaw-decode.c:
	  alawdec, mulawdec: demote some debug messages from ERROR to WARNING or DEBUG
	  Non-ok flow returns may happen for a variety of perfectly legitimate and expected reasons
	  (temporarily not linked, seeking, pipeline shutdown), so we really shouldn't spew ERROR
	  debug messages to stderr in those cases. Fixes #570781. (Seems like someone already took
	  care of some of these.)

2009-02-28 15:26:00 +0200  René Stadler <mail@renestadler.de>

	* gst/replaygain/gstrgvolume.c:
	  rgvolume: Improve log message for peak values >1.0 by clamping explicitly.

2009-02-27 23:25:32 -0800  David Schleef <ds@schleef.org>

	* ext/dv/gstdvdec.c:
	  Fix the field dominance
	  PAL is TFF, NTSC is BFF.  Some day I will learn to keep this
	  straight.

2009-02-27 20:40:31 +0100  LRN <lrn1986@gmail.com>

	* sys/directdraw/gstdirectdrawsink.c:
	  directdrawsink: Fix type mismatches
	  Fixes bug #573343.

2009-02-27 20:28:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	  Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good

2009-02-27 20:24:53 +0100  LRN <lrn1986@gmail.com>

	* gst/udp/gstudpnetutils.c:
	  udp: Don't set errno to EAFNOSUPPORT unconditionally
	  Fixes bug #573342.

2009-02-27 11:17:50 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/replaygain/gstrgvolume.c:
	  rgvolume: ignore out-of-range peak values
	  If the peak value is > 1 (and thus nonsensical) ignore it. Prevents
	  rgvolume reducing volume to effectively silent on files with bogus peak
	  values.

2009-02-27 13:29:41 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix SEEK event handling in push mode, and SEEKABLY query handling
	  Standard pull mode loop based SEEK handling fails in push mode,
	  so convert the SEEK event appropriately and dispatch to upstream.
	  Also cater for NEWSEGMENT event handling, and properly inform
	  downstream and application of SEEKABLE capabilities, depending
	  on scheduling mode and upstream.

2009-02-27 11:04:08 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Remove gst_util_dump_mem() calls.

2009-02-26 19:07:35 +0100  Julien Moutte <julien@fluendo.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix SEEK event handling in push mode
	  When in push mode we should not try to handle the SEEK event as there's
	  no code to handle it properly. Propagate upstream.

2009-02-26 19:05:06 +0100  Patrick Radizi <patrick dot radizi at axis dot com>

	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add the .h file change too
	  Add the .h file change for the new property.

2009-02-26 19:03:52 +0100  Patrick Radizi <patrick dot radizi at axis dot com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add property to disable RTCP
	  Some old servers don't like us doing RTCP and thus we need a property to disable
	  it. See #573173.

2009-02-26 13:19:31 +0100  Jan Smout <jan dot smout at gmail dot com>

	* gst/udp/gstudpnetutils.c:
	  udp: fix gst_udp_set_loop_ttl() again
	  Fix the gst_udp_set_loop_ttl() function that was commented out in a
	  previous commit. See #573115.

2009-02-26 13:06:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtpvrawdepay: fail on interlaced video
	  Fail on interlaced video until we support it.

2009-02-26 13:00:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: fail on interlaced video
	  Detect and fail when trying to payload interlaced video.

2009-02-25 20:47:15 -0800  David Schleef <ds@schleef.org>

	* Makefile.am:
	* configure.ac:
	* win32/common/config.h.in:
	  Change how win32/common/config.h is updated
	  Generate win32/common/config.h-new directly from config.h.in,
	  using shell variables in configure and some hard-coded information.
	  Change top-level makefile so that 'make win32-update' copies the
	  generated file to win32/common/config.h, which we keep in source
	  control.  It's kept in source control so that the git tree is
	  buildable from VS.
	  This change is similar to the one recently applied to GStreamer
	  and gst-plugins-good.  The previous config.h file in -good was in
	  pretty bad shape, so unlike core and base, I didn't attempt to
	  leave it strictly the same, but fixed it as necessary.  Needs
	  testing I cannot do myself.

2009-02-25 19:58:29 -0800  David Schleef <ds@schleef.org>

	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdec.h:
	  dvdec: Add interlacing info to caps and buffers

2009-02-25 14:57:33 +0000  Jan Schmidt <thaytan@noraisin.net>

	* common:
	* configure.ac:
	  build: Update shave init statement for changes in common. Bump common.

2009-02-25 14:01:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix compilation
	  Fix compilation on systems MSG_ERRQUEUE and IP_RECVERR.

2009-02-19 20:14:10 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: error out instead of crashing if no caps have been set
	  Don't crash if we receive a buffer without caps. Fixes #572413.

2009-02-25 11:35:31 +0100  Peter Kjellerstedt <pkj@axis.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Make sure the sockaddr length used for recvfrom() is big enough.
	  Previously the sockaddr length used for recvfrom() was calculated as
	  sizeof (struct sockaddr). However, this is too little to hold an IPv6
	  address, so the full size of the gst_sockaddr union should be used
	  instead.

2009-02-25 11:32:28 +0100  Peter Kjellerstedt <pkj@axis.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Unify the use of union gst_sockaddr.

2009-02-25 11:32:07 +0000  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Automatic update of common submodule
	  From 9cf8c9b to a6ce5c6

2009-02-25 12:05:22 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: avoid crashing on subtitles
	  Avoid a crash in avi with subtitles by only dereferencing the video description
	  when we actually are dealing with video in the _invert function.

2009-02-25 11:45:05 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  docs: various doc fixes
	  No short-desc as we have them in the element details.
	  Also keep things (Makefile.am and sections.txt) sorted.
	  Reword ambigous returns. No text after since please.

2009-02-24 17:58:32 +0000  Jan Schmidt <thaytan@noraisin.net>

	* gst/udp/gstudpsrc.c:
	  udp: Fix strict-aliasing warnings from gcc 4.4.0
	  Fix strict aliasing warnings by defining a union on the different
	  sockaddr structs that we need.

2009-02-24 17:35:46 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtp: Fix compiler warning in h264 payloader
	  Fix an undefined behaviour warning from gcc 4.4.0
	  Patch By: Tim-Philipp Müller <tim.muller@collabora.co.uk>
	  Fixes: #570995
	  Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>

2009-02-22 17:23:09 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* docs/plugins/Makefile.am:
	  Use shave for the build output

2009-02-24 14:55:28 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/gconf/Makefile.am:
	* ext/gconf/gstgconf.c:
	* ext/gconf/gstgconf.h:
	* ext/gconf/gstgconfelements.h:
	  gconf: Rename gconf.[ch] to gstgconf.[ch] to prevent name conflicts

2009-02-24 14:41:26 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: Also use "(c)inf" to fill the comment tag

2009-01-26 11:06:13 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: perform UDP SETUP according to MS RTSP spec
	  MS RTSP spec states that the UDP port pair used in subsequent SETUP
	  requests for various streams must be identical (since there will actually
	  be only 1 stream of muxed asf packets).  Following traditional specs and
	  using different port pairs in the SETUPs for separate streams will result
	  in all but the first one failing and only one stream being streamed.
	  So, in appropriate circumstances, retry UDP SETUP using previously used
	  port pair.  Fixes #552650.

2009-02-23 20:49:37 +0100  Aurelien Grimaud <gstelzz at yahoo dot fr>

	* gst/udp/gstudpsrc.c:
	  Read ICMP error messages instead of looping
	  When we are dealing with connected sockets shared between a udpsrc and a udpsink
	  we might receive ICMP connection refused error messages in udpsrc that will
	  cause it to go into a bursty loop because the poll returns right away without a
	  message to read.
	  Instead of looping, read the error message from the error queue in udpsrc.
	  Fixes #567857.

2009-02-23 19:53:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  Conditionally compile code for YVYU
	  Only compile the code for the YVYU format when the format is actually defined.
	  Spotted by tmatth on IRC.

2009-02-17 11:01:47 -0800  Levente Farkas <lfarkas@lfarkas.org>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: Make sort_by_frame_size conditionally compiled
	  sort_by_frame_size is declared static and only used inside
	  an ifdef, so use the same ifdef to define the function.  Fixes #572185
	  Signed-off-by: David Schleef <ds@schleef.org>

2009-02-23 17:05:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  Add YVYU format to caps
	  Add YVYU format to the caps. We don't have anything to handle these caps yet,
	  though.

2009-02-23 15:48:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  Some cleanups
	  Remove some unused variables.
	  Avoid a useless _resync call.
	  Correctly use a gboolean.

2009-02-23 15:43:51 +0100  Wai-Ming Ho <waiming at ailuropoda dot net>

	* gst/rtp/gstrtph264pay.c:
	  Always add PPS to the sprop-parameters-set
	  Rework the parsing code that under certain circumstances dropped the PPS from
	  the sprop-parameters-set.
	  Fixes #572854.

2009-02-23 12:14:23 +0100  Arnout Vandecappelle <arnout at mind dot be>

	* gst/matroska/matroska-mux.c:
	  Don't do crazy things with 0/1 framerates
	  We use 0/1 framerates to mark variable framerates and matroskamux should not try
	  to calculate a frame duration for it.
	  Fixes #571294.

2009-02-23 11:45:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  Require newer gst-p-b for the RTSP extensions.
	  --

2009-02-23 11:42:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  Call new receive_request method
	  Call the receive_request extension methods so that extensions can handle the
	  server request if they want.

2009-02-23 11:13:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspext.c:
	* gst/rtsp/gstrtspext.h:
	  Add method for hadling server requests
	  Add method to handle server requests on the list of RTSP extensions.

2009-02-13 14:39:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/law/alaw-decode.c:
	* gst/law/mulaw-decode.c:
	  Don't use GST_ERROR for non-error cases.
	  Turn a GST_ERROR line into a GST_DEBUG line so that we don't spam the log with
	  errors. Fixes #570781.

2009-02-22 19:30:32 +0100  Sjoerd Simons <sjoerd@luon.net>

	* ext/gconf/gstgconfvideosink.c:
	* ext/gconf/gstgconfvideosink.h:
	* ext/gconf/gstgconfvideosrc.c:
	* ext/gconf/gstgconfvideosrc.h:
	  gconfvideo(src|sink): Disconnect GConf notifications
	  Fixes bug #571321.

2009-02-22 19:25:39 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Unref the buffer and not the memory address of the buffer

2009-02-22 18:47:35 +0100  Olivier Crete <tester@tester.ca>

	* gst/law/alaw-decode.c:
	* gst/law/mulaw-decode.c:
	  alaw/mulaw: Implement _getcaps function for alaw/mulaw decoders
	  Fixes bug #572358.

2009-02-22 18:46:03 +0100  Olivier Crete <tester@tester.ca>

	* gst/law/alaw-encode.c:
	* gst/law/mulaw-encode.c:
	  alaw/mulaw: Don't require both, rate and channel, to be set in _getcaps
	  Fixes bug #572358.

2009-02-22 18:32:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix alignment issues by using GST_READ_*
	  Reading integers from random memory addresses will result
	  in SIGBUS on some architectures if the memory address
	  is not correctly aligned. This can happen at two
	  places in avidemux so we should use GST_READ_UINT32_LE
	  and friends here. Fixes bug #572256.

2009-02-22 18:08:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pulse/pulsemixerctrl.c:
	  pulsemixer: Don't use g_atomic_int_(get|set) for accessing the mixer track flags
	  g_atomic_int_(get|set) only work on ints and the flags are
	  an enum (which on most architectures is stored as an int).
	  Also the way the flags were accessed atomically would still
	  leave a possible race condition and we don't do it in any
	  other mixer track implementation, let alone at any other
	  place where an integer could be changed from different
	  threads. Removing the g_atomic_int_(get|set) will only
	  introduce a new race condition on architectures where
	  integers could be half-written while reading them
	  which shouldn't be the case for any modern architecture
	  and if we really care about this we need to use
	  g_atomic_int_(get|set) at many other places too.
	  Apart from that g_atomic_int_(set|get) will result in
	  aliasing warnings if their argument is explicitely
	  casted to an int *. Fixes bug #571153.

2009-02-22 15:52:06 +0000  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Automatic update of common submodule
	  From 5d7c9cc to 9cf8c9b

2009-02-22 12:41:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/raw1394/gsthdv1394src.c:
	  hdv1394src: Don't use void * pointer arithmetic

2009-02-21 11:13:43 -0800  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From 80c627d to 5d7c9cc

2009-02-21 18:42:46 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	  Back to development -> 0.10.14.1

2009-02-20 18:16:02 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  Document rtpdtmfdepay a bit

2009-02-20 17:41:37 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmf.c:
	  Moved dtmf elements from gst-plugins-farsight to -bad

2009-02-20 17:40:57 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfdepay.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  Fix up documentation blobs SGML

2009-02-20 17:37:43 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmf.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfcommon.h:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  Re-indent to Gst style

2009-02-18 13:30:44 -0500  Laurent Glayal <spglegle@yahoo.fr>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Missing format directive

2008-12-04 21:21:44 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	  [MOVED FROM GST-P-FARSIGHT] Allow setting a maximum duration to a RTP DTMF event

2008-12-04 21:11:17 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  [MOVED FROM GST-P-FARSIGHT] Improve the minimum quanta to make it impossible for the duration to fall down to 0

2008-12-01 18:31:48 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	  [MOVED FROM GST-P-FARSIGHT] Allow setting a minimum size of a sound quanta in the dtmf depayloader

2008-12-11 17:54:18 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/.git-darcs-dir:
	  [MOVED FROM GST-P-FARSIGHT] Remove .git-darcs-dir files

2008-12-01 17:37:10 -0500  Håvard Graff <havard.graff@tandberg.com>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  [MOVED FROM GST-P-FARSIGHT] Do wierd casting of the volume to make MSVC happy

2008-10-15 16:21:50 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Clarify the documentation of the "event-type" field when specifying dtmf events

2008-07-22 21:39:38 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Remove g_debugs
	  20080722213938-3e2dc-44a82d017fe66f3112301c410aa0b543de6156ad.gz

2008-06-13 23:57:23 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Take rate from the peers caps if possible
	  20080613235723-3e2dc-15690ee42708c539e1be12e20e076a5613faea96.gz

2008-06-13 23:41:44 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Put the sample rate in dtmfsrc into a variable
	  20080613234144-3e2dc-e60070943bec829b703b8821c7aa4351a02deebe.gz

2008-06-13 23:30:06 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Take the clock-rate from the caps in rtpdtmfsrc
	  20080613233006-3e2dc-a7d4e918643f4f8c1bb2cc2678558c654025920e.gz

2008-04-28 22:22:37 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/Makefile.am:
	  [MOVED FROM GST-P-FARSIGHT] Link modules with libm where required
	  20080428222237-3e2dc-b1e9120c1e9ca1a510bfd7c27e2d45f0d4a12504.gz

2008-04-12 23:44:18 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfdepay.c:
	  [MOVED FROM GST-P-FARSIGHT] Fix byte ordering issues with dtmfsrc and rtpdtmfdepay.. use of G_STRINGIFY to avoid error on MSVC
	  20080412234418-4f0f6-4828d1613dfcd564afd236dfc8fb57a299092f83.gz

2008-03-20 19:14:38 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	  [MOVED FROM GST-P-FARSIGHT] Fix copyrights again, per smcv's advice..
	  20080320191438-4f0f6-671c9db5d996a4601df017ceab4af6d16469c966.gz

2008-03-19 21:17:31 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Make it clear that dtmfsrc also takes named events as input
	  20080319211731-3e2dc-26c729f6dc8db27e71cf6b22646a81530dbf862f.gz

2008-03-20 18:48:41 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  [MOVED FROM GST-P-FARSIGHT] debug message made into errors because that's what they are...
	  20080320184841-4f0f6-8a2d283297b02713dade0ae4acaa5f6e0f67eace.gz

2008-03-20 18:39:37 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  [MOVED FROM GST-P-FARSIGHT] Clean unused stuff...
	  20080320183937-4f0f6-bcb841cdc07f9e9677512f4b50b4b659a58c6783.gz

2008-03-20 18:39:12 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	  [MOVED FROM GST-P-FARSIGHT] Fix copyrights
	  20080320183912-4f0f6-689365d5a406632e3d088fac74e4fb6f8a4eb0ea.gz

2008-03-20 01:13:01 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/Makefile.am:
	* gst/dtmf/gstdtmf.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Adding support for rtpdtmfdepay
	  20080320011301-4f0f6-d36a5d24be20336e36c4796d75476c9b5ee1a7e1.gz

2008-03-19 19:32:51 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] encoding name has to be upper-case
	  20080319193251-3e2dc-1581b33be9b486e35ec4948009677ccd5ffdc098.gz

2008-03-20 00:51:47 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfcommon.h:
	* gst/dtmf/gstrtpdtmfdepay.c:
	* gst/dtmf/gstrtpdtmfdepay.h:
	  [MOVED FROM GST-P-FARSIGHT] Adding necessary files for rtpdtmfdepay
	  20080320005147-4f0f6-550fe22f70152f3aab3dcd7a6b02cbf81e89232d.gz

2008-03-20 00:50:41 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Fix typos
	  20080320005041-4f0f6-9d22fa5d155e35b605ea85b1fd9e7197a882a1f0.gz

2008-02-16 13:41:40 +0000  Sjoerd Simons <sjoerd@luon.net>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] dtmfsrc: Correctly set the endianess in the caps to the machines endianess
	  20080216134140-93b9a-40a3a9d7ac1679c5e0dfd24a6b91e4aba6cc6496.gz

2007-09-17 17:52:33 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Search&Replace oops
	  20070917175233-3e2dc-57f579c4b890993f49fa8e9e6470a3eb79d2b922.gz

2007-09-17 17:51:33 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] events dont yet belong in the caps
	  20070917175133-3e2dc-fd1d83b7826b898110fc571ae7c3440f1887434d.gz

2007-09-17 16:08:20 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Add patch to make it work with maemo dsp sources that payload incorrectly
	  20070917160820-3e2dc-06b1b1d1b0918b30dabea5a0714cb732b3b8d8dd.gz

2007-09-17 04:26:49 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Oops, set to no preroll when playing->paused too
	  20070917042649-3e2dc-94adb6aa0617e815a6e233232dabb4bbc48dc82c.gz

2007-09-17 00:36:54 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Complete port to basesrc
	  20070917003654-3e2dc-db0f84dabd9dd1ac929a0461865b8aaa8ef91a77.gz

2007-09-17 00:24:12 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Add caps negotiation function
	  20070917002412-3e2dc-ca266816e9629746e9083c5bb8b7f73b94a9b2b0.gz

2007-09-17 00:16:59 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Properly free non-start events
	  20070917001659-3e2dc-a571777e3ecfb90989f87412f554aa10a31cc2ca.gz

2007-09-17 00:15:52 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Make interval and packet_redundancy into uint
	  20070917001552-3e2dc-60032e547b3669b87317c981d985c156aab91b40.gz

2007-09-16 19:44:08 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Make the rtp dtmf src use basesrc
	  20070916194408-3e2dc-734000130dce2434a014acf843d641ff0e60aa5a.gz

2007-09-16 19:41:01 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Make dtmf src code nicer
	  20070916194101-3e2dc-a8be8c509c65400d1d3962da02e67d15d2054316.gz

2007-09-14 04:20:42 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Implement stopping in a nice thread safe way
	  20070914042042-3e2dc-1fe257ff4b72aca4b0eb5f285a14650b8df268c3.gz

2007-09-14 04:18:34 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Remove get_times (Wim says its only good for really fake sources)
	  20070914041834-3e2dc-fff4d5da2a145f19e7b610a1027d2c4d4bc5eae0.gz

2007-09-13 21:21:45 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] using the unlock method of basesrc
	  20070913212145-4f0f6-0e438a681bf1651c0cc0d8fa3269aed3f1668b6b.gz

2007-09-13 21:12:26 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] more debug
	  20070913211226-4f0f6-bc32b5828fc8e0323c8a6eee779a38145aacd593.gz

2007-09-13 20:46:14 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] added debugs
	  20070913204614-4f0f6-68c2a69ae7a1efca6e13c116dbad7f9b686f0242.gz

2007-09-13 19:20:53 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Make sure to unlock the thread when going to ready and to flush the queue when moving to paused or playing
	  20070913192053-4f0f6-76c3925380d1a30988286170535a65dea64a5583.gz

2007-09-13 17:55:20 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Changed dtmfsrc into a subclass of GstBaseSrc
	  20070913175520-4f0f6-16ca4bf93690072f3e836d1c8a5b52cf7a421916.gz

2007-09-04 22:57:53 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Add another fix for a possible race condition
	  20070904225753-4f0f6-5ba8c4260c002bb27eb98e9faba3c15799357b57.gz

2007-09-04 21:52:24 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Add comment to explain push back
	  20070904215224-3e2dc-d92ac1f403dcf571546a7c53f18809f840eea51d.gz

2007-09-04 20:55:09 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Properly do the locking to avoid race conditions with clock unscheduling
	  20070904205509-3e2dc-da19900b51af6aedb6547f4f392bef4d1061dec2.gz

2007-09-01 00:03:24 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] oups, I did it again...
	  20070901000324-4f0f6-3d8b46691ee520537b06c511a5e732f5b812b844.gz

2007-08-31 23:54:28 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] oups, sorry.. DTMF, not RTP_DTMF for this file...
	  20070831235428-4f0f6-00b606bfb4892e4f217c440b611cc794ab0de55a.gz

2007-08-31 23:44:13 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Fixes the deadlock when pausing the dtmfsrc and rtpdtmfsrc. Had to push something on the async queue to release the blocking async_queue_pop(). Thanks to Olivier for the solution.
	  20070831234413-4f0f6-793cf35fc43636e7275258cc7063fc068f5efa0a.gz

2007-08-28 22:15:34 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] ClockID when waiting for buffer is now unscheduled when stopping the task. Various fixes to avoid bugs (thanks to -Wall -Werror). Fixes to allow the merge of the branch.
	  20070828221534-4f0f6-b0d6a4fe48c4e2a16b9ff69cb310087c970ce48e.gz

2007-08-28 17:15:46 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Cleaned up the code a bit, no use of GST_* and return value verification from gst_*
	  20070828171546-4f0f6-bdeb4b1b7f99f9464aabe5c43bd4a4d2025262b6.gz

2007-08-27 19:56:10 +0000  Olivier Crete <olivier.crete@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Fix overly long lines and tabs
	  20070827195610-3e2dc-396a3fa01e16f184e4109c71fe2deb6e516bdf0d.gz

2007-08-27 19:26:18 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] untabbified dtmfsrc
	  20070827192618-4f0f6-77d68070464f1b5f9a46cb6eec2d922340143c04.gz

2007-08-27 17:24:24 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Fix RTP timestamps by sending a new_segment event to the payloader
	  20070827172424-4f0f6-d20907e3d436d50bfe74eb4fc3d2d6d7b6b6dbc5.gz

2007-08-27 17:23:39 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Better handling of packets, we send the same duration for all packets to avoid huge packets when min duration defines are modified.
	  20070827172339-4f0f6-cc93304437ea376fff6458c74c46c19f6920d329.gz

2007-08-27 17:23:22 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] Changing minimum values to work better on some gateways
	  20070827172322-4f0f6-5bf2bffa59a8244538dced795fa7d7649452ca91.gz

2007-08-22 20:16:53 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] The DTMF tone generator now respects the volume argument passed in the event
	  20070822201653-4f0f6-8b7ff874006e11f5a74d0fd91e5a9a43cd082ada.gz

2007-08-22 18:01:33 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] don't know why I did that...
	  20070822180133-4f0f6-6a7382f6c7d3630f91da384e1904763c7ea6fa1a.gz

2007-08-22 17:55:33 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Ported the event queue work from dtmfsrc to rtpdtmfsrc
	  Added a queue based system for the rtpdtmfsrc. Now it waits for start/stop messages on the queue, and makes sure that the minimum duty cycle (120ms) is respected between each
	  tone, including inter-digit silence.
	  20070822175533-4f0f6-f27414c406f1f7b00c9a9084a988cf3a7930fe5c.gz

2007-08-22 17:54:44 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	  [MOVED FROM GST-P-FARSIGHT] ouch, printing with arguments but without %s.. that made it segfault a few times...
	  20070822175444-4f0f6-445ea6ce7a9668d04cf999af772a504ec74fb67a.gz

2007-08-22 17:51:26 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Moved the timestamp from the event to dtmfsrc structure since we have only one event at a time, so let's keep it stored in the dtmfsrc struct
	  20070822175126-4f0f6-53bcda2bd8ae8c56d29e62e69ac19a30e08ad350.gz

2007-08-20 20:38:26 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Added a queue based system for the dtmfsrc. Now it waits for start/stop messages on the queue, and makes sure that the minimum duty cycle (120ms) is respected between each tone, including inter-digit silence.
	  20070820203826-4f0f6-750a22b612a5e495e767666934465c34fe32074b.gz

2007-08-20 18:48:52 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/Makefile.am:
	* gst/dtmf/gstdtmf.c:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Added dtmfsrc, a DTMF Tone Generator, and made it part of the 'dtmf' plugin.
	  20070820184852-4f0f6-a0d85e67708290aebafa89ab79d3cedd5815b620.gz

2007-08-20 18:48:00 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/dtmf/.git-darcs-dir:
	* gst/dtmf/Makefile.am:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  [MOVED FROM GST-P-FARSIGHT] Moved rtpdtmf to dtmf directory
	  20070820184800-4f0f6-fa33ea974510161de8c9951c39087af3613b65a4.gz

2009-02-21 12:47:00 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* ext/flac/gstflacdec.c:
	  respect DEFAULT segment by clipping the last buffer to be sent

=== release 0.10.14 ===

2009-02-19 20:09:07 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.14

2009-02-19 20:07:41 +0000  Jan Schmidt <thaytan@noraisin.net>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2009-02-19 13:16:39 +0000  Jan Schmidt <thaytan@noraisin.net>

	* gst/audiofx/audioecho.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosrc.c:
	  Update Since: tags in autodetect srcs and audioecho

2009-02-19 11:12:58 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ChangeLog:
	  Update ChangeLog for 0.10.13.3

2009-02-19 11:09:03 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* win32/common/config.h:
	  0.10.13.3 pre-release

2009-02-10 11:25:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsemixerctrl.c:
	  pulsemixer: Fix compiler warnings.
	  Cast (enum *) to (int *), not necessarily technically right,
	  but plugs #571153.

2009-02-13 18:03:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Issue property change notification in streaming thread, rather than PA thread.
	  pa_threaded_mainloop_lock() (a.o.) and by extension get_property should
	  not be done from a PA thread, but the latter may occur as a result of a
	  property change notification.  Fixes #571204 (though current situation
	  not ideal, e.g. post message rather than signal).

2009-02-10 11:27:51 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/videocrop/gstaspectratiocrop.c:
	  aspectratiocrop: Don't forget to call parent finalize implementation.
	  This fixes a memory leak (leaking the contained elements of the bin).

2009-02-10 08:43:59 +0100  Edward Hervey <bilboed@bilboed.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: Fix build. Fixes #571038

2009-02-09 12:18:36 +0100  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Bump revision to use for common submodule.

2009-02-07 16:00:49 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ChangeLog:
	  ChangeLog: Update ChangeLog for 0.10.13.2

2009-02-07 15:58:55 +0000  Jan Schmidt <thaytan@noraisin.net>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: Update translations for 0.10.13.2

2009-02-07 15:46:07 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* win32/common/config.h:
	  Release 0.10.13.2

2009-02-07 15:40:53 +0000  Jan Schmidt <thaytan@noraisin.net>

	* po/LINGUAS:
	* po/mt.po:
	  po: Add Maltese translation

2009-02-06 16:16:05 -0800  David Schleef <ds@schleef.org>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_dump.c:
	* gst/qtdemux/qtdemux_dump.h:
	* gst/qtdemux/qtdemux_fourcc.h:
	* gst/qtdemux/qtdemux_types.c:
	  qtdemux: Add handling for stps atoms
	  stps atoms contain "partial sync" information, which means that it's
	  a sync point where pts != dts.  This is needed to properly handle
	  MPEG2, H.264, Dirac, etc., in quicktime.

2009-02-05 15:51:42 -0800  Michael Smith <msmith@songbirdnest.com>

	* ext/flac/gstflacdec.c:
	  flacdec: if we aborted reading, don't do into an infinite loop.
	  If our read callback ran out of data, so had to abort reading, we return
	  GST_FLOW_ERROR instead of going into an infinite loop.

2009-02-05 10:19:37 -0800  Michael Smith <msmith@songbirdnest.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: remove non-embedded mode and fix memory management.
	  Remove non-embedded mode. Embed mode becomes default and only mode.
	  embed property is retained for binary compatibility.
	  Added autorelease pools around all objc functions that might be called
	  from a non-main thread.

2009-02-05 20:02:01 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* ext/flac/gstflacdec.c:
	  debug on the object

2009-02-04 16:40:13 -0800  Michael Smith <msmith@songbirdnest.com>

	* sys/osxaudio/gstosxringbuffer.c:
	  osxaudio fixes: multichannel and changing caps.
	  Ensure we create the ringbuffer segment size as a multiple of the
	  bytes per sample (fixes 6-channel output).
	  Reset the segoffset when acquiring the ringbuffer, so we don't retain
	  a bogus offset when caps change.

2009-02-04 11:38:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Keep track of connected state
	  Keep track of the state of the connection and don't try to send TEARDOWN when
	  the server has closed the connection.

2009-02-04 09:20:28 +0100  Robin Stocker <robin@nibor.org>

	* gst/matroska/matroska-demux.c:
	  Read Matroska Title element for the TITLE tag
	  Not all Matroska files have a Tags element which contains
	  information about the title among other things. Most video
	  Matroska files only contain the Title element so we
	  should parse this too. Fixes bug #570435.

2009-02-03 22:34:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure.ac: bump core/base requirements to released versions

2009-02-03 17:10:30 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/audioecho.c:
	  Fix audioecho unit test on 32 bit systems
	  Cast the new value for the "delay" property to GstClockTime.
	  Integers without type are passed to vararg functions with
	  an integer type that can hold a pointer.

2009-02-03 14:09:26 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: Don't reset frequency bands from user settings. Fixes #570343.
	  Move reallocating the history buffer out of _compute_frequencies() and call the
	  right function as needed. Add some logging and tweak the formatting of existing
	  logging. Simplify setting need_new_coefficients when changing properties.

2009-02-03 11:52:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioecho.c:
	  Use guint64 instead of guint for storing guint64

2009-02-02 18:37:35 +0100  Jonathan Matthew <notverysmart@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	  Use correct flag for the GNOME proxy configuration
	  Fixes bug #552140.

2009-02-02 13:08:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/icles/v4l2src-test.c:
	  Fix compiler warnings
	  fix compiler warnings due to unused return values of scanf.

2009-01-31 11:08:30 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/icles/v4l2src-test.c:
	  Fix format string compiler warning

2009-01-30 22:24:14 +0200  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  Add releaseinfo with online url.

2009-01-30 18:04:11 +0000  Jan Schmidt <jan.schmidt@sun.com>

	* tests/check/Makefile.am:
	* tests/icles/Makefile.am:
	  Fix up some compile flags

2009-01-30 17:35:49 +0000  Jan Schmidt <jan.schmidt@sun.com>

	* gst/videocrop/gstvideocrop.c:
	  Don't use Glib 2.16 function g_strcmp0.

2009-01-30 17:34:45 +0000  Jan Schmidt <jan.schmidt@sun.com>

	* gst/qtdemux/qtdemux.c:
	  Don't do void pointer arithmetic

2009-01-30 17:26:19 +0000  Jan Schmidt <jan.schmidt@sun.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  Fix Forte compiler warnings.
	  Don't do void pointer arithmetic. Don't have an unreachable statement.

2009-01-30 17:29:45 +0000  Jan Schmidt <thaytan@noraisin.net>

	* common:
	  Bump common

2009-01-26 10:33:55 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/avi/gstavidemux.c:
	  Remove useless processing for non-raw formats

2009-01-30 15:34:31 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	* gst/qtdemux/qtdemux_types.c:
	  Add support for the 'Requirement' and 'Encoder' tags

2009-01-30 15:33:19 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  Modify private-tag name formatter so that it doesn't go mad at fourcc starting with '(c)'.

2009-01-30 14:40:51 +0100  Brijesh Singh <brijesh.ksingh@gmail.com>

	* sys/v4l2/gstv4l2tuner.c:
	  Fix comparison of the tuner norms
	  The V4L2 tuner norms that a device supports could
	  be a subset of some norm (e.g. NTSC instead of NTSC_M).
	  The comparison should be done by & instead of ==.
	  See http://www.linuxtv.org/downloads/video4linux/API/V4L2_API/spec-single/v4l2.html#STANDARD
	  Fixes bug #569820.

2009-01-30 08:53:06 +0100  Edward Hervey <bilboed@bilboed.com>

	* autogen.sh:
	* common:
	  Use a symbolic link for the pre-commit client-side hook

2009-01-29 14:08:56 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/videocrop/gstaspectratiocrop.c:
	  Only unref the peer when there is one.

2009-01-29 11:07:59 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	* gst/interleave/deinterleave.c:
	* gst/interleave/interleave.c:
	* sys/directdraw/gstdirectdrawsink.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/v4l2/gstv4l2src.c:
	* sys/waveform/gstwaveformsink.c:
	  Remove version numbers from a few gst-launch examples.
	  The majority of the examples doe not use -0.10 and this will also help us to maintain the docs.

2009-01-29 10:10:08 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/directdraw/gstdirectdrawsink.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstossmixerelement.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	  Update and add documentation for platform specific plugins (sys).
	  Link to properties. Correct titles for examples. Fix examples.

2009-01-29 09:45:25 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/multipart/multipartmux.c:
	  Add ' to framerate argument and remove the word 'simple' as all our pipelines are apparently simple.

2009-01-29 09:42:56 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	  Add examples for the jpeg elements.

2009-01-28 21:40:11 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ext/pulse/pulsesink.c:
	  Fix compile error in the last commit

2009-01-28 20:34:40 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* ext/pulse/pulseprobe.c:
	* ext/pulse/pulseprobe.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  Rewrite the pulse plugin, conditionally enabling new behaviour with newer pulseaudio.
	  Fixes: #567794
	  * Hook pulsesink's volume property up with the stream volume -- not the
	  sink volume in PA.
	  * Read the device description directly from the sink instead of going
	  via the mixer.
	  * Properly implement _reset() methods for both sink and source to avoid
	  deadlocks when shutting down a pipeline.
	  * Replace all simple pa_threaded_mainloop_wait() by proper loops to
	  guarantee that we wait for the right event in case multiple events are
	  fired.  While this is not strictly necessary in many cases it
	  certainly is more correct and makes me sleep better at night.
	  * Replace CHECK_DEAD_GOTO macros with proper functions
	  * Extend the number of supported channels to 32 since that is the actual
	  limit in PA.
	  * Get rid of _dispose() methods since we don't need them.
	  * Increase the volume property upper limit of the sink to 1000.
	  * Reset function pointers after we disconnect a stream/context. Better
	  fix for bug 556986.
	  * Reset the state of the element properly if open/prepare fails
	  * Cork the PA stream when the pipeline is paused. This allows the PA
	  * daemon to
	  close audio device on pause and thus save a bit of power.
	  * Set PA stream properties based on GST tags such as GST_TAG_TITLE,
	  GST_TAG_ARTIST, and so on.
	  Signed-off-by: Lennart Poettering <lennart@poettering.net>

2009-01-28 17:46:06 +0200  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* ext/aalib/gstaasink.c:
	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttimeoverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/esd/esdmon.c:
	* ext/esd/esdsink.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/gconf/gstgconfaudiosink.c:
	* ext/gconf/gstgconfaudiosrc.c:
	* ext/gconf/gstgconfvideosink.c:
	* ext/gconf/gstgconfvideosrc.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/hal/gsthalaudiosink.c:
	* ext/hal/gsthalaudiosrc.c:
	* ext/hal/hal.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libcaca/gstcacasink.h:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/pulse/pulsemixer.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackparse.c:
	* gst/matroska/matroska-mux.h:
	* gst/udp/gstudpsrc.c:
	  Update and add documentation for plugins with deps (ext).
	  Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered. Fix warnings that gtk-doc points out.

2009-01-28 15:57:20 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioecho.c:
	* gst/audiofx/audioecho.h:
	  Limit the delay by a new max-delay property
	  Introduce a new max-delay property that can only
	  be set before going to PLAYING or PAUSED. This
	  is used to limit the maximum delay and is set
	  to the current delay by default.
	  Using this will make sure that we have enough data
	  in our internal ringbuffer for the echo. With dynamic
	  reallocation of the ringbuffer as used before silence
	  could've been used as the echo directly after setting
	  a new delay.

2009-01-28 11:58:42 +0100  Edward Hervey <bilboed@bilboed.com>

	* win32/common/config.h:
	  Revert previous bogus commit

2009-01-28 12:29:42 +0200  Stefan Kost <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audioamplify.c:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiodynamic.c:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiokaraoke.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/auparse/gstauparse.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/cutter/gstcutter.c:
	* gst/debug/gstpushfilesrc.c:
	* gst/debug/gsttaginject.c:
	* gst/debug/progressreport.c:
	* gst/equalizer/gstiirequalizer10bands.c:
	* gst/equalizer/gstiirequalizer3bands.c:
	* gst/equalizer/gstiirequalizernbands.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/law/mulaw.c:
	* gst/level/gstlevel.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/qtdemux/qtdemux.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/spectrum/gstspectrum.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videomixer/videomixer.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* win32/common/config.h:
	  Update and add documentation for plugins with no deps (gst).
	  Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.

2009-01-27 23:09:05 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/examples/spectrum/demo-audiotest.c:
	* tests/examples/spectrum/demo-osssrc.c:
	  Fix example apps by drawing in the main-loop.

2009-01-27 20:33:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	  tests: fix build of aspectratio crop unit test in uninstalled environment.

2009-01-27 20:30:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* .gitignore:
	  Make git ignore backup files

2009-01-26 16:14:47 +0100  Peter Kjellerstedt <pkj@axis.com>

	* gst/multipart/multipartdemux.c:
	  Plug a memory leak in a debug message.

2009-01-22 15:59:40 +0100  Peter Kjellerstedt <pkj@axis.com>

	* gst/udp/gstudpnetutils.c:
	  Correct return value from gst_udp_get_addr() when no known family is found.

2009-01-26 09:51:36 +0100  Jonathan Matthew <jonathan@d14n.org>

	* configure.ac:
	* ext/soup/gstsouphttpsrc.c:
	  Use libsoup-gnome for proxy configuration if available
	  If libsoup-gnome is found use this as it will give us
	  the GNOME proxy configuration. Otherwise use normal
	  libsoup.
	  The GNOME proxy configuration will only be used if
	  the proxy properties are not set on souphttpsrc
	  and if the http_proxy environment variable is not
	  set.
	  Fixes bug #552140.

2009-01-25 19:26:46 -0800  David Schleef <ds@schleef.org>

	* gst/qtdemux/qtdemux.c:
	  Add a few more video fourcc's

2009-01-24 14:48:00 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/videocrop/gstaspectratiocrop.c:
	* tests/check/Makefile.am:
	* tests/check/elements/aspectratiocrop.c:
	  Add unit test for aspectratiocrop Fixes bug #527951
	  Add unit test for aspectratiocrop and refactor this element. Added
	  finalize function to cleanup leaking mutex.

2009-01-25 14:34:09 +0000  Jan Schmidt <thaytan@noraisin.net>

	* tests/check/elements/.gitignore:
	  Ignore check binaries

2009-01-24 18:28:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioecho.c:
	  Save some allocations if the echo delay is increased often
	  Save some allocations if the echo delay is increased often
	  during playback by always allocating enough memory to hold
	  data up to the next complete second, i.e. in the worst case
	  allocate memory for one additional second.

2009-01-24 14:25:08 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update plugin version in documentation

2009-01-23 21:47:40 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/videocrop/gstvideocrop.c:
	  Fix link in documentation of videocrop element

2009-01-23 21:46:13 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* gst/videocrop/gstaspectratiocrop.c:
	  Add documentation for aspectratiocrop

2009-01-24 13:21:39 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* win32/common/config.h:
	  Update win32/common/config.h for the new development cycle

2009-01-24 11:53:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audioecho.c:
	  Add note that audioecho's reverb sounds metallic
	  Add a note to the docs that audioecho's reverb will
	  sound metallic. This happens because for a real
	  reverb filter additional filtering is necessary.
	  Also note which values should be used for the delay
	  property to get an echo effect.

2009-01-23 23:38:10 +0000  Jan Schmidt <thaytan@noraisin.net>

	* .gitignore:
	* docs/plugins/.gitignore:
	* po/.gitignore:
	* tests/examples/audiofx/.gitignore:
	  More entries for the gitignores

2009-01-23 20:36:27 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* tests/check/elements/videocrop.c:
	  skip video/x-raw-gray in videocrop unit test
	  A recent commit added video/x-raw-gray support to videocrop. However
	  this lets the videocrop unit test fail. Because videotestsrc can't
	  generate this format.

2009-01-23 15:39:46 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/videocrop/Makefile.am:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstaspectratiocrop.h:
	* gst/videocrop/gstvideocrop.c:
	  Add aspectratiocrop element. Fixes bug #527951
	  Add new aspectratiocrop element that crops the video
	  to a specified aspect ratio using videocrop.

2009-01-23 10:49:28 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/videocrop/gstvideocrop.c:
	  Fix navigation event forwarding while cropping. Fixes bug #567992.
	  Fix the navigation event forwarding while cropping by adjusting
	  the mouse position by the amount of cropped pixels.

2009-01-23 10:04:39 +0100  Brian Cameron <brian.cameron@sun.com>

	* configure.ac:
	  Fix linking on Solaris. Fixes bug #568809.
	  Check for the socket library which is needed
	  for socket() on Solaris.

2009-01-22 22:41:43 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	  Bump version number again -> 0.10.13.1

2009-01-22 22:41:01 +0000  Jan Schmidt <thaytan@noraisin.net>

	* gst-plugins-good.doap:
	  Add releases 0.10.12 and 0.10.13 to the doap file

2009-01-22 18:08:50 +0200  Stefan Kost <ensonic@users.sf.net>

	* common:
	  Update common snapshot.

2009-01-22 14:25:07 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* win32/common/config.h:
	  Back to devel -> 0.10.12.1

2009-01-22 01:29:40 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	  Release 0.10.12

2009-01-21 17:22:39 -0800  David Schleef <ds@schleef.org>

	* gst/qtdemux/qtdemux.c:
	  Fix for security advisory TKADV2009-0xx
	  Fix potential buffer overflows while reading quicktime headers.
	  Security issue noticed by Tobias Klein.

2009-01-21 12:56:55 +0000  Jan Schmidt <thaytan@noraisin.net>

	* ext/flac/gstflacdec.c:
	  Fix typo and small flaw in flac decoder

2009-01-22 13:49:35 +0100  Sebastian Dröge <slomo@circular-chaos.org>

	* common:
	  Fix pre-commit hook

2009-01-22 10:40:34 +0100  Sebastian Dröge <slomo@circular-chaos.org>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* gst/audiofx/Makefile.am:
	* gst/audiofx/audioecho.c:
	* gst/audiofx/audioecho.h:
	* gst/audiofx/audiofx.c:
	* tests/check/Makefile.am:
	* tests/check/elements/audioecho.c:
	  Rename audioreverb to audioecho. Fixes bug #568395.
	  The element can add an echo and a simple reverb effect to
	  an audio stream but for a real reverb filter it would need
	  some additional filtering to prevent a metallic-sounding
	  result.

2009-01-22 12:21:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  Free leftover udp ports (if any) when a setup request fails.

2009-01-22 06:05:26 +0100  Edward Hervey <bilboed@bilboed.com>

	* autogen.sh:
	* common:
	  Install and use pre-commit indentation hook from common

2009-01-21 13:25:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  Whitespace fixes and some improved debug lines.

2009-01-21 04:31:58 +0100  Edward Hervey <bilboed@bilboed.com>

	* autogen.sh:
	  autogen.sh : Use git submodule

2009-01-20 15:33:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/gstv4l2src.c: Fix error code (the message string also needs love, but not today).
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read):
	  Fix error code (the message string also needs love, but not today).

2009-01-19 11:44:36 +0000  Luotao Fu <l.fu@pengutronix.de>

	  gst/videocrop/gstvideocrop.c: Add 8bit grayscale support to videocrop plugin. Fixes #567952.
	  Original commit message from CVS:
	  Patch by: Luotao Fu <l dot fu at pengutronix dot de>
	  * gst/videocrop/gstvideocrop.c:
	  (gst_video_crop_get_image_details_from_caps):
	  Add 8bit grayscale support to videocrop plugin. Fixes #567952.

2009-01-19 11:22:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/audioreverb.c: Set the default value in the instance init function.
	  Original commit message from CVS:
	  * gst/audiofx/audioreverb.c: (gst_audio_reverb_init):
	  Set the default value in the instance init function.

2009-01-19 11:19:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add an echo/reverb filter to the audiofx plugin, with configurable echo delay, intensity and feedback. Fixes bug #567...
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofx.c: (plugin_init):
	  * gst/audiofx/audioreverb.c: (gst_audio_reverb_base_init),
	  (gst_audio_reverb_class_init), (gst_audio_reverb_init),
	  (gst_audio_reverb_finalize), (gst_audio_reverb_set_property),
	  (gst_audio_reverb_get_property), (gst_audio_reverb_setup),
	  (gst_audio_reverb_stop), (gst_audio_reverb_transform_ip):
	  * gst/audiofx/audioreverb.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/audioreverb.c: (setup_reverb),
	  (cleanup_reverb), (GST_START_TEST), (audioreverb_suite):
	  Add an echo/reverb filter to the audiofx plugin, with configurable
	  echo delay, intensity and feedback. Fixes bug #567874.

2009-01-19 10:13:53 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/spectrum/gstspectrum.*: Implement a simple compensation algorithm for rounding errors.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state),
	  (gst_spectrum_transform_ip):
	  * gst/spectrum/gstspectrum.h:
	  Implement a simple compensation algorithm for rounding errors.
	  This makes sure that a spectrum message is posted on the bus
	  every interval nanoseconds. Fixes bug #567955.

2009-01-15 21:16:45 +0000  Michael Smith <msmith@xiph.org>

	  sys/osxaudio/Makefile.am: Link against CoreServices (needed for osx 10.4) and fix up the linker flags. Fixes #567853.
	  Original commit message from CVS:
	  * sys/osxaudio/Makefile.am:
	  Link against CoreServices (needed for osx 10.4) and fix up the linker
	  flags. Fixes #567853.

2009-01-15 14:53:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Catch invalid and commonly wrong playback rates in the elst atoms.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_segments):
	  Catch invalid and commonly wrong playback rates in the elst atoms.
	  Fixes #567800.

2009-01-15 11:40:23 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/spectrum/gstspectrum.c: Don't call gst_fft_f32_free() with NULL to prevent a crash. Fixes bug #567642.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state):
	  Don't call gst_fft_f32_free() with NULL to prevent a
	  crash. Fixes bug #567642.

2009-01-14 15:44:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/spectrum/gstspectrum.*: Use correct types for frame/fft counters and some minor cleanup.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip):
	  * gst/spectrum/gstspectrum.h:
	  Use correct types for frame/fft counters and some minor
	  cleanup.

2009-01-14 15:37:07 +0000  Lennart Poettering <lennart@poettering.net>

	  ext/pulse/pulseprobe.c: Fix refcount loop, resulting in a thread leak. Fixes bug #567746.
	  Original commit message from CVS:
	  Patch by: Lennart Poettering <lennart at poettering dot net>
	  * ext/pulse/pulseprobe.c: (gst_pulseprobe_new),
	  (gst_pulseprobe_free):
	  Fix refcount loop, resulting in a thread leak. Fixes bug #567746.

2009-01-14 10:46:54 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/spectrum/: Post a spectrum message on the bus for every interval, even if the interval is small than the length o...
	  Original commit message from CVS:
	  * gst/spectrum/Makefile.am:
	  * gst/spectrum/README:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
	  (gst_spectrum_class_init), (gst_spectrum_init),
	  (gst_spectrum_reset_state), (gst_spectrum_finalize),
	  (gst_spectrum_set_property), (gst_spectrum_start),
	  (gst_spectrum_stop), (gst_spectrum_setup),
	  (gst_spectrum_transform_ip):
	  * gst/spectrum/gstspectrum.h:
	  Post a spectrum message on the bus for every interval, even
	  if the interval is small than the length of the FFT.
	  Fixes bug #567642.
	  Major cleanup of the spectrum element.

2009-01-13 19:23:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add audioiirfilter and audiofirfilter elements which allow generic IIR/FIR filters to be implemented by providing the...
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofirfilter.c: (gst_audio_fir_filter_base_init),
	  (gst_audio_fir_filter_class_init),
	  (gst_audio_fir_filter_update_kernel), (gst_audio_fir_filter_init),
	  (gst_audio_fir_filter_setup), (gst_audio_fir_filter_finalize),
	  (gst_audio_fir_filter_set_property),
	  (gst_audio_fir_filter_get_property):
	  * gst/audiofx/audiofirfilter.h:
	  * gst/audiofx/audiofx.c: (plugin_init):
	  * gst/audiofx/audioiirfilter.c: (gst_audio_iir_filter_base_init),
	  (gst_audio_iir_filter_class_init),
	  (gst_audio_iir_filter_update_coefficients),
	  (gst_audio_iir_filter_init), (gst_audio_iir_filter_setup),
	  (gst_audio_iir_filter_finalize),
	  (gst_audio_iir_filter_set_property),
	  (gst_audio_iir_filter_get_property):
	  * gst/audiofx/audioiirfilter.h:
	  Add audioiirfilter and audiofirfilter elements which allow
	  generic IIR/FIR filters to be implemented by providing the
	  filter coefficients. Fixes bug #567577.
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.signals:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-video4linux2.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  Add documentation for the audioiirfilter and audiofirfilter
	  elements.
	  * tests/check/Makefile.am:
	  * tests/check/elements/audiofirfilter.c: (on_message),
	  (on_rate_changed), (on_handoff), (GST_START_TEST),
	  (audiofirfilter_suite):
	  * tests/check/elements/audioiirfilter.c: (on_message),
	  (on_rate_changed), (on_handoff), (GST_START_TEST),
	  (audioiirfilter_suite):
	  * tests/examples/Makefile.am:
	  * tests/examples/audiofx/Makefile.am:
	  * tests/examples/audiofx/firfilter-example.c: (on_message),
	  (on_rate_changed), (main):
	  * tests/examples/audiofx/iirfilter-example.c: (on_message),
	  (on_rate_changed), (main):
	  Add unit tests and example applications for the two filter
	  elements.

2009-01-13 19:09:19 +0000  Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br>

	  gst/qtdemux/qtdemux.c: Fix format string for guint64.
	  Original commit message from CVS:
	  Patch by: Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br>
	  * gst/qtdemux/qtdemux.c:
	  Fix format string for guint64.

2009-01-13 19:04:09 +0000  Michael Smith <msmith@xiph.org>

	  sys/osxaudio/Makefile.am: osxaudio plugin now requires AudioUnit framework, so link against that.
	  Original commit message from CVS:
	  * sys/osxaudio/Makefile.am:
	  osxaudio plugin now requires AudioUnit framework, so link against that.
	  Clean up tabs v spaces while I'm there.

2009-01-13 17:49:07 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/rtp/server-alsasrc-PCMA.c: Add some example code for printing the RTP manager stats.
	  Original commit message from CVS:
	  * tests/examples/rtp/server-alsasrc-PCMA.c: (print_source_stats),
	  (print_stats), (main):
	  Add some example code for printing the RTP manager stats.

2009-01-13 08:24:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Use a custom mutex for protecting the instance fields instead of the GstObject lock. Using the latter c...
	  Original commit message from CVS:
	  * gst/audiofx/audiochebband.c: (gst_audio_cheb_band_class_init),
	  (gst_audio_cheb_band_init), (gst_audio_cheb_band_finalize),
	  (gst_audio_cheb_band_set_property):
	  * gst/audiofx/audiochebband.h:
	  * gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_class_init),
	  (gst_audio_cheb_limit_init), (gst_audio_cheb_limit_finalize),
	  (gst_audio_cheb_limit_set_property):
	  * gst/audiofx/audiocheblimit.h:
	  * gst/audiofx/audiowsincband.c: (gst_audio_wsincband_class_init),
	  (gst_audio_wsincband_init), (gst_audio_wsincband_finalize),
	  (gst_audio_wsincband_set_property):
	  * gst/audiofx/audiowsincband.h:
	  * gst/audiofx/audiowsinclimit.c: (gst_audio_wsinclimit_class_init),
	  (gst_audio_wsinclimit_init), (gst_audio_wsinclimit_finalize),
	  (gst_audio_wsinclimit_set_property):
	  * gst/audiofx/audiowsinclimit.h:
	  Use a custom mutex for protecting the instance fields instead of
	  the GstObject lock. Using the latter can lead to deadlocks, especially
	  with the FIR filters when updating the latency.

2009-01-11 19:03:38 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Implement a base class for generic audio FIR filters.
	  Original commit message from CVS:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofxbasefirfilter.c:
	  (gst_audio_fx_base_fir_filter_dispose),
	  (gst_audio_fx_base_fir_filter_base_init),
	  (gst_audio_fx_base_fir_filter_class_init),
	  (gst_audio_fx_base_fir_filter_init),
	  (gst_audio_fx_base_fir_filter_push_residue),
	  (gst_audio_fx_base_fir_filter_setup),
	  (gst_audio_fx_base_fir_filter_transform),
	  (gst_audio_fx_base_fir_filter_start),
	  (gst_audio_fx_base_fir_filter_stop),
	  (gst_audio_fx_base_fir_filter_query),
	  (gst_audio_fx_base_fir_filter_query_type),
	  (gst_audio_fx_base_fir_filter_event),
	  (gst_audio_fx_base_fir_filter_set_kernel):
	  * gst/audiofx/audiofxbasefirfilter.h:
	  * gst/audiofx/audiofxbaseiirfilter.c:
	  Implement a base class for generic audio FIR filters.
	  * gst/audiofx/audiowsincband.c:
	  (gst_gst_audio_wsincband_mode_get_type),
	  (gst_gst_audio_wsincband_window_get_type),
	  (gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
	  (gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
	  (gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
	  (gst_audio_wsincband_get_property):
	  * gst/audiofx/audiowsincband.h:
	  * gst/audiofx/audiowsinclimit.c:
	  (gst_audio_wsinclimit_mode_get_type),
	  (gst_audio_wsinclimit_window_get_type),
	  (gst_audio_wsinclimit_base_init),
	  (gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
	  (gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
	  (gst_audio_wsinclimit_set_property),
	  (gst_audio_wsinclimit_get_property):
	  * gst/audiofx/audiowsinclimit.h:
	  * tests/check/elements/audiowsincband.c: (GST_START_TEST):
	  * tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
	  Use this new base class for audiowsincband and audiowsinclimit.
	  Also cleanup both elements.

2009-01-08 18:17:13 +0000  Michael Smith <msmith@xiph.org>

	  gst/qtdemux/qtdemux.c: In push mode, error out if we get EOS before we've created any srcpads.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c:
	  In push mode, error out if we get EOS before we've created any srcpads.
	  Handle (in pull mode) some files that have a truncated moov atom where
	  the final sub-atom is a 'free' atom and the contents of that are not
	  present in the file.

2009-01-08 15:56:46 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/matroska/: Some cleanups, refactoring and minor enhancements in caps handling.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps):
	  Some cleanups, refactoring and minor enhancements in caps handling.
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
	  (gst_matroska_mux_init), (gst_matroska_pad_reset),
	  (gst_matroska_pad_free), (gst_matroska_mux_reset),
	  (gst_matroska_mux_video_pad_setcaps),
	  (gst_matroska_mux_request_new_pad):
	  * tests/check/elements/matroskamux.c: (teardown_src_pad):
	  Only remove, release or reset what is appropriate upon state change.

2009-01-07 20:38:50 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/pulse/pulsesink.*: Use a mutex to protect the current stream pointer, and ignore callbacks for stream objects tha...
	  Original commit message from CVS:
	  * ext/pulse/pulsesink.c:
	  * ext/pulse/pulsesink.h:
	  Use a mutex to protect the current stream pointer, and ignore
	  callbacks for stream objects that have been destroyed already.
	  Fixes problems with unprepare/prepare cycles caused by the input
	  caps changing, without reintroducing bug #556986.

2009-01-07 16:09:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/v4l2/gstv4l2src.c: Remove () from translateable string, so that it makes more sense.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c:
	  Remove () from translateable string, so that it makes more sense.

2009-01-07 09:43:13 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/gstavimux.c: Minor fix/cleanup in header field calculation.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_audsink_set_caps):
	  Minor fix/cleanup in header field calculation.

2009-01-06 17:48:10 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/matroska/matroska-mux.*: Remove internal taglist and fully use tagsetter interface.
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
	  (gst_matroska_mux_handle_sink_event), (gst_matroska_mux_finish):
	  * gst/matroska/matroska-mux.h:
	  Remove internal taglist and fully use tagsetter interface.

2009-01-06 14:50:29 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/gstavimux.*: Ensure header size invariance during subsequent rewrite by using tags snapshot.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_reset),
	  (gst_avi_mux_riff_get_avi_header):
	  * gst/avi/gstavimux.h:
	  Ensure header size invariance during subsequent rewrite by using
	  tags snapshot.

2009-01-05 17:31:13 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/pulse/pulsesink.c: Don't wait for the pulse mainloop when destroying the stream.
	  Original commit message from CVS:
	  * ext/pulse/pulsesink.c: (gst_pulsesink_destroy_stream):
	  Don't wait for the pulse mainloop when destroying the stream.
	  Fixes a deadlock when the pulsedaemon goes away while pulsesink
	  is PLAYING. Fixes bug #556986.

2009-01-05 12:30:40 +0000  Sascha Hauer <s.hauer@pengutronix.de>

	  sys/v4l2/gstv4l2src.c: Add support for grayscale v4l2 devices. Fixes bug #566616.
	  Original commit message from CVS:
	  Patch by: Sascha Hauer <s dot hauer at pengutronix dot de>
	  Luotao Fu <l dot fu at pengutronix dot de>
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure),
	  (gst_v4l2_get_caps_info):
	  Add support for grayscale v4l2 devices. Fixes bug #566616.

2009-01-05 11:42:09 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/qtdemux/: Streamline tag handling and pass unparsed tags as binary blob in private tag.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_tag_add_str),
	  (qtdemux_tag_add_tmpo), (qtdemux_tag_add_covr),
	  (qtdemux_tag_add_date), (qtdemux_tag_add_gnre),
	  (qtdemux_tag_add_blob), (qtdemux_parse_udta):
	  * gst/qtdemux/qtdemux.h:
	  * gst/qtdemux/quicktime.c: (plugin_init):
	  Streamline tag handling and pass unparsed tags as binary blob
	  in private tag.

2009-01-05 10:13:29 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Implement a base class for IIR filters.
	  Original commit message from CVS:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofxbaseiirfilter.c:
	  (gst_audio_fx_base_iir_filter_base_init),
	  (gst_audio_fx_base_iir_filter_dispose),
	  (gst_audio_fx_base_iir_filter_class_init),
	  (gst_audio_fx_base_iir_filter_init),
	  (gst_audio_fx_base_iir_filter_calculate_gain),
	  (gst_audio_fx_base_iir_filter_set_coefficients),
	  (gst_audio_fx_base_iir_filter_setup), (process),
	  (gst_audio_fx_base_iir_filter_transform_ip),
	  (gst_audio_fx_base_iir_filter_stop):
	  * gst/audiofx/audiofxbaseiirfilter.h:
	  Implement a base class for IIR filters.
	  * gst/audiofx/audiochebband.c: (gst_audio_cheb_band_base_init),
	  (gst_audio_cheb_band_class_init), (gst_audio_cheb_band_init),
	  (generate_coefficients), (gst_audio_cheb_band_set_property),
	  (gst_audio_cheb_band_setup):
	  * gst/audiofx/audiochebband.h:
	  * gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_base_init),
	  (gst_audio_cheb_limit_class_init), (gst_audio_cheb_limit_init),
	  (generate_coefficients), (gst_audio_cheb_limit_set_property),
	  (gst_audio_cheb_limit_setup):
	  * gst/audiofx/audiocheblimit.h:
	  Use the IIR filter base class for the chebyshev filters.

2009-01-02 20:39:34 +0000  Justin Karnegas <justin@affinix.com>

	  sys/osxaudio/: Rewrite osxaudio to work more flexibly and more reliably, using a different abstraction layer of corea...
	  Original commit message from CVS:
	  Patch by: Justin Karnegas <justin@affinix.com> and
	  Michael Smith <msmith@songbirdnest.com>
	  * sys/osxaudio/gstosxaudio.c:
	  * sys/osxaudio/gstosxaudioelement.c:
	  * sys/osxaudio/gstosxaudioelement.h:
	  * sys/osxaudio/gstosxaudiosink.c:
	  * sys/osxaudio/gstosxaudiosink.h:
	  * sys/osxaudio/gstosxaudiosrc.c:
	  * sys/osxaudio/gstosxaudiosrc.h:
	  * sys/osxaudio/gstosxringbuffer.c:
	  * sys/osxaudio/gstosxringbuffer.h:
	  Rewrite osxaudio to work more flexibly and more reliably, using a
	  different abstraction layer of coreaudio that is the recommended way of
	  doing low-level audio I/O on OSX.
	  Fixes byg #564948.

2009-01-02 16:31:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/rtp/server-decodebin-H263p-AMR.sh: Add example RTP transcoding pipeline from any file decodedable with...
	  Original commit message from CVS:
	  * tests/examples/rtp/server-decodebin-H263p-AMR.sh:
	  Add example RTP transcoding pipeline from any file decodedable with
	  uridecodebin.

2009-01-02 15:20:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/rtp/: Add two C examples of using gstrtpbin as a sender and a receiver.
	  Original commit message from CVS:
	  * tests/examples/rtp/.cvsignore:
	  * tests/examples/rtp/Makefile.am:
	  * tests/examples/rtp/client-PCMA.c: (pad_added_cb), (main):
	  * tests/examples/rtp/server-alsasrc-PCMA.c: (main):
	  Add two C examples of using gstrtpbin as a sender and a receiver.

2008-12-31 11:20:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ChangeLog: Remove conflict marker from ChangeLog
	  Original commit message from CVS:
	  * ChangeLog:
	  Remove conflict marker from ChangeLog

2008-12-28 09:50:31 +0000  j^ <j@oil21.org>

	  gst/qtdemux/qtdemux.c: Add codec mapping for xvid, fmp4 and ac3 tracks.
	  Original commit message from CVS:
	  Patch by: j^ <j at oil21.org>
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps),
	  (qtdemux_audio_caps):
	  Add codec mapping for xvid, fmp4 and ac3 tracks.
	  Fixes #565850

2008-12-23 12:10:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/jpeg/gstsmokeenc.*: Implement getcaps function.
	  Original commit message from CVS:
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init),
	  (gst_smokeenc_getcaps), (gst_smokeenc_setcaps),
	  (gst_smokeenc_chain), (gst_smokeenc_change_state):
	  * ext/jpeg/gstsmokeenc.h:
	  Implement getcaps function.
	  Set caps on the pad and on all outgoing buffers.
	  Fixes #565441.

2008-12-19 09:36:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/pulse/pulsemixerctrl.c: And remove temporary comment pointing to the bug ticket.
	  Original commit message from CVS:
	  * ext/pulse/pulsemixerctrl.c:
	  And remove temporary comment pointing to the bug ticket.
	  * gst/avi/gstavimux.c:
	  Move reoccuring logging to LOG and log instance too.

2008-12-17 17:28:39 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/pulse/pulsemixerctrl.c: Don't leak the pa_operation.
	  Original commit message from CVS:
	  * ext/pulse/pulsemixerctrl.c:
	  Don't leak the pa_operation.

2008-12-16 16:19:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Require core cvs.
	  Original commit message from CVS:
	  * configure.ac:
	  Require core cvs.

2008-12-16 16:07:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: Rename api from _flush to _reset_tags.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Rename api from _flush to _reset_tags.

2008-12-16 14:22:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: Use new tagsetter api to flush tags.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Use new tagsetter api to flush tags.

2008-12-16 13:14:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/deinterleave.c: Increase timeout to 3 minutes to prevent timeouts.
	  Original commit message from CVS:
	  * tests/check/elements/deinterleave.c: (deinterleave_suite):
	  Increase timeout to 3 minutes to prevent timeouts.

2008-12-16 12:52:24 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/interleave.c: Increase timeout to 3 minutes to prevent timeouts.
	  Original commit message from CVS:
	  * tests/check/elements/interleave.c: (interleave_suite):
	  Increase timeout to 3 minutes to prevent timeouts.

2008-12-16 11:57:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.*: Totally remove the internal taglists and fully use tagsetter.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  * gst/avi/gstavimux.h:
	  Totally remove the internal taglists and fully use tagsetter.

2008-12-15 15:59:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: Instead of filtering wrongly just use the mergemode. Applications is use KEEP_ALL if they want t...
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Instead of filtering wrongly just use the mergemode. Applications is
	  use KEEP_ALL if they want to supress tag-events. Fixes #563221 for
	  avi for real (I hope). Everyone chime in, before I fix the others.

2008-12-15 12:45:35 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/pulse/pulsemixerctrl.c: Add note about memleak.
	  Original commit message from CVS:
	  * ext/pulse/pulsemixerctrl.c:
	  Add note about memleak.

2008-12-13 16:23:09 +0000  Edward Hervey <bilboed@bilboed.com>

	  m4/Makefile.am: A couple more .m4 that aren't shipped anymore with gettext 0.17.
	  Original commit message from CVS:
	  * m4/Makefile.am:
	  A couple more .m4 that aren't shipped anymore with gettext 0.17.

2008-12-13 15:34:01 +0000  Edward Hervey <bilboed@bilboed.com>

	  Switch to using GstStaticPadTemplate.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_base_init),
	  (gst_flac_dec_init):
	  * gst/law/alaw-decode.c: (gst_alaw_dec_base_init),
	  (gst_alaw_dec_init):
	  * gst/law/alaw-encode.c: (gst_alaw_enc_base_init),
	  (gst_alaw_enc_init):
	  * gst/law/alaw.c: (plugin_init):
	  * gst/law/mulaw-decode.c: (gst_mulawdec_base_init),
	  (gst_mulawdec_init):
	  * gst/law/mulaw-encode.c: (gst_mulawenc_base_init),
	  (gst_mulawenc_init):
	  * gst/law/mulaw.c: (plugin_init):
	  Switch to using GstStaticPadTemplate.
	  * gst/udp/gstudpnetutils.c: (gst_udp_get_addr):
	  Don't forget to free the addrinfo structure.
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
	  (gst_wavparse_sink_activate):
	  Don't forget to unref the GstAdapter.

2008-12-13 12:58:24 +0000  Edward Hervey <bilboed@bilboed.com>

	  m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
	  Original commit message from CVS:
	  * m4/Makefile.am:
	  inttypes.m4 hasn't been available since gettext-0.15, and since we now
	  require gettext >= 0.17 ... we can remove it from the list of files to
	  dist.

2008-12-10 15:03:23 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  smaller spec file updates
	  Original commit message from CVS:
	  smaller spec file updates

2008-12-09 17:55:22 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: More logging.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  More logging.
	  * gst/avi/gstavimux.c:
	  Handle more metadata fields. Better estimate of metadata size. Don't
	  merge received tags, if application has specified tags using
	  GST_TAG_MERGE_REPLACE_ALL. Fixes #563221 for avi.

2008-12-09 14:30:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/Makefile.am: Also ignore pulsemixer for the states unit test.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Also ignore pulsemixer for the states unit test.

2008-12-09 14:19:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpjpegdepay.c: Add an EOI marker at the end of the jpeg frame when it's missing.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpjpegdepay.c: (gst_rtp_jpeg_depay_process):
	  Add an EOI marker at the end of the jpeg frame when it's missing.
	  Fixes #563056.

2008-12-09 10:47:14 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/videocrop.c: Update the unit test for the new color values for BT.601 red.
	  Original commit message from CVS:
	  * tests/check/elements/videocrop.c: (check_1x1_buffer):
	  Update the unit test for the new color values for BT.601 red.
	  Fixes bug #563510.

2008-12-09 10:28:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/dv/gstdvdemux.c: Restore previous behaviour of not passing QoS and navigation events upstream, which presumably w...
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_src_event):
	  Restore previous behaviour of not passing QoS and navigation
	  events upstream, which presumably wasn't meant to be changed.

2008-12-09 09:39:53 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/dv/gstdvdemux.c: Add srcpads only when needed and remove them again when going back to READY. This prevents stall...
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_add_video_pad),
	  (gst_dvdemux_add_audio_pad), (gst_dvdemux_remove_pads),
	  (gst_dvdemux_demux_audio), (gst_dvdemux_demux_video),
	  (gst_dvdemux_chain), (gst_dvdemux_loop),
	  (gst_dvdemux_change_state):
	  Add srcpads only when needed and remove them again when going
	  back to READY. This prevents stalled pipelines if there's no
	  audio inside the DV stream, which happens for many MXF files.

2008-12-09 09:09:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/souphttpsrc.c: The ports in libsoup are unsigned integers and not signed integers.
	  Original commit message from CVS:
	  * tests/check/elements/souphttpsrc.c: (GST_START_TEST),
	  (run_server):
	  The ports in libsoup are unsigned integers and not signed
	  integers.

2008-12-08 18:31:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/dv/gstdvdemux.c: Forward all events upstream unless it's something we really don't handle. This fixes latency con...
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_src_event):
	  Forward all events upstream unless it's something we really
	  don't handle. This fixes latency configuration of pipelines.

2008-12-08 18:24:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/dv/: Really call dv_init() exactly one time, not one time for the demuxer and one time for the decoder.
	  Original commit message from CVS:
	  * ext/dv/gstdv.c: (plugin_init):
	  * ext/dv/gstdvdec.c: (gst_dvdec_class_init):
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_class_init):
	  Really call dv_init() exactly one time, not one time for
	  the demuxer and one time for the decoder.

2008-12-08 12:37:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4apay.c: Copy incomming timestamp to outgoing packets.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_handle_buffer):
	  Copy incomming timestamp to outgoing packets.

2008-12-08 12:36:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4vpay.c: Don't try to push packets before we could find a valid config startcode. Fixes #563509.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush),
	  (gst_rtp_mp4v_pay_event):
	  Don't try to push packets before we could find a valid config
	  startcode. Fixes #563509.

2008-12-07 19:22:48 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/gstsunaudiomixerctrl.c: Set the mixer fd before calling ioctl() on it. Fixes bug #563414.
	  Original commit message from CVS:
	  Patch by: Brian Cameron <brian.cameron at sun dot com>
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_open):
	  Set the mixer fd before calling ioctl() on it. Fixes bug #563414.

2008-12-07 19:01:35 +0000  Alexandre Rostovtsev <tetromino@gmail.com>

	  configure.ac: Make usage of libv4l optional by a configure parameter.
	  Original commit message from CVS:
	  Patch by: Alexandre Rostovtsev <tetromino at gmail dot com>
	  * configure.ac:
	  Make usage of libv4l optional by a configure parameter.
	  Fixes bug #563504.

2008-12-05 09:24:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add documentation for matroskamux and matroskademux and update the inspection xml files.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/inspect/plugin-1394.xml:
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gamma.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-goom2k1.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-interleave.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-monoscope.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multifile.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-pulseaudio.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-shout2send.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-soup.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-video4linux2.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  * gst/matroska/matroska-demux.c:
	  * gst/matroska/matroska-demux.h:
	  * gst/matroska/matroska-mux.c:
	  * gst/matroska/matroska-mux.h:
	  Add documentation for matroskamux and matroskademux and
	  update the inspection xml files.

2008-12-04 20:10:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
	  Original commit message from CVS:
	  * configure.ac:
	  Apparently AC_CONFIG_MACRO_DIR breaks when using more
	  than one macro directory, reverting last change.

2008-12-04 19:47:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
	  Original commit message from CVS:
	  * configure.ac:
	  Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
	  our M4 macros.

2008-11-30 16:24:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/udp/gstmultiudpsink.c: Provide the parameters that are required for the format string to fix a compiler warning.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
	  Provide the parameters that are required for the format string
	  to fix a compiler warning.

2008-11-29 20:05:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/autodetect/gstautoaudiosrc.c: Fix classification.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosrc.c:
	  Fix classification.

2008-11-29 13:31:55 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
	  Original commit message from CVS:
	  Patch by: Cygwin Ports maintainer
	  <yselkowitz at users dot sourceforge dot net>
	  * autogen.sh:
	  * configure.ac:
	  Require gettext 0.17 because older versions don't mix with libtool
	  2.2. At build time an older gettext version will still work.
	  Fixes bug #556091.

2008-11-28 15:10:50 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/udp/gstmultiudpsink.c: Make gst_multiudpsink_render() ignore errors from sendto() instead of breaking streaming. ...
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt <pkj at axis com>
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
	  Make gst_multiudpsink_render() ignore errors from sendto() instead of
	  breaking streaming. Emit a warning instead. Fixes #562572.

2008-11-27 16:43:24 +0000  Ron McOuat <rmcouat@smartt.com>

	  Add support for basic and digest authentication in souphttpsrc.
	  Original commit message from CVS:
	  Patch by: Ron McOuat <rmcouat at smartt dot com>
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init),
	  (gst_soup_http_src_init), (gst_soup_http_src_dispose),
	  (gst_soup_http_src_set_property), (gst_soup_http_src_get_property),
	  (gst_soup_http_src_authenticate_cb), (gst_soup_http_src_start):
	  * ext/soup/gstsouphttpsrc.h:
	  * tests/check/elements/souphttpsrc.c: (basic_auth_cb),
	  (digest_auth_cb), (run_test), (GST_START_TEST),
	  (souphttpsrc_suite), (run_server):
	  Add support for basic and digest authentication in souphttpsrc.
	  Fixes bug #561775.

2008-11-27 12:13:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavenc/: Add support for a-law and mu-law encoded wav files. Fixes bug #562434.
	  Original commit message from CVS:
	  Patch by: Pepijn Van Eeckhoudt
	  <pepijn dot vaneeckhoudt at luciad dot com>
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	  (gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
	  * gst/wavenc/gstwavenc.h:
	  * gst/wavenc/riff.h:
	  Add support for a-law and mu-law encoded wav files. Fixes bug #562434.

2008-11-27 11:22:56 +0000  이문형 <iwings@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Prevent further read/write actions taken to the connect-failed socket by erroring out quickly....
	  Original commit message from CVS:
	  Patch by: 이문형 <iwings at gmail dot com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
	  Prevent further read/write actions taken to the connect-failed socket by
	  erroring out quickly. See #562258.

2008-11-26 21:19:47 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/examples/level/level-example.c: Set fakesink to sync. Otherwise people might question the message interval. Nev...
	  Original commit message from CVS:
	  * tests/examples/level/level-example.c:
	  Set fakesink to sync. Otherwise people might question the message
	  interval. Nevertheless the timestamp in the message is what matters.

2008-11-25 18:13:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/icles/.cvsignore: cvsignore newly generated file.
	  Original commit message from CVS:
	  * tests/icles/.cvsignore:
	  cvsignore newly generated file.

2008-11-25 18:03:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Fix the descriptions and fix some email addresses.
	  Original commit message from CVS:
	  * gst/rtp/gstasteriskh263.c:
	  * gst/rtp/gstasteriskh263.h:
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps):
	  * gst/rtp/gstrtpL16depay.h:
	  * gst/rtp/gstrtpL16pay.c:
	  * gst/rtp/gstrtpL16pay.h:
	  * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps):
	  * gst/rtp/gstrtpac3depay.h:
	  * gst/rtp/gstrtpamrdepay.c:
	  * gst/rtp/gstrtpamrdepay.h:
	  * gst/rtp/gstrtpamrpay.c:
	  * gst/rtp/gstrtpamrpay.h:
	  * gst/rtp/gstrtpdepay.c:
	  * gst/rtp/gstrtpdepay.h:
	  * gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps):
	  * gst/rtp/gstrtpg726depay.c:
	  * gst/rtp/gstrtpg726pay.c:
	  * gst/rtp/gstrtpg729depay.c:
	  * gst/rtp/gstrtpg729pay.c:
	  * gst/rtp/gstrtpgsmdepay.c:
	  * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps):
	  * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps):
	  * gst/rtp/gstrtph263depay.h:
	  * gst/rtp/gstrtph263pay.c:
	  * gst/rtp/gstrtph263pay.h:
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
	  * gst/rtp/gstrtph263pdepay.h:
	  * gst/rtp/gstrtph263ppay.c:
	  * gst/rtp/gstrtph263ppay.h:
	  * gst/rtp/gstrtph264depay.c:
	  * gst/rtp/gstrtph264depay.h:
	  * gst/rtp/gstrtph264pay.c:
	  * gst/rtp/gstrtph264pay.h:
	  * gst/rtp/gstrtpilbcdepay.c:
	  * gst/rtp/gstrtpilbcpay.c:
	  * gst/rtp/gstrtpjpegdepay.h:
	  * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps):
	  * gst/rtp/gstrtpmp1sdepay.h:
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	  * gst/rtp/gstrtpmp2tdepay.h:
	  * gst/rtp/gstrtpmp2tpay.c:
	  * gst/rtp/gstrtpmp2tpay.h:
	  * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps):
	  * gst/rtp/gstrtpmp4apay.c:
	  * gst/rtp/gstrtpmp4apay.h:
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps):
	  * gst/rtp/gstrtpmp4gdepay.h:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmp4gpay.h:
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps):
	  * gst/rtp/gstrtpmp4vdepay.h:
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event):
	  * gst/rtp/gstrtpmp4vpay.h:
	  * gst/rtp/gstrtpmpadepay.c:
	  * gst/rtp/gstrtpmpadepay.h:
	  * gst/rtp/gstrtpmpapay.c:
	  * gst/rtp/gstrtpmpapay.h:
	  * gst/rtp/gstrtpmpvdepay.c:
	  * gst/rtp/gstrtpmpvdepay.h:
	  * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
	  * gst/rtp/gstrtppcmapay.c:
	  * gst/rtp/gstrtppcmudepay.c:
	  * gst/rtp/gstrtppcmupay.c:
	  * gst/rtp/gstrtpspeexdepay.c:
	  * gst/rtp/gstrtpspeexpay.c:
	  * gst/rtp/gstrtpsv3vdepay.c:
	  * gst/rtp/gstrtpsv3vdepay.h:
	  * gst/rtp/gstrtptheoradepay.c:
	  * gst/rtp/gstrtptheoradepay.h:
	  * gst/rtp/gstrtptheorapay.c:
	  * gst/rtp/gstrtptheorapay.h:
	  * gst/rtp/gstrtpvorbisdepay.c:
	  * gst/rtp/gstrtpvorbisdepay.h:
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
	  * gst/rtp/gstrtpvorbispay.h:
	  * gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps):
	  * gst/rtp/gstrtpvrawpay.c:
	  Fix the descriptions and fix some email addresses.

2008-11-25 17:47:24 +0000  Julien Moutte <julien@moutte.net>

	  gst/qtdemux/qtdemux.c: Add MPG1 and MPG2 fourcc to supported qtdemux video codecs as I found some video clips using t...
	  Original commit message from CVS:
	  2008-11-25  Julien Moutte  <julien@fluendo.com>
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add MPG1 and MPG2
	  fourcc
	  to supported qtdemux video codecs as I found some video clips
	  using
	  those.

2008-11-25 16:26:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/autodetect/: Post an error when we can't set the internal ghostpad target.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
	  * gst/autodetect/gstautoaudiosrc.c: (gst_auto_audio_src_detect):
	  * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
	  (gst_auto_video_sink_detect):
	  * gst/autodetect/gstautovideosrc.c: (gst_auto_video_src_detect):
	  Post an error when we can't set the internal ghostpad target.

2008-11-25 16:06:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videocrop/gstvideocrop.*: Fix renegotiation when changing properties using the new basetransform features. Fixes ...
	  Original commit message from CVS:
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_init),
	  (gst_video_crop_transform), (gst_video_crop_transform_caps),
	  (gst_video_crop_set_caps), (gst_video_crop_set_property):
	  * gst/videocrop/gstvideocrop.h:
	  Fix renegotiation when changing properties using the new basetransform
	  features. Fixes #561502.
	  * tests/icles/Makefile.am:
	  * tests/icles/videocrop2-test.c: (make_pipeline), (main):
	  Add crazy interactive test unit for dynamically changing properties.

2008-11-24 12:20:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Add some more debugging.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (new_session_pad),
	  (gst_rtspsrc_parse_range):
	  Add some more debugging.
	  Use the reanges received from the server unconditionally.
	  Fixes #561625.

2008-11-23 15:08:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/pulse/pulsesink.c: Change #if 0 to something more expresive and add pointer to related bug ticket.
	  Original commit message from CVS:
	  * ext/pulse/pulsesink.c:
	  Change #if 0 to something more expresive and add pointer to related
	  bug ticket.

2008-11-23 11:17:01 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	* ChangeLog:
	  ChangeLog surgery
	  Original commit message from CVS:
	  ChangeLog surgery

2008-11-23 11:14:42 +0000  Tal Shalif <tshalif@nargila.org>

	  gst/qtdemux/qtdemux.c: Use G_{BIG,LITTLE}_ENDIAN instead of the non-GLib variants as the latter don't exist on some s...
	  Original commit message from CVS:
	  Patch by: Tal Shalif <tshalif at nargila dot org>
	  * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
	  Use G_{BIG,LITTLE}_ENDIAN instead of the non-GLib variants as
	  the latter don't exist on some systems (mingw). Fixes bug #561992.

2008-11-21 13:43:29 +0000  Zeeshan Ali <zeeshanak@gnome.org>

	  ext/soup/gstsouphttpsrc.c: Add transferMode.dnla.org header to HTTP requests as this is required by the DLNA specs an...
	  Original commit message from CVS:
	  Patch by: Zeeshan Ali <zeeshanak at gnome dot org>
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_build_message):
	  Add transferMode.dnla.org header to HTTP requests as this is
	  required by the DLNA specs and doesn't hurt in other situations.
	  Fixes bug #561802.

2008-11-20 23:59:07 +0000  Michael Smith <msmith@xiph.org>

	  sys/osxvideo/osxvideosink.*: Handle video window resizing more correctly, avoiding crashes when embedding the window ...
	  Original commit message from CVS:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  Handle video window resizing more correctly, avoiding crashes when
	  embedding the window and resizing it.

2008-11-20 22:56:58 +0000  Michael Smith <msmith@xiph.org>

	  gst/udp/: Fix multiudpsink on OSX by passing the specific length of the socket, refactor that into a function shared ...
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c:
	  * gst/udp/gstudpnetutils.c:
	  * gst/udp/gstudpnetutils.h:
	  * gst/udp/gstudpsrc.c:
	  Fix multiudpsink on OSX by passing the specific length of the socket,
	  refactor that into a function shared with the same thing in udpsrc.

2008-11-20 20:07:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Fix the scaling code.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	  (uint64_ceiling_scale), (gst_wavparse_calculate_duration),
	  (gst_wavparse_stream_headers):
	  Fix the scaling code.
	  Fix parsing of the INFO chunks, we were reading the wrong number of
	  bytes.  Fixes #561580.

2008-11-20 14:30:40 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/matroska/matroska-mux.c: Fix NULL pointer dereference of an unset codec_id in the recently added Dirac paths
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  Fix NULL pointer dereference of an unset codec_id in the recently
	  added Dirac paths

2008-11-20 13:58:43 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Just keep disabling elements that hang the states test until it works.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Just keep disabling elements that hang the states test until it
	  works.

2008-11-20 13:46:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/libpng/gstpngenc.c: Don't flush downstream after every buffer - that's not what this libpng callback is for at all!
	  Original commit message from CVS:
	  * ext/libpng/gstpngenc.c:
	  Don't flush downstream after every buffer - that's not what
	  this libpng callback is for at all!

2008-11-17 14:04:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/v4l2src_calls.c: Turns out we don't always get the frame sizes in a predefined order from lowest to highest ...
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c:
	  (gst_v4l2src_probe_caps_for_format_and_size), (sort_by_frame_size),
	  (gst_v4l2src_probe_caps_for_format):
	  Turns out we don't always get the frame sizes in a predefined
	  order from lowest to highest resolution, so let's just sort the
	  list by frame size once we've queried the possible resolutions
	  rather than assume any particular order. Fixes probed caps for
	  the camera in my HP2133 mini notebook and makes v4l2src default
	  to a decent size.

2008-11-16 14:41:32 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/matroska/: Make mkvdemux aware of E-AC3.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps):
	  * gst/matroska/matroska-ids.h:
	  Make mkvdemux aware of E-AC3.

2008-11-14 18:41:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Add a jpeg depayloader.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpjpegdepay.c: (gst_rtp_jpeg_depay_base_init),
	  (gst_rtp_jpeg_depay_class_init), (gst_rtp_jpeg_depay_init),
	  (gst_rtp_jpeg_depay_finalize), (MakeTables), (MakeQuantHeader),
	  (MakeHuffmanHeader), (MakeDRIHeader), (MakeHeaders),
	  (gst_rtp_jpeg_depay_setcaps), (gst_rtp_jpeg_depay_process),
	  (gst_rtp_jpeg_depay_change_state),
	  (gst_rtp_jpeg_depay_plugin_init):
	  * gst/rtp/gstrtpjpegdepay.h:
	  Add a jpeg depayloader.
	  * gst/rtp/gstrtpjpegpay.c:
	  Set the default properties on the payloader to better defaults.

2008-11-14 15:42:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/gstv4l2.c: Give it a primary rank for autovideosrc.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2.c:
	  Give it a primary rank for autovideosrc.

2008-11-14 11:41:55 +0000  Bjorn Ostby <bjornos@axis.com>

	  gst/rtp/: Add JPEG payloader. Fixes #560756.
	  Original commit message from CVS:
	  Patch by: Bjorn Ostby <bjornos at axis dot com>
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpjpegpay.c: (gst_rtp_jpeg_pay_base_init),
	  (gst_rtp_jpeg_pay_class_init), (gst_rtp_jpeg_pay_init),
	  (gst_rtp_jpeg_pay_setcaps), (gst_rtp_jpeg_pay_header_size),
	  (gst_rtp_jpeg_pay_read_quant_table),
	  (gst_rtp_jpeg_pay_scan_marker), (gst_rtp_jpeg_pay_handle_buffer),
	  (gst_rtp_jpeg_pay_set_property), (gst_rtp_jpeg_pay_get_property),
	  (gst_rtp_jpeg_pay_plugin_init):
	  * gst/rtp/gstrtpjpegpay.h:
	  Add JPEG payloader. Fixes #560756.

2008-11-13 17:45:59 +0000  Fabricio Godoy <skarllot@gmail.com>

	  sys/: Fix some spelling mistakes. Fixes #556802.
	  Original commit message from CVS:
	  Patch by: Fabricio Godoy <skarllot at gmail dot com>
	  * sys/oss/gstosssink.c: (gst_oss_sink_open):
	  * sys/oss/gstosssrc.c: (gst_oss_src_open):
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_mmap):
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  Fix some spelling mistakes. Fixes #556802.

2008-11-13 16:24:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/: Add presets for equalizer. Fixes #522183.
	  Original commit message from CVS:
	  * gst/equalizer/GstIirEqualizer10Bands.prs:
	  * gst/equalizer/GstIirEqualizer3Bands.prs:
	  * gst/equalizer/Makefile.am:
	  * gst/equalizer/gstiirequalizer10bands.c:
	  * gst/equalizer/gstiirequalizer3bands.c:
	  Add presets for equalizer. Fixes #522183.

2008-11-13 16:17:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Remove google extension again, it's not needed anymore because we never send multiple transports anymore.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/gstrtsp.c: (plugin_init):
	  * gst/rtsp/gstrtspgoogle.c:
	  * gst/rtsp/gstrtspgoogle.h:
	  Remove google extension again, it's not needed anymore because we never
	  send multiple transports anymore.

2008-11-13 16:11:16 +0000  Eric Zhang <chao.zhang@access-company.com>

	  gst/rtsp/gstrtspsrc.*: Add property to configure NAT traversal method.
	  Original commit message from CVS:
	  Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type),
	  (gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
	  (gst_rtspsrc_get_property), (gst_rtspsrc_create_stream),
	  (gst_rtspsrc_stream_free),
	  (gst_rtspsrc_stream_configure_udp_sinks),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_send_dummy_packets),
	  (gst_rtspsrc_create_transports_string),
	  (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Add property to configure NAT traversal method.
	  Ignore EOS from the internal sinks.
	  Implement sending dummy packets as a (simple) method to open up
	  some firewalls.
	  Send PLAY request to the server after we started the udp sources.
	  Fixes #559545.

2008-11-13 14:04:40 +0000  Yotam <sh.yotam@gmail.com>

	  gst/rtp/gstrtpmp4vpay.c: Flush the remaining frames on EOS. Fixes #560641.
	  Original commit message from CVS:
	  Patch by: Yotam <sh dot yotam at gmail dot com>
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event):
	  Flush the remaining frames on EOS. Fixes #560641.

2008-11-12 16:37:06 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/rtp/gstrtpg729pay.c: Fix compiler warning about printf formatting.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpg729pay.c: (gst_rtp_g729_pay_handle_buffer):
	  Fix compiler warning about printf formatting.

2008-11-12 11:55:14 +0000  Andy Wingo <wingo@pobox.com>

	  gst/qtdemux/qtdemux.*: Queue up new segment events instead of sending them from the seeking thread.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.h (struct _GstQTDemux):
	  * gst/qtdemux/qtdemux.c (gst_qtdemux_do_seek): Queue up new
	  segment events instead of sending them from the seeking thread.
	  Fixes #559288.
	  (gst_qtdemux_push_pending_newsegment): New helper, sends out
	  queued newsegment events.
	  (gst_qtdemux_loop_state_movie): Voilà, call it here. Only need to
	  call it here, as we only seek when looping, and only push in the
	  movie state.

2008-11-11 19:52:05 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/qtdemux/: Add cover and alternative copyright tag, and enhance some existing ones by marking them as container at...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_tag_add_tmpo),
	  (qtdemux_tag_add_covr), (qtdemux_parse_udta):
	  * gst/qtdemux/qtdemux_fourcc.h:
	  * gst/qtdemux/qtdemux_types.c:
	  Add cover and alternative copyright tag, and enhance some existing
	  ones by marking them as container atoms.

2008-11-11 17:33:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpg729pay.c: Don't ignore the return value of setcaps.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpg729pay.c: (gst_rtp_g729_pay_set_caps):
	  Don't ignore the return value of setcaps.

2008-11-11 17:29:03 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/gstrtpg729pay.*: Replace G729 payloader with an improved version. Fixes #532409.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtpg729pay.c: (gst_rtp_g729_pay_base_init),
	  (gst_rtp_g729_pay_class_init), (gst_rtp_g729_pay_init),
	  (gst_rtp_g729_pay_set_caps), (gst_rtp_g729_pay_handle_buffer):
	  * gst/rtp/gstrtpg729pay.h:
	  Replace G729 payloader with an improved version. Fixes #532409.

2008-11-11 16:00:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Only send one transport at a time for improved compatibility with some broken servers. See #53...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_transports_string),
	  (gst_rtspsrc_change_state):
	  Only send one transport at a time for improved compatibility with some
	  broken servers. See #537832.

2008-11-11 15:16:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Only pause/play in the seek handler when the source was playing.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
	  (gst_rtspsrc_perform_seek):
	  Only pause/play in the seek handler when the source was playing.
	  Fixes #529379.

2008-11-11 12:18:23 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.c: Fix muxing of Dirac streams if the input already has the format we need, i.e. is the out...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_handle_dirac_packet):
	  Fix muxing of Dirac streams if the input already has the format
	  we need, i.e. is the output of matroskademux.

2008-11-11 10:06:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: Don't segfault on string typed tags being NULL. Fixes #560155.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Don't segfault on string typed tags being NULL. Fixes #560155.

2008-11-10 16:44:45 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/matroska/matroska-mux.c: Fix mapping AAC profile to Matroska codec id.
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (aac_codec_data_to_codec_id),
	  (gst_matroska_mux_audio_pad_setcaps):
	  Fix mapping AAC profile to Matroska codec id.

2008-11-10 16:36:09 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/qtdemux/qtdemux.c: Refactor some raw audio caps building, and handle >16-bit cases.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
	  (qtdemux_video_caps), (qtdemux_audio_caps):
	  Refactor some raw audio caps building, and handle >16-bit cases.
	  Fix/replace building caps from a string description.

2008-11-10 13:59:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/: Make author name consistent with others.
	  Original commit message from CVS:
	  * gst/audiofx/audiowsincband.c:
	  * gst/audiofx/audiowsinclimit.c:
	  * gst/cutter/gstcutter.c:
	  Make author name consistent with others.

2008-11-10 12:13:21 +0000  Eric Zhang <chao.zhang@access-company.com>

	  gst/rtsp/gstrtspsrc.c: Pause the RTSP stream before doing a new play request.
	  Original commit message from CVS:
	  Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_perform_seek),
	  (gst_rtspsrc_stream_configure_udp_sink):
	  Pause the RTSP stream before doing a new play request.
	  Make sure that adding the udpsinks does not cause the rtspsrc to become
	  a sink. Fixes #559547.

2008-11-05 14:42:35 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Implement Dirac muxing into Matroska comforming to the spec, i.e. put all Dirac packages up to a pictu...
	  Original commit message from CVS:
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska-mux.c: (gst_matroska_pad_free),
	  (gst_matroska_mux_handle_dirac_packet),
	  (gst_matroska_mux_write_data):
	  Implement Dirac muxing into Matroska comforming to the spec, i.e.
	  put all Dirac packages up to a picture into a Matroska block.
	  TODO: Implement writing of the ReferenceBlock Matroska elements,
	  currently the Dirac muxing is only 100% correct if Matroska version 2
	  is selected for muxing.

2008-11-04 12:32:48 +0000  Bastien Nocera <hadess@hadess.net>

	  Optionally use libv4l to access v4l2 devices. Fixes bug #545033.
	  Original commit message from CVS:
	  Patch by: Bastien Nocera <hadess at hadess dot net>,
	  Hans de Goede <jwrdegoede at fedoraproject dot org>
	  * configure.ac:
	  * sys/v4l2/Makefile.am:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read):
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	  (gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_close),
	  (gst_v4l2_get_norm), (gst_v4l2_set_norm), (gst_v4l2_get_frequency),
	  (gst_v4l2_set_frequency), (gst_v4l2_signal_strength),
	  (gst_v4l2_get_attribute), (gst_v4l2_set_attribute),
	  (gst_v4l2_get_input), (gst_v4l2_set_input):
	  * sys/v4l2/v4l2_calls.h:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize),
	  (gst_v4l2_buffer_new), (gst_v4l2_buffer_pool_finalize),
	  (gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate),
	  (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_probe_caps_for_format_and_size),
	  (gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame),
	  (gst_v4l2src_set_capture), (gst_v4l2src_capture_init),
	  (gst_v4l2src_capture_start), (gst_v4l2src_capture_stop),
	  (gst_v4l2src_get_nearest_size):
	  Optionally use libv4l to access v4l2 devices. Fixes bug #545033.

2008-11-04 12:28:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Don't install static libs for plugins. Fixes #550851 for -good.
	  Original commit message from CVS:
	  * ext/aalib/Makefile.am:
	  * ext/annodex/Makefile.am:
	  * ext/cairo/Makefile.am:
	  * ext/dv/Makefile.am:
	  * ext/esd/Makefile.am:
	  * ext/flac/Makefile.am:
	  * ext/gconf/Makefile.am:
	  * ext/gdk_pixbuf/Makefile.am:
	  * ext/hal/Makefile.am:
	  * ext/jpeg/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/libcaca/Makefile.am:
	  * ext/libmng/Makefile.am:
	  * ext/libpng/Makefile.am:
	  * ext/mikmod/Makefile.am:
	  * ext/pulse/Makefile.am:
	  * ext/raw1394/Makefile.am:
	  * ext/shout2/Makefile.am:
	  * ext/soup/Makefile.am:
	  * ext/speex/Makefile.am:
	  * ext/taglib/Makefile.am:
	  * ext/wavpack/Makefile.am:
	  * gst/alpha/Makefile.am:
	  * gst/apetag/Makefile.am:
	  * gst/audiofx/Makefile.am:
	  * gst/auparse/Makefile.am:
	  * gst/autodetect/Makefile.am:
	  * gst/avi/Makefile.am:
	  * gst/cutter/Makefile.am:
	  * gst/debug/Makefile.am:
	  * gst/effectv/Makefile.am:
	  * gst/equalizer/Makefile.am:
	  * gst/flx/Makefile.am:
	  * gst/goom/Makefile.am:
	  * gst/goom2k1/Makefile.am:
	  * gst/icydemux/Makefile.am:
	  * gst/id3demux/Makefile.am:
	  * gst/interleave/Makefile.am:
	  * gst/law/Makefile.am:
	  * gst/level/Makefile.am:
	  * gst/matroska/Makefile.am:
	  * gst/median/Makefile.am:
	  * gst/monoscope/Makefile.am:
	  * gst/multifile/Makefile.am:
	  * gst/multipart/Makefile.am:
	  * gst/oldcore/Makefile.am:
	  * gst/qtdemux/Makefile.am:
	  * gst/replaygain/Makefile.am:
	  * gst/rtp/Makefile.am:
	  * gst/rtsp/Makefile.am:
	  * gst/smpte/Makefile.am:
	  * gst/spectrum/Makefile.am:
	  * gst/udp/Makefile.am:
	  * gst/videobox/Makefile.am:
	  * gst/videocrop/Makefile.am:
	  * gst/videofilter/Makefile.am:
	  * gst/videomixer/Makefile.am:
	  * gst/wavenc/Makefile.am:
	  * gst/wavparse/Makefile.am:
	  * sys/directdraw/Makefile.am:
	  * sys/directsound/Makefile.am:
	  * sys/oss/Makefile.am:
	  * sys/osxaudio/Makefile.am:
	  * sys/osxvideo/Makefile.am:
	  * sys/sunaudio/Makefile.am:
	  * sys/v4l2/Makefile.am:
	  * sys/waveform/Makefile.am:
	  * sys/ximage/Makefile.am:
	  Don't install static libs for plugins. Fixes #550851 for -good.

2008-10-31 18:17:50 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/Makefile.am: Include $(FLAC_CFLAGS) in CFLAGS to make sure to find the FLAC headers.
	  Original commit message from CVS:
	  * ext/flac/Makefile.am:
	  Include $(FLAC_CFLAGS) in CFLAGS to make sure to find the FLAC headers.
	  This fixes compilation if FLAC is installed in an uncommon location
	  that is not already handled by other CFLAGS. Fixes bug #558711.

2008-10-31 10:08:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/v4l2src_calls.c: Guard more uncommon formats with ifdefs so that we can compile on older versions.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_format_get_rank):
	  Guard more uncommon formats with ifdefs so that we can compile on older
	  versions.

2008-10-31 10:00:18 +0000  Nick Haddad <nick@haddads.net>

	  gst/avi/gstavidemux.c: Invert other uncompressed RGB formats. Fixes #558554.
	  Original commit message from CVS:
	  Patch by: Nick Haddad <nick at haddads dot net>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_is_uncompressed),
	  (gst_avi_demux_invert), (gst_avi_demux_process_next_entry),
	  (gst_avi_demux_stream_data):
	  Invert other uncompressed RGB formats. Fixes #558554.

2008-10-30 15:08:49 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavenc/gstwavenc.*: Add support for float/double as input and remove the (nowadays) useless parsing of the depth ...
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	  (gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
	  * gst/wavenc/gstwavenc.h:
	  Add support for float/double as input and remove the (nowadays)
	  useless parsing of the depth as we require width==depth.

2008-10-30 10:31:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Narrow down the caps of the mpeg audio pay/depayloaders to only accept mpeg version 1. Fixes #558427.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps):
	  * gst/rtp/gstrtpmpapay.c:
	  Narrow down the caps of the mpeg audio pay/depayloaders to only accept
	  mpeg version 1. Fixes #558427.

2008-10-29 18:28:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpL16pay.c: Only put an integral amount of samples in the RTP packet.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_flush),
	  (gst_rtp_L16_pay_getcaps):
	  Only put an integral amount of samples in the RTP packet.
	  Fixes #556641.

2008-10-28 17:42:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpchannels.*: Add method to get possible channel positions.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpchannels.c: (gst_rtp_channels_get_by_index):
	  * gst/rtp/gstrtpchannels.h:
	  Add method to get possible channel positions.

2008-10-28 17:39:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/Makefile.am: Also commit updated makefile
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  Also commit updated makefile

2008-10-28 14:56:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavenc/gstwavenc.c: Don't allow width=32,depth=24 as input. WAV requires that the width is the next integer multi...
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
	  Don't allow width=32,depth=24 as input. WAV requires that the width
	  is the next integer multiply of 8 from the depth.

2008-10-28 10:01:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Add mappings for multichannel support. Does not completely just work because the getcaps function does not ...
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps):
	  * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
	  (gst_rtp_L16_pay_getcaps):
	  * gst/rtp/gstrtpchannels.c: (check_channels),
	  (gst_rtp_channels_get_by_pos), (gst_rtp_channels_get_by_order),
	  (gst_rtp_channels_create_default):
	  * gst/rtp/gstrtpchannels.h:
	  Add mappings for multichannel support. Does not completely just work
	  because the getcaps function does not yet return the allowed channel
	  mappings. See #556641.

2008-10-28 06:50:57 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/goom/: Add license headers in all source files. Remove filter.c from
	  Original commit message from CVS:
	  * gst/goom/Makefile.am:
	  * gst/goom/README:
	  * gst/goom/config_param.c:
	  * gst/goom/convolve_fx.c:
	  * gst/goom/drawmethods.c:
	  * gst/goom/drawmethods.h:
	  * gst/goom/filters.c:
	  * gst/goom/filters_mmx.s:
	  * gst/goom/flying_stars_fx.c:
	  * gst/goom/goom.h:
	  * gst/goom/goom_config.h:
	  * gst/goom/goom_config_param.h:
	  * gst/goom/goom_core.c:
	  * gst/goom/goom_filters.h:
	  * gst/goom/goom_fx.h:
	  * gst/goom/goom_graphic.h:
	  * gst/goom/goom_plugin_info.h:
	  * gst/goom/goom_tools.c:
	  * gst/goom/goom_tools.h:
	  * gst/goom/goom_typedefs.h:
	  * gst/goom/goom_visual_fx.h:
	  * gst/goom/graphic.c:
	  * gst/goom/ifs.c:
	  * gst/goom/ifs.h:
	  * gst/goom/lines.c:
	  * gst/goom/lines.h:
	  * gst/goom/mathtools.c:
	  * gst/goom/mathtools.h:
	  * gst/goom/mmx.c:
	  * gst/goom/motif_goom1.h:
	  * gst/goom/motif_goom2.h:
	  * gst/goom/plugin_info.c:
	  * gst/goom/ppc_drawings.h:
	  * gst/goom/ppc_zoom_ultimate.h:
	  * gst/goom/sound_tester.c:
	  * gst/goom/sound_tester.h:
	  * gst/goom/surf3d.c:
	  * gst/goom/surf3d.h:
	  * gst/goom/tentacle3d.c:
	  * gst/goom/tentacle3d.h:
	  * gst/goom/v3d.c:
	  * gst/goom/v3d.h:
	  * gst/goom/xmmx.c:
	  Add license headers in all source files. Remove filter.c from
	  EXTRA_DIST, as its in SOURCES already. Mention the files in the REDME
	  which are not used right now. Fixes #557709.

2008-10-27 11:28:30 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/gstrtpL16pay.c: Implement getcaps in rtpL16pay. Fixes #556484.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_class_init),
	  (gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_getcaps):
	  Implement getcaps in rtpL16pay. Fixes #556484.

2008-10-27 11:03:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpL16depay.c: Check if clock-rate and channels are valid.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps),
	  (gst_rtp_L16_depay_process):
	  Check if clock-rate and channels are valid.
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  Use the marker bit to set the DISCONT flag on outgoing buffers.
	  * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps),
	  (gst_rtp_ac3_depay_process):
	  Don't ignore the return value of set_caps.
	  No need to validate the buffer, the base class does that for us.
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	  (gst_rtp_amr_depay_process):
	  * gst/rtp/gstrtpamrdepay.h:
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  No need to set output caps on the buffers, the base class does that for
	  us.
	  The subclass will make sure we are negotiated.
	  * gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps),
	  (gst_rtp_dv_depay_process), (gst_rtp_dv_depay_reset):
	  * gst/rtp/gstrtpdvdepay.h:
	  Clean up caps negotiation.
	  The subclass will make sure we are negotiated.
	  * gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_setcaps),
	  (gst_rtp_g726_depay_process):
	  Clean up caps negotiation.
	  Use the marker bit to set the DISCONT flag on outgoing buffers.
	  * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_init),
	  (gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process):
	  * gst/rtp/gstrtpg729depay.h:
	  The subclass will make sure we are negotiated.
	  Use the marker bit to set the DISCONT flag on outgoing buffers.
	  * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_setcaps),
	  (gst_rtp_gsm_depay_process):
	  Clean up caps negotiation.
	  Use the marker bit to set the DISCONT flag on outgoing buffers.
	  * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps):
	  Clean up caps negotiation.
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps),
	  (gst_rtp_h263_depay_process):
	  Clean up caps negotiation.
	  No need to validate the buffer, the base class does that for us.
	  * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_setcaps),
	  (gst_rtp_h263_pay_flush), (gst_rtp_h263_pay_handle_buffer):
	  * gst/rtp/gstrtph263pay.h:
	  Don't ignore the return value of set_outcaps.
	  Do some more timestamps.
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	  (gst_rtp_h263p_depay_process):
	  Clean up caps negotiation.
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init),
	  (gst_rtp_h263p_pay_setcaps), (gst_rtp_h263p_pay_flush),
	  (gst_rtp_h263p_pay_handle_buffer):
	  * gst/rtp/gstrtph263ppay.h:
	  Don't ignore the return value of set_outcaps.
	  Do some more timestamps.
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps),
	  (gst_rtp_h264_depay_process):
	  Clean up caps negotiation.
	  Don't ignore the return value of setcaps.
	  Fix possible caps leak.
	  No need to validate the buffer, the base class does that for us.
	  * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps):
	  Add some more debug info.
	  * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps),
	  (gst_rtp_ilbc_depay_process):
	  Clean up caps negotiation.
	  Use the marker bit to set the DISCONT flag on outgoing buffers.
	  * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_sink_setcaps):
	  Clean up caps negotiation.
	  * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps),
	  (gst_rtp_mp1s_depay_process):
	  Clean up caps negotiation.
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  No need to set caps on buffers, subclass does that for us.
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
	  (gst_rtp_mp2t_depay_process):
	  Clean up caps negotiation.
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  No need to set caps on buffers, subclass does that for us.
	  * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
	  (gst_rtp_mp4a_depay_process):
	  Clean up caps negotiation.
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_new_caps),
	  (gst_rtp_mp4a_pay_setcaps):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps),
	  (gst_rtp_mp4g_depay_process):
	  Clean up caps negotiation.
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  No need to set caps on buffers, subclass does that for us.
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize),
	  (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	  (gst_rtp_mp4v_depay_process):
	  Clean up caps negotiation.
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  No need to set caps on buffers, subclass does that for us.
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_new_caps),
	  (gst_rtp_mp4v_pay_setcaps):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps),
	  (gst_rtp_mpa_depay_process):
	  Clean up caps negotiation.
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  Use the marker bit to set the DISCONT flag on outgoing buffers.
	  * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_setcaps):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_setcaps),
	  (gst_rtp_mpv_depay_process):
	  Clean up caps negotiation.
	  Actually set output caps.
	  No need to validate the buffer, the base class does that for us.
	  * gst/rtp/gstrtpmpvpay.c: (gst_rtp_mpv_pay_setcaps):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps),
	  (gst_rtp_pcma_depay_process):
	  Clean up caps negotiation.
	  Set output buffer duration because we can.
	  Use the marker bit to set the DISCONT flag on outgoing buffers.
	  * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_setcaps):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps),
	  (gst_rtp_pcmu_depay_process):
	  Clean up caps negotiation.
	  Use the marker bit to set the DISCONT flag on outgoing buffers.
	  * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_setcaps):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init),
	  (gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process):
	  Clean up caps negotiation.
	  Set output caps on the pad and header buffers.
	  Set duration on output buffers because we can.
	  * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_parse_ident):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_setcaps),
	  (gst_rtp_sv3v_depay_process):
	  Clean up caps negotiation.
	  No need to validate the buffer, the base class does that for us.
	  No need to set caps out output buffers, subclass does that.
	  * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps),
	  (gst_rtp_theora_depay_process):
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_class_init),
	  (gst_rtp_theora_pay_flush_packet), (encode_base64),
	  (gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
	  (gst_rtp_theora_pay_handle_buffer):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps),
	  (gst_rtp_vorbis_depay_process):
	  Don't ignore the return value of setcaps.
	  No need to validate the buffer, the base class does that for us.
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
	  Don't ignore the return value of set_outcaps.
	  * gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps):
	  Clean up caps negotiation, don't ignore setcaps return.
	  * gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_setcaps):
	  Don't ignore the return value of set_outcaps.

2008-10-27 10:35:07 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/matroska/matroska-demux.c: Forward unknown events upstream.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_handle_src_event):
	  Forward unknown events upstream.

2008-10-27 10:33:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/elements/icydemux.c: Add some refcount check
	  Original commit message from CVS:
	  * tests/check/elements/icydemux.c: (icydemux_found_pad):
	  Add some refcount check
	  * tests/check/elements/rtp-payloading.c: (rtp_pipeline_run):
	  Don't ignore the result of write(), fixes a  compiler warning for me.
	  * tests/icles/videobox-test.c: (main):
	  Make the output a little more pretty.

2008-10-27 09:26:19 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/esd/esdmon.c: Add doc blob.
	  Original commit message from CVS:
	  * ext/esd/esdmon.c:
	  Add doc blob.

2008-10-27 09:21:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: Add the docs of the new elements.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  Add the docs of the new elements.

2008-10-27 09:04:37 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/autodetect/: Fix "Since" tags in the documentation.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosrc.c:
	  (gst_auto_audio_src_class_init):
	  * gst/autodetect/gstautovideosrc.c:
	  (gst_auto_video_src_class_init):
	  Fix "Since" tags in the documentation.

2008-10-27 09:00:29 +0000  Sjoerd Simons <sjoerd@luon.net>

	  ext/soup/gstsouphttpsrc.c: Add support for souphttpsrc to act as a live source. This makes it possible to get timesta...
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init),
	  (gst_soup_http_src_set_property), (gst_soup_http_src_get_property):
	  Add support for souphttpsrc to act as a live source. This makes it
	  possible to get timestamped buffers in combination with the
	  "do-timestamp" property. Fixes bug #556019.

2008-10-27 08:54:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/autodetect/: Implement src plugins. Little code/string cleanup in the sinks.
	  Original commit message from CVS:
	  * gst/autodetect/Makefile.am:
	  * gst/autodetect/gstautoaudiosink.c:
	  * gst/autodetect/gstautoaudiosrc.c:
	  * gst/autodetect/gstautoaudiosrc.h:
	  * gst/autodetect/gstautodetect.c:
	  * gst/autodetect/gstautovideosink.c:
	  * gst/autodetect/gstautovideosrc.c:
	  * gst/autodetect/gstautovideosrc.h:
	  Implement src plugins. Little code/string cleanup in the sinks.
	  Fixes #523813.

2008-10-27 08:45:11 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/matroska/matroska-mux.c: Fix a memory leak when pads are requested but the pipeline never goes into PLAYING.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt <pkj at axis com>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
	  (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad):
	  Fix a memory leak when pads are requested but the pipeline never
	  goes into PLAYING.
	  Correctly remove request pads, no matter if they have collected
	  data or not.
	  Fixes bug #557710.

2008-10-27 08:40:02 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/udp/gstudpnetutils.h: Define the correct WINVER so getaddinfo() can be used when using mingw32. Fixes bug #557294.
	  Original commit message from CVS:
	  Patch by: <lrn1986 at gmail dot com>
	  * gst/udp/gstudpnetutils.h:
	  Define the correct WINVER so getaddinfo() can be used when using
	  mingw32. Fixes bug #557294.

2008-10-27 08:36:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/udp/: Fix "argument type mismatch" compiler warnings on Windows.
	  Original commit message from CVS:
	  Patch by: <lrn1986 at gmail dot com>
	  * gst/udp/gstdynudpsink.c: (gst_dynudpsink_render):
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Fix "argument type mismatch" compiler warnings on Windows.
	  Fixes bug #557293.

2008-10-27 08:30:51 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.c: Don't calculate the filter coefficients for every single buffer but only when it's n...
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c: (update_coefficients):
	  Don't calculate the filter coefficients for every single buffer
	  but only when it's needed. Fixes bug #557260.

2008-10-26 20:05:43 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to development -> 0.10.11.1
	  Original commit message from CVS:
	  * configure.ac:
	  Back to development -> 0.10.11.1

2008-10-26 20:04:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-plugins-good.doap: Fix version number of 0.10.11 release in doap file
	  Original commit message from CVS:
	  * gst-plugins-good.doap:
	  Fix version number of 0.10.11 release in doap file

=== release 0.10.11 ===

2008-10-24 22:41:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.11
	  Original commit message from CVS:
	  Release 0.10.11

2008-10-24 22:20:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2008-10-24 16:30:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Commit 0.10.10.4 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  Commit 0.10.10.4 pre-release

2008-10-21 12:42:45 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/gstavimux.c: Fix VPRP chunk setup in avimux.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Fix VPRP chunk setup in avimux.
	  Fixes: #556010
	  Patch By: Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

2008-10-21 12:38:35 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	  gst/videobox/gstvideobox.c: support dynamically changing properties in videobox
	  Original commit message from CVS:
	  * gst/videobox/gstvideobox.c:
	  support dynamically changing properties in videobox
	  Fixed: #557085
	  Patch By: Wim Taymans <wim.taymans@collabora.co.uk>

2008-10-16 17:10:42 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: 0.10.10.3 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  0.10.10.3 pre-release

2008-10-16 15:30:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Don't run the states test on pulsesrc and pulsesink
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Don't run the states test on pulsesrc and pulsesink

2008-10-16 11:52:44 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Commit 0.10.10.2 pre-release bump that actually went out on 2008-10-11
	  Original commit message from CVS:
	  * configure.ac:
	  Commit 0.10.10.2 pre-release bump that actually went
	  out on 2008-10-11

2008-10-15 15:42:29 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: Skip entries for streams that don't have a output pad yet, thereby avoiding calling pad functi...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
	  Skip entries for streams that don't have a output pad yet, thereby
	  avoiding calling pad functions with a NULL pad.
	  Fixes #556424

2008-10-15 09:39:27 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Remove previous wrong commit
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: Remove previous wrong commit
	  * tests/check/elements/icydemux.c: (icydemux_found_pad):
	  Remove problematic and useless refcount check.
	  Fixes #556381

2008-10-15 09:27:27 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Remove problematic and useless refcount check.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	  Remove problematic and useless refcount check.
	  Fixes #556381

2008-10-13 18:10:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Don't install static libs for plugins. Fixes #550851 for ugly.
	  Original commit message from CVS:
	  * ext/a52dec/Makefile.am:
	  * ext/amrnb/Makefile.am:
	  * ext/cdio/Makefile.am:
	  * ext/dvdnav/Makefile.am:
	  * ext/dvdread/Makefile.am:
	  * ext/lame/Makefile.am:
	  * ext/mad/Makefile.am:
	  * ext/mpeg2dec/Makefile.am:
	  * ext/sidplay/Makefile.am:
	  * gst/ac3parse/Makefile.am:
	  * gst/asfdemux/Makefile.am:
	  * gst/dvdlpcmdec/Makefile.am:
	  * gst/dvdsub/Makefile.am:
	  * gst/iec958/Makefile.am:
	  * gst/mpegaudioparse/Makefile.am:
	  * gst/mpegstream/Makefile.am:
	  * gst/realmedia/Makefile.am:
	  * gst/synaesthesia/Makefile.am:
	  Don't install static libs for plugins. Fixes #550851 for ugly.

2008-10-10 12:28:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/flac/: Cast some size_t arguments to guint to avoid compiler warnings on 64-bit systems.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c (gst_flac_dec_read_stream):
	  * ext/flac/gstflacenc.c (gst_flac_enc_write_callback):
	  Cast some size_t arguments to guint to avoid compiler
	  warnings on 64-bit systems.

2008-10-09 14:27:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Return TRUE instead of FALSE from the event handler when we swallowed the event.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event):
	  Return TRUE instead of FALSE from the event handler when we swallowed the
	  event.

2008-10-08 15:59:56 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  remove old CDIO plugin now in ugly
	  Original commit message from CVS:
	  remove old CDIO plugin now in ugly

2008-10-08 14:47:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Reset header state. Fixes #555321.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	  (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index):
	  Reset header state. Fixes #555321.

2008-10-08 13:31:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.*: For timestamping audio packets we need to take into account the amount of blocks in one entry ...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
	  (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index):
	  * gst/avi/gstavidemux.h:
	  For timestamping audio packets we need to take into account the
	  amount of blocks in one entry using the blockalign. Fixes some sync
	  issues with zero-padded audio blocks in the beginning of avi files.

2008-10-08 10:42:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/multifile/gstmultifilesrc.c: Implement DEFAULT and BUFFER position queries. See #555260.
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init),
	  (gst_multi_file_src_query):
	  Implement DEFAULT and BUFFER position queries. See #555260.

2008-10-08 09:29:00 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/ximage/gstximagesrc.c: Fix build for systems that don't have XDamage.
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_stop):
	  Fix build for systems that don't have XDamage.

2008-10-07 09:58:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/rtp/: Add some more H263p server and client examples.
	  Original commit message from CVS:
	  * tests/examples/rtp/client-H263p.sdp:
	  * tests/examples/rtp/client-H263p.sh:
	  * tests/examples/rtp/server-VTS-H263p.sh:
	  Add some more H263p server and client examples.

2008-10-03 17:03:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Depend on released versions of core and base.
	  Original commit message from CVS:
	  * configure.ac::
	  Depend on released versions of core and base.

2008-10-03 16:13:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/pulse/: Return -1 instead of 0 in error cases. Fixes #554771.
	  Original commit message from CVS:
	  * ext/pulse/pulsesink.c: (gst_pulsesink_write):
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_read):
	  Return -1 instead of 0 in error cases. Fixes #554771.

2008-10-03 15:54:07 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/ximage/gstximagesrc.c: Stop leaking the cursor image.
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_start),
	  (gst_ximage_src_stop), (gst_ximage_src_ximage_get):
	  Stop leaking the cursor image.
	  Unref the last_ximage and the cached cursor image on shutdown.
	  Fixes #551570.

2008-10-03 11:32:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/gstv4l2object.h: Getting the Class from an instance is not just a matter of casting it to the class struct b...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2object.h:
	  Getting the Class from an instance is not just a matter of casting it to
	  the class struct but it involves calling G_OBJECT_GET_CLASS on the
	  instance. Fixes #549784.

2008-10-01 21:22:26 +0000  Michael Smith <msmith@xiph.org>

	  configure.ac: Fix libs for linking directsound.
	  Original commit message from CVS:
	  * configure.ac:
	  Fix libs for linking directsound.
	  * sys/directsound/gstdirectsoundsink.c:
	  Fix buffer sizing to prevent racing the ringbuffer at startup.
	  Add volume property.

2008-09-27 00:43:07 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/pulse/pulsesink.c: Fix problems with pulsesink randomly erroring with code 'OK' after a format change on the stre...
	  Original commit message from CVS:
	  * ext/pulse/pulsesink.c:
	  Fix problems with pulsesink randomly erroring with code 'OK' after a
	  format change on the stream by waiting when disconnecting the stream.

2008-09-26 14:44:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpamrdepay.c: Mark DISCONT on output buffers when the marker bit signals a new talk spurt.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init),
	  (gst_rtp_amr_depay_process):
	  Mark DISCONT on output buffers when the marker bit signals a new talk
	  spurt.
	  * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
	  Set the marker bit for buffers with a DISCONT flag to signal a talk
	  spurt.

2008-09-26 13:55:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added MP4A-LATM payloader to match the depayloader.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type),
	  (gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init),
	  (gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize),
	  (gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps),
	  (gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer),
	  (gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init):
	  * gst/rtp/gstrtpmp4apay.h:
	  Added MP4A-LATM payloader to match the depayloader.

2008-09-25 15:11:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videomixer/videomixer.c: Handle segments a little better. Fixes #537361.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_fill_queues),
	  (gst_videomixer_sink_event):
	  Handle segments a little better. Fixes #537361.

2008-09-25 12:07:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Don't assume the server supports PAUSE by default. Fixes #551048.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
	  Don't assume the server supports PAUSE by default. Fixes #551048.

2008-09-25 11:30:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Switch on the socket family to get the addrlen size right.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	  (gst_udpsrc_set_uri), (gst_udpsrc_start):
	  Switch on the socket family to get the addrlen size right.

2008-09-25 10:34:39 +0000  Daniel Franke <df@dfranke.us>

	  gst/udp/gstudpsrc.c: OS X's bind() implementation is picky about its addrlen parameter and fails with EINVAL if it is...
	  Original commit message from CVS:
	  Patch by: Daniel Franke <df at dfranke dot us>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create), (gst_udpsrc_start):
	  OS X's bind() implementation is picky about its addrlen parameter and
	  fails with EINVAL if it is larger than expected for the socket's address
	  family. Set the length to the expected length instead. Fixes #553191.

2008-09-23 18:08:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Handle the case where we cannot do desribe or when the describe result does not contain a vali...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	  Handle the case where we cannot do desribe or when the describe result
	  does not contain a valid SDP message.

2008-09-23 17:31:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.c: Fix setting the qos.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_set_property):
	  Fix setting the qos.

2008-09-17 14:50:42 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Some 'broken' files out there have atom lengths of zero... which basically results in qtdemux ...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header),
	  (gst_qtdemux_chain):
	  Some 'broken' files out there have atom lengths of zero...
	  which basically results in qtdemux consuming that atom again and again
	  until the *end of night* !
	  Detect that and emits an adequate element error message.

2008-09-17 13:49:04 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/: Fix build flags order.
	  Original commit message from CVS:
	  * gst/interleave/Makefile.am:
	  * gst/matroska/Makefile.am:
	  Fix build flags order.
	  * tests/check/elements/audioamplify.c: (GST_START_TEST):
	  * tests/check/elements/audiodynamic.c: (GST_START_TEST):
	  * tests/check/elements/audioinvert.c: (GST_START_TEST):
	  * tests/check/elements/audiopanorama.c: (GST_START_TEST):
	  Format fixes.
	  * tests/check/elements/multifile.c:
	  Pull in unistd.h

2008-09-15 21:10:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4gdepay.*: Handle interleaved streams by reordering AU in a queue.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_init),
	  (gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_setcaps),
	  (gst_rtp_mp4g_depay_clear_queue), (gst_rtp_mp4g_depay_flush_queue),
	  (gst_rtp_mp4g_depay_queue), (gst_rtp_mp4g_depay_process),
	  (gst_rtp_mp4g_depay_change_state):
	  * gst/rtp/gstrtpmp4gdepay.h:
	  Handle interleaved streams by reordering AU in a queue.

2008-09-15 16:04:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4gdepay.c: Change some of the ranges in the caps, mostly for the amount of bits we can use.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_bs_parse_init),
	  (gst_bs_parse_read), (gst_rtp_mp4g_depay_process):
	  Change some of the ranges in the caps, mostly for the amount of bits we
	  can use.
	  Added a little bitstream parse and use it to parse the AU header fields.
	  Check for malformed and wrongly sized packets better.
	  Implement more header field parsing.
	  Handle the size of fragmented packets correctly.

2008-09-14 11:32:15 +0000  Jonathan Matthew <notverysmart@gmail.com>

	  gst/qtdemux/qtdemux.c: Add mapping for 'tiff' => image/tiff
	  Original commit message from CVS:
	  Patch by: Jonathan Matthew <notverysmart@gmail.com>
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add mapping for 'tiff' => image/tiff
	  Fixes #552213

2008-09-11 11:26:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/raw1394/: Pretend to care about the result of write() which works around compiler warnings.
	  Original commit message from CVS:
	  * ext/raw1394/gstdv1394src.c: (SEND_COMMAND):
	  * ext/raw1394/gsthdv1394src.c: (SEND_COMMAND):
	  Pretend to care about the result of write() which works around
	  compiler warnings.

2008-09-04 09:25:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacenc.c: Make sure the desired default values are actually set, not only registered as defaults (actual...
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_class_init):
	  Make sure the desired default values are actually set, not only
	  registered as defaults (actual problem is that the stereo-specific
	  values are only updated if channels==2, which is not the case yet
	  when the object is created, so the default values for the
	  mid-side-stereo and loose-mid-side-stereo settings are never
	  set in _update_quality()). Makes flacenc create smaller files by
	  default (for stereo input), and fixes #550791.

2008-09-03 12:39:35 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/qtdemux/: Add support for video/mj2 mime-type and its additional atoms/boxes.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
	  (gst_qtdemux_loop_state_header), (qtdemux_parse_node),
	  (qtdemux_parse_trak), (qtdemux_video_caps):
	  * gst/qtdemux/qtdemux.h:
	  * gst/qtdemux/qtdemux_fourcc.h:
	  * gst/qtdemux/qtdemux_types.c:
	  Add support for video/mj2 mime-type and its additional atoms/boxes.
	  Fixes #550646.

2008-09-03 11:10:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/debug/gsttaginject.c: Add warning when tags parameter is unparsable and give example for quoting in the docs.
	  Original commit message from CVS:
	  * gst/debug/gsttaginject.c:
	  Add warning when tags parameter is unparsable and give example for
	  quoting in the docs.

2008-09-02 15:27:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Add mapping for IMA Loki SDL MJPEG ADPCM codec.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
	  Add mapping for IMA Loki SDL MJPEG ADPCM codec.
	  Add some alternative byteswapped mappings that seem to pop up sometimes.
	  Fixes #550288.

2008-09-02 09:40:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/: Add 'ca' to LINGUAS; add some more files with translations and some files which should be ignored by translation...
	  Original commit message from CVS:
	  * po/LINGUAS:
	  * po/POTFILES.in:
	  * po/POTFILES.skip:
	  Add 'ca' to LINGUAS; add some more files with translations and some
	  files which should be ignored by translation tools.

2008-09-02 08:51:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/speex/: Use integer encoding and decoding functions instead of converting the integer input to float in the eleme...
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
	  * ext/speex/gstspeexdec.h:
	  * ext/speex/gstspeexenc.c: (gst_speex_enc_encode):
	  * ext/speex/gstspeexenc.h:
	  Use integer encoding and decoding functions instead of converting
	  the integer input to float in the element. The libspeex integer
	  functions are doing this for us already or, if libspeex was compiled
	  in integer mode, they're doing everything using integer arithmetics.
	  Also saves some copying around.

2008-09-01 13:29:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Fix --disable-external
	  Original commit message from CVS:
	  * configure.ac:
	  Fix --disable-external

2008-08-31 17:09:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.*: Handle non-zero start timestamps and stream discontinuities correctly. This only has an ...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
	  (gst_wavpack_enc_push_block), (gst_wavpack_enc_chain):
	  * ext/wavpack/gstwavpackenc.h:
	  Handle non-zero start timestamps and stream discontinuities
	  correctly. This only has an effect if we're muxing into
	  a container format as the raw WavPack stream must contain
	  continous sample numbers.

2008-08-31 15:02:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/speex/gstspeexenc.c: Correct the timestamp and granulepos calculation by one Speex frame.
	  Original commit message from CVS:
	  * ext/speex/gstspeexenc.c: (gst_speex_enc_encode):
	  Correct the timestamp and granulepos calculation by one Speex
	  frame.

2008-08-31 14:39:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/speex/gstspeexdec.c: Correctly take the granulepos from upstream if possible and correctly handle the granulepos ...
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
	  Correctly take the granulepos from upstream if possible and
	  correctly handle the granulepos in various calculations: the
	  granulepos is the sample number of the _last_ sample in a frame, not
	  the first.
	  * ext/speex/gstspeexenc.c: (gst_speex_enc_sinkevent),
	  (gst_speex_enc_encode), (gst_speex_enc_chain),
	  (gst_speex_enc_change_state):
	  * ext/speex/gstspeexenc.h:
	  Handle non-zero start timestamps in the encoder and detect/handle
	  stream discontinuities. Fixes bug #547075.

2008-08-31 08:32:45 +0000  Craig Keogh <cskeogh@adam.com.au>

	  ext/annodex/gstcmmlparser.c: Fix compiler warnings caused by passing a string as format string instead of "%s" and th...
	  Original commit message from CVS:
	  Patch by: Craig Keogh <cskeogh at adam dot com dot au>
	  * ext/annodex/gstcmmlparser.c: (gst_cmml_parser_parse_chunk):
	  Fix compiler warnings caused by passing a string as format string
	  instead of "%s" and then the string. This is only exposed by -Wformat=2
	  as used by default on Ubuntu. Fixes bug #550015.

2008-08-30 14:15:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Make stuff compile with GST_DISABLE_GST_DEBUG.
	  Original commit message from CVS:
	  * ext/raw1394/gsthdv1394src.c: (gst_hdv1394src_create):
	  * gst/alpha/gstalpha.c: (gst_alpha_get_unit_size):
	  * gst/audiofx/audiocheblimit.c: (generate_coefficients):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_src_convert):
	  * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
	  (gst_ebml_read_element_length):
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_check_subtitle_buffer):
	  Make stuff compile with GST_DISABLE_GST_DEBUG.

2008-08-29 00:28:55 +0000  Michael Smith <msmith@xiph.org>

	  gst/law/: Ref caps before passing to gst_pad_template_new(), since that takes ownership.
	  Original commit message from CVS:
	  * gst/law/alaw.c:
	  * gst/law/mulaw.c:
	  Ref caps before passing to gst_pad_template_new(), since that takes
	  ownership.

2008-08-28 10:09:16 +0000  Mersad Jelacic <mersad@axis.com>

	  gst/multipart/: Convert audio/x-adpcm to and from the audio/G726-X in the muxer and demuxer. Fixes #549551.
	  Original commit message from CVS:
	  Patch by: Mersad Jelacic <mersad at axis dot com>
	  * gst/multipart/multipartdemux.c:
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_get_mime):
	  Convert audio/x-adpcm to and from the audio/G726-X in the muxer and
	  demuxer. Fixes #549551.

2008-08-27 16:12:39 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/osxaudio/: Fix the build on macosx.
	  Original commit message from CVS:
	  * sys/osxaudio/gstosxaudiosink.c:
	  (gst_osx_audio_sink_select_device):
	  * sys/osxaudio/gstosxaudiosrc.c:
	  (gst_osx_audio_src_create_ringbuffer),
	  (gst_osx_audio_src_select_device):
	  * sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_acquire):
	  Fix the build on macosx.

2008-08-27 15:42:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/icydemux/gsticydemux.c: Small docs fix: in the example pipeline, we need to pass iradio-mode=true to the source, ...
	  Original commit message from CVS:
	  * gst/icydemux/gsticydemux.c:
	  Small docs fix: in the example pipeline, we need to pass
	  iradio-mode=true to the source, so the server actually sends
	  an ICY stream.

2008-08-27 00:08:20 +0000  Michael Smith <msmith@xiph.org>

	  sys/osxaudio/gstosxaudio.c: Oops. Revert more completely.
	  Original commit message from CVS:
	  * sys/osxaudio/gstosxaudio.c:
	  Oops. Revert more completely.

2008-08-26 23:57:05 +0000  Michael Smith <msmith@xiph.org>

	  sys/osxaudio/gstosxaudio.c: Revert accidental element rename from testing.
	  Original commit message from CVS:
	  * sys/osxaudio/gstosxaudio.c:
	  Revert accidental element rename from testing.

2008-08-26 23:53:40 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-plugins-good.doap: Pull in 0.10.10 doap entry from release branch
	  Original commit message from CVS:
	  * gst-plugins-good.doap:
	  Pull in 0.10.10 doap entry from release branch

2008-08-26 23:05:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Update version number to reflect 0.10.10 release from branch.
	  Original commit message from CVS:
	  * configure.ac:
	  Update version number to reflect 0.10.10 release from
	  branch.

2008-08-26 21:13:08 +0000  Michael Smith <msmith@xiph.org>

	  sys/osxaudio/: Rewrite caps setting and ring buffer initialisation.
	  Original commit message from CVS:
	  * sys/osxaudio/Makefile.am:
	  * sys/osxaudio/gstosxaudio.c:
	  * sys/osxaudio/gstosxaudiosink.c:
	  * sys/osxaudio/gstosxaudiosink.h:
	  * sys/osxaudio/gstosxaudiosrc.c:
	  * sys/osxaudio/gstosxaudiosrc.h:
	  * sys/osxaudio/gstosxringbuffer.c:
	  * sys/osxaudio/gstosxringbuffer.h:
	  Rewrite caps setting and ring buffer initialisation.
	  Previously we never told CoreAudio what format we were going to send it,
	  so it only worked due to luck, and not at all on some hardware.
	  Now we explicitly advertise what formats the hardware supports, and then
	  configure the selected one correctly.

2008-08-26 12:27:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/: Fix memory leaks. Small code cleanups : No need for empty _init(). No need to memset instance structures. ...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2object.c:
	  * sys/v4l2/gstv4l2src.c:
	  * sys/v4l2/gstv4l2src.h:
	  * sys/v4l2/v4l2_calls.c:
	  * sys/v4l2/v4l2src_calls.c:
	  Fix memory leaks. Small code cleanups : No need for empty _init(). No
	  need to memset instance structures. Some more FIXME's.

2008-08-26 08:11:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/icles/.cvsignore: Ignore more.
	  Original commit message from CVS:
	  * tests/icles/.cvsignore:
	  Ignore more.

2008-08-26 08:00:57 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: Ignore files.
	  Original commit message from CVS:
	  * gst/goom/.cvsignore:
	  * gst/goom2k1/.cvsignore:
	  Ignore files.

2008-08-26 07:51:42 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/cairo/gsttextoverlay.c: Fix compiler warning.
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c:
	  Fix compiler warning.

2008-08-26 05:42:15 +0000  David Schleef <ds@schleef.org>

	  ext/cairo/gsttextoverlay.c: Fix obvious memleak.
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c: Fix obvious memleak.

2008-08-25 14:15:43 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/matroska/: Add Real[Audio|Video] support to Matroska containers.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_send_event),
	  (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_video_pad_setcaps),
	  (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_finish):
	  Add Real[Audio|Video] support to Matroska containers.
	  It works fine for:
	  * decoding real audio/video streams contained in mkv
	  * 'transmuxing' real (.rm) files into .mkv files
	  It will not work though for encoding real[audio/video] streams that
	  don't contain the 'mdpr_data' extra data on the caps.
	  The reason why this will not work is because I never intended to
	  duplicate virtually all the 'mdpr' block creation into mkvmux.
	  Fixes #536067

2008-08-25 09:48:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/law/: The encoder can't really renegotiate at the time they perform a pad-alloc so make the srcpads use fixed caps.
	  Original commit message from CVS:
	  * gst/law/alaw-encode.c: (gst_alaw_enc_init), (gst_alaw_enc_chain):
	  * gst/law/mulaw-conversion.c:
	  * gst/law/mulaw-encode.c: (gst_mulawenc_init),
	  (gst_mulawenc_chain):
	  The encoder can't really renegotiate at the time they perform a
	  pad-alloc so make the srcpads use fixed caps.
	  Check the buffer size after a pad-alloc because the returned size might
	  not be right when the downstream element does not know the size of the
	  new buffer (capsfilter). Fixes #549073.

2008-08-23 15:43:49 +0000  Filippo Argiolas <filippo.argiolas@gmail.com>

	  sys/v4l2/gstv4l2tuner.c: v4l2src doesn't have a property named "norm" so don't try to notify about changes to that pr...
	  Original commit message from CVS:
	  Patch by: Filippo Argiolas <filippo dot argiolas at gmail dot com>
	  * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_set_norm_and_notify):
	  v4l2src doesn't have a property named "norm" so don't try to notify
	  about changes to that property. The "norm" property and related
	  code are commented out currently. Fixes bug #549090.

2008-08-23 15:33:49 +0000  Mike Ruprecht <cmaiku@gmail.com>

	  sys/v4l2/gstv4l2object.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged ...
	  Original commit message from CVS:
	  Patch by: Mike Ruprecht <cmaiku at gmail dot com>
	  * sys/v4l2/gstv4l2object.c: (gst_v4l2_class_probe_devices):
	  Reprobe devices again instead of taking a cached list as new
	  devices could've been plugged in. Fixes bug #549062.

2008-08-22 16:04:02 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/autodetect/Makefile.am: Don't link the autodetect plugin with GConf as it doesn't use GConf. Fixes bug #545463.
	  Original commit message from CVS:
	  * gst/autodetect/Makefile.am:
	  Don't link the autodetect plugin with GConf as it doesn't
	  use GConf. Fixes bug #545463.

2008-08-22 12:24:23 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/ebml-read.c: Change some GST_ELEMENT_ERRORs to GST_ERROR_OBJECT to make it possible to ignore errors and...
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
	  (gst_ebml_read_element_length), (gst_ebml_read_uint),
	  (gst_ebml_read_sint), (gst_ebml_read_float),
	  (gst_ebml_read_header):
	  Change some GST_ELEMENT_ERRORs to GST_ERROR_OBJECT to make it
	  possible to ignore errors and not post any ERROR messages on
	  the bus.
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_contents):
	  Ignore any errors and not just EOS when parsing the contents of
	  a SeekHead. Errors here are usually caused by truncated files
	  and playback of the file works fine. Fixes playback of the
	  audio_only_chapter_seekbroken.mka file from the MPlayer samples
	  archive.

2008-08-22 11:29:26 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  gst/multipart/: Conform to RFC2046. audio/basic is mulaw 8000Hz mono.
	  Original commit message from CVS:
	  * gst/multipart/multipartdemux.c:
	  * gst/multipart/multipartmux.c:
	  Conform to RFC2046. audio/basic is mulaw 8000Hz mono.

2008-08-21 21:56:19 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	* ChangeLog:
	* sys/directdraw/gstdirectdrawsink.c:
	  sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc, gst_directdraw_sink_bufferpool_clear):
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc,
	  gst_directdraw_sink_bufferpool_clear):
	  Fix two more buffer ref leaks.

2008-08-21 15:28:09 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  sys/directdraw/gstdirectdrawsink.c: Fix buffer ref leak.
	  Original commit message from CVS:
	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas at tandberg com>
	  * sys/directdraw/gstdirectdrawsink.c:
	  (gst_directdraw_sink_show_frame):
	  Fix buffer ref leak.

2008-08-21 13:27:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavenc/gstwavenc.c: Revert the last commit. wavenc still supports width!=depth for 32 bit width. Thanks Tim.
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
	  Revert the last commit. wavenc still supports width!=depth for 32 bit
	  width. Thanks Tim.

2008-08-21 13:22:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: If the duration of a block is unknown only use the timestamp for the first lace and us...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock):
	  If the duration of a block is unknown only use the timestamp for the
	  first lace and use GST_CLOCK_TIME_NONE as duration for the following
	  laces. Otherwise every lace has the same timestamp which leads to
	  various problems. Really fixes bug #548831.

2008-08-21 12:56:01 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavenc/gstwavenc.c: If we're not allowing width!=depth in wavenc we should also disable the code that was added t...
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
	  If we're not allowing width!=depth in wavenc we should also disable
	  the code that was added to support width!=depth.

2008-08-21 12:52:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: Don't calculate the default duration of a frame from the audio sampling rate. This onl...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
	  Don't calculate the default duration of a frame from the audio sampling
	  rate. This only works for raw audio if every frame contains a single
	  sample and results in broken buffer durations for other formats
	  if no specified default duration is given or the blocks have no
	  duration. Fixes bug #548831.

2008-08-21 12:34:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: Allow zero sized blocks instead of returning GST_FLOW_OK. Such blocks are used for tex...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock):
	  Allow zero sized blocks instead of returning GST_FLOW_OK. Such blocks
	  are used for text/plain subtitles as a gap-filler in some files.

2008-08-21 12:12:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/gstv4l2src.c: Add S910 and PWC formats with a low priority.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure),
	  (gst_v4l2_get_caps_info):
	  Add S910 and PWC formats with a low priority.
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_format_get_rank),
	  (gst_v4l2src_probe_caps_for_format):
	  Add more debugging.

2008-08-20 21:54:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacenc.c: Fix compilation against older libflac versions.
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c:
	  Fix compilation against older libflac versions.

2008-08-20 17:46:48 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/pulse/: Use GST_BOILERPLATE everywhere and fix coding style at some places.
	  Original commit message from CVS:
	  * ext/pulse/pulsemixer.c: (gst_pulsemixer_class_init),
	  (gst_pulsemixer_set_property), (gst_pulsemixer_get_property):
	  * ext/pulse/pulsemixerctrl.c: (gst_pulsemixer_ctrl_subscribe_cb),
	  (gst_pulsemixer_ctrl_open), (gst_pulsemixer_ctrl_timeout_event),
	  (gst_pulsemixer_ctrl_set_volume):
	  * ext/pulse/pulsemixertrack.c: (gst_pulsemixer_track_new):
	  * ext/pulse/pulseprobe.c: (gst_pulseprobe_open):
	  * ext/pulse/pulsesink.c: (gst_pulsesink_class_init),
	  (gst_pulsesink_init), (gst_pulsesink_open),
	  (gst_pulsesink_prepare), (gst_pulsesink_write),
	  (gst_pulsesink_delay), (gst_pulsesink_reset):
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
	  (gst_pulsesrc_init):
	  Use GST_BOILERPLATE everywhere and fix coding style at some places.
	  Fix a locking issue in pulsesink's prepare function.
	  * ext/pulse/pulseutil.c: (gst_pulse_channel_map_to_gst):
	  Check if the created channel layout is valid for GStreamer.

2008-08-20 17:42:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspgoogle.c: Things that can happen when your brain is in google mode trying to deal with their google r...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspgoogle.c:
	  Things that can happen when your brain is in google mode trying to
	  deal with their google rtsp server extensions and trying to type your
	  google mail account.

2008-08-20 17:30:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fi...
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/gstrtsp.c: (plugin_init):
	  * gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send),
	  (gst_rtsp_google_after_send), (gst_rtsp_google_get_transports),
	  (_do_init), (gst_rtsp_google_base_init),
	  (gst_rtsp_google_class_init), (gst_rtsp_google_init),
	  (gst_rtsp_google_finalize), (gst_rtsp_google_change_state),
	  (gst_rtsp_google_extension_init):
	  * gst/rtsp/gstrtspgoogle.h:
	  Add google RTSP extension, it can only handle udp and responds with
	  unsupported if we do anything else. Fixes #546465.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send),
	  (gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd),
	  (gst_rtspsrc_create_transports_string),
	  (gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	  (gst_rtspsrc_close), (gst_rtspsrc_pause):
	  Make transport setup code a bit better using GString.
	  Add some more debug.
	  Check for closed connections before doing anything on them.

2008-08-20 17:17:55 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/pulse/: If downstream provides no channel layout and >2 channels should be used use the default layout that pulse...
	  Original commit message from CVS:
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
	  (gst_pulsesrc_create_stream), (gst_pulsesrc_negotiate),
	  (gst_pulsesrc_prepare):
	  * ext/pulse/pulseutil.c: (gst_pulse_gst_to_channel_map),
	  (gst_pulse_channel_map_to_gst):
	  * ext/pulse/pulseutil.h:
	  If downstream provides no channel layout and >2 channels should be
	  used use the default layout that pulseaudio chooses and also
	  add this layout to the caps. Fixes bug #547258.

2008-08-20 11:51:38 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/udp/: Avoid leaking internally allocated file descriptors when setting custom file descriptors. Fixes #543101.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt <pkj at axis com>
	  * gst/udp/gstdynudpsink.c: (gst_dynudpsink_init),
	  (gst_dynudpsink_finalize), (gst_dynudpsink_set_property),
	  (gst_dynudpsink_init_send), (gst_dynudpsink_close):
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init),
	  (gst_multiudpsink_finalize), (gst_multiudpsink_set_property):
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_finalize),
	  (gst_udpsrc_set_property):
	  Avoid leaking internally allocated file descriptors when setting
	  custom file descriptors. Fixes #543101.

2008-08-20 11:48:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Don't try to configure RTCP back to the server when the server did not give us a valid port nu...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
	  Don't try to configure RTCP back to the server when the server did not
	  give us a valid port number.

2008-08-20 10:59:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videobox/gstvideobox.c: Use new basetransform method to renegotiate. Fixes #544956.
	  Original commit message from CVS:
	  * gst/videobox/gstvideobox.c: (gst_video_box_set_property):
	  Use new basetransform method to renegotiate. Fixes #544956.
	  * tests/icles/Makefile.am:
	  * tests/icles/videobox-test.c: (make_pipeline), (main):
	  Add videobox renegotiation example.

2008-08-19 21:03:22 +0000  David Schleef <ds@schleef.org>

	  gst/wavenc/gstwavenc.c: Remove depth ranges and replace with sane values.  Fixes #548530.
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: Remove depth ranges and replace
	  with sane values.  Fixes #548530.

2008-08-18 15:05:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/pulse/: The bytes_per_sample and silence_sample fields of the GstRingBufferSpec are already filled with the corre...
	  Original commit message from CVS:
	  * ext/pulse/pulsesink.c: (gst_pulsesink_prepare):
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_prepare):
	  The bytes_per_sample and silence_sample fields of the GstRingBufferSpec
	  are already filled with the correct values by
	  gst_ring_buffer_parse_caps() so there's no need to set them again
	  with wrong values.

2008-08-16 14:54:56 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: Some AVI 2.0 (ODML) files don't respect the 'specifications' completely and instead of using t...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
	  (gst_avi_demux_read_subindexes_push):
	  Some AVI 2.0 (ODML) files don't respect the 'specifications' completely
	  and instead of using the 'ix##' nomenclature, use '##ix'.
	  They're still valid though, this fixes the duration and indexes for
	  virtually all the ODML files I have.

2008-08-15 17:26:18 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/: Update the vorbis RTP pay/depay to RFC 5215.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps),
	  (gst_rtp_vorbis_depay_process):
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
	  Update the vorbis RTP pay/depay to RFC 5215.
	  Fixes #547842.

2008-08-14 22:07:02 +0000  David Schleef <ds@schleef.org>

	  gst/qtdemux/qtdemux.c: Add 'hdv6' as a HDV format for 1080i/60 with 3:2 pulldown, i.e., 24p.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: Add 'hdv6' as a HDV format for 1080i/60
	  with 3:2 pulldown, i.e., 24p.

2008-08-14 12:47:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/elements/level.c: Fix compilation some more.
	  Original commit message from CVS:
	  * tests/check/elements/level.c: (GST_START_TEST):
	  Fix compilation some more.

2008-08-14 11:44:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Require -base CVS for wavparse acid chunk parsing.
	  Original commit message from CVS:
	  * configure.ac::
	  Require -base CVS for wavparse acid chunk parsing.

2008-08-13 13:57:01 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/pulse/pulsesink.*: Add "device-name" property to pulsesink too and currently commented out and not working suppor...
	  Original commit message from CVS:
	  * ext/pulse/pulsesink.c: (gst_pulsesink_class_init),
	  (gst_pulsesink_init), (gst_pulsesink_finalize),
	  (gst_pulsesink_set_volume), (gst_pulsesink_get_volume),
	  (gst_pulsesink_set_property), (gst_pulsesink_get_property),
	  (gst_pulsesink_prepare), (gst_pulsesink_change_state):
	  * ext/pulse/pulsesink.h:
	  Add "device-name" property to pulsesink too and currently commented
	  out and not working support for a "volume" property.

2008-08-13 13:17:15 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  configure.ac: Remove more cdio stuff (moved to ugly)
	  Original commit message from CVS:
	  * configure.ac:
	  Remove more cdio stuff (moved to ugly)

2008-08-13 12:37:26 +0000  Laszlo Pandy <laszlok2@gmail.com>

	  ext/pulse/pulsesrc.c: Add "device-name" property, which provides a human readable string for the audio device, to mak...
	  Original commit message from CVS:
	  Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
	  (gst_pulsesrc_get_property):
	  Add "device-name" property, which provides a human readable string
	  for the audio device, to make it more consisten with other audio
	  sources. Fixes bug #547519.

2008-08-13 12:34:13 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/pulse/: Improve debugging a bit by including the parent object in pulsemixerctrl and pulseprobe objects and using...
	  Original commit message from CVS:
	  * ext/pulse/pulsemixer.c: (gst_pulsemixer_change_state):
	  * ext/pulse/pulsemixerctrl.c: (gst_pulsemixer_ctrl_subscribe_cb),
	  (gst_pulsemixer_ctrl_open), (gst_pulsemixer_ctrl_new),
	  (gst_pulsemixer_ctrl_free), (gst_pulsemixer_ctrl_timeout_event):
	  * ext/pulse/pulsemixerctrl.h:
	  * ext/pulse/pulseprobe.c: (gst_pulseprobe_open),
	  (gst_pulseprobe_enumerate), (gst_pulseprobe_new),
	  (gst_pulseprobe_free), (gst_pulseprobe_needs_probe),
	  (gst_pulseprobe_probe_property), (gst_pulseprobe_get_values):
	  * ext/pulse/pulseprobe.h:
	  * ext/pulse/pulsesink.c: (gst_pulsesink_init):
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_init), (gst_pulsesrc_delay),
	  (gst_pulsesrc_change_state):
	  Improve debugging a bit by including the parent object in pulsemixerctrl
	  and pulseprobe objects and using GST_WARNING_OBJECT instead of
	  GST_WARNING.
	  Use the parent GObject subclass instead of a random struct as GObject
	  parameter for G_OBJECT_WARN_INVALID_PROPERTY_ID. This fixes a crash
	  when probing for another property than "device".

2008-08-13 12:21:22 +0000  Laszlo Pandy <laszlok2@gmail.com>

	  ext/pulse/pulsemixer.c: Fix property probing after the device property is set by calling set_server when the server p...
	  Original commit message from CVS:
	  Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
	  * ext/pulse/pulsemixer.c: (gst_pulsemixer_set_property):
	  Fix property probing after the device property is set by calling
	  set_server when the server property changes. Fixes bug #547518.

2008-08-13 12:11:34 +0000  Laszlo Pandy <laszlok2@gmail.com>

	  ext/pulse/pulsemixer.c: Fix property probing after the device property is set by calling set_server when the server p...
	  Original commit message from CVS:
	  Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
	  * ext/pulse/pulsemixer.c: (gst_pulsemixer_set_property):
	  Fix property probing after the device property is set by calling
	  set_server when the server property changes. Fixes bug #547518.

2008-08-13 12:01:01 +0000  Laszlo Pandy <laszlok2@gmail.com>

	  ext/pulse/: Implement GstPropertyProbe interface on pulsesink for detecting sink devices and on pulsesrc for detectin...
	  Original commit message from CVS:
	  Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
	  * ext/pulse/pulsesink.c: (gst_pulsesink_interface_supported),
	  (gst_pulsesink_implements_interface_init),
	  (gst_pulsesink_init_interfaces), (gst_pulsesink_init),
	  (gst_pulsesink_finalize), (gst_pulsesink_set_property),
	  (gst_pulsesink_get_type):
	  * ext/pulse/pulsesink.h:
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported),
	  (gst_pulsesrc_init_interfaces), (gst_pulsesrc_init),
	  (gst_pulsesrc_finalize), (gst_pulsesrc_set_property):
	  * ext/pulse/pulsesrc.h:
	  Implement GstPropertyProbe interface on pulsesink for detecting
	  sink devices and on pulsesrc for detecting source devices.
	  Fixes bugs #547227 and #547217.

2008-08-13 09:17:20 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.c: Don't terminate on fabs(in)>1.0. Init doubles as doubles.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c:
	  Don't terminate on fabs(in)>1.0. Init doubles as doubles.

2008-08-13 08:33:57 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/v4l2/gstv4l2src.c: Properly set the maximum latency value, in the same way it is done in v4lsrc.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_query):
	  Properly set the maximum latency value, in the same way it is done in
	  v4lsrc.
	  * sys/v4l2/v4l2src_calls.c:
	  Simplify fraction equality check, no need to use GValues for this.

2008-08-12 12:04:24 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/v4l2/gstv4l2src.c: Add warning messages stating exactly why the latency query failed.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_query):
	  Add warning messages stating exactly why the latency query failed.
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture):
	  In some cases, the negotiated framerate might be the default one which
	  is already set internally. But we still need to mark it down in fps_n
	  and fps_d so that the latency query can happen properly.

2008-08-12 11:28:47 +0000  Edward Hervey <bilboed@bilboed.com>

	  docs/plugins/inspect/plugin-1394.xml: Whoops, forgot one doc file for people who can't/don't build the raw1394 plugin.
	  Original commit message from CVS:
	  * docs/plugins/inspect/plugin-1394.xml:
	  Whoops, forgot one doc file for people who can't/don't build the
	  raw1394 plugin.

2008-08-12 09:22:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Pull changes from 0.10.9.2 pre-release branch moving the libcdio
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * ext/Makefile.am:
	  * ext/cdio/Makefile.am:
	  * ext/cdio/gstcdio.c:
	  * ext/cdio/gstcdio.h:
	  * ext/cdio/gstcdiocddasrc.c:
	  * ext/cdio/gstcdiocddasrc.h:
	  Pull changes from 0.10.9.2 pre-release branch moving the libcdio
	  CDDA source to -ugly.
	  * po/LINGUAS:
	  * po/POTFILES.in:
	  * po/id.po:
	  Pull in new translation from 0.10.9.2 release branch.

2008-08-11 15:05:13 +0000  Edward Hervey <bilboed@bilboed.com>

	  docs/plugins/: Integrate documentation for new hdv1394src element.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  Integrate documentation for new hdv1394src element.

2008-08-11 14:36:13 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/raw1394/: mpeg2-ts (HDV) variant of firewire capture element.
	  Original commit message from CVS:
	  * ext/raw1394/Makefile.am:
	  * ext/raw1394/gst1394.c: (plugin_init):
	  * ext/raw1394/gsthdv1394src.c: (_do_init),
	  (gst_hdv1394src_base_init), (gst_hdv1394src_class_init),
	  (gst_hdv1394src_init), (gst_hdv1394src_dispose),
	  (gst_hdv1394src_set_property), (gst_hdv1394src_get_property),
	  (gst_hdv1394src_from_raw1394handle),
	  (gst_hdv1394src_iec61883_receive), (gst_hdv1394src_bus_reset),
	  (gst_hdv1394src_create), (gst_hdv1394src_discover_avc_node),
	  (gst_hdv1394src_start), (gst_hdv1394src_stop),
	  (gst_hdv1394src_unlock), (gst_hdv1394src_update_device_name),
	  (gst_hdv1394src_uri_get_type), (gst_hdv1394src_uri_get_protocols),
	  (gst_hdv1394src_uri_get_uri), (gst_hdv1394src_uri_set_uri),
	  (gst_hdv1394src_uri_handler_init):
	  * ext/raw1394/gsthdv1394src.h:
	  mpeg2-ts (HDV) variant of firewire capture element.
	  Fixes #350830

2008-08-11 10:53:06 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/level/gstlevel.c: Fix compilation (also known as the classic 'fix code that someone committed without compiling i...
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_message_new):
	  Fix compilation (also known as the classic 'fix code that someone
	  committed without compiling it first').

2008-08-10 19:40:27 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/level.c: Add a test for level in stereo mode.
	  Original commit message from CVS:
	  * tests/check/elements/level.c:
	  Add a test for level in stereo mode.

2008-08-10 19:35:05 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/examples/spectrum/: Demo how to draw analyzer results synced to the clock.
	  Original commit message from CVS:
	  * tests/examples/spectrum/demo-audiotest.c:
	  * tests/examples/spectrum/demo-osssrc.c:
	  Demo how to draw analyzer results synced to the clock.

2008-08-10 15:52:42 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/level/gstlevel.c: Little renaming (l -> level).
	  Original commit message from CVS:
	  * gst/level/gstlevel.c:
	  Little renaming (l -> level).
	  * gst/spectrum/gstspectrum.c:
	  * gst/spectrum/gstspectrum.h:
	  Also send full timestamp/duration details here.

2008-08-10 11:32:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/level/gstlevel.*: Send same timestamp/duration details as videoanalysis. This gives applications better chance to...
	  Original commit message from CVS:
	  * gst/level/gstlevel.c:
	  * gst/level/gstlevel.h:
	  Send same timestamp/duration details as videoanalysis. This gives
	  applications better chance to sync analysis results with playback.

2008-08-09 14:02:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.c: We need to drop one additional buffer for FLAC as the fLaC marker and STREAMINFO block a...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_handle_sink_event),
	  (flac_streamheader_to_codecdata):
	  We need to drop one additional buffer for FLAC as the fLaC
	  marker and STREAMINFO block are merged into one buffer in the caps.
	  Also don't pretend to support NEWSEGMENT events, otherwise we
	  will most probably write some invalid data.

2008-08-09 13:48:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.c: Add support for muxing FLAC into Matroska containers.
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (flac_streamheader_to_codecdata),
	  (gst_matroska_mux_audio_pad_setcaps):
	  Add support for muxing FLAC into Matroska containers.
	  Fixes bug #311586.

2008-08-09 08:58:26 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacenc.c: Actually provide the variables required for the format string.
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_check_discont):
	  Actually provide the variables required for the format string.

2008-08-08 16:20:26 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.*: Close the current segment if we're doing a non-flushing seek and send the close-segmen...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
	  (gst_matroska_demux_element_send_event),
	  (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop):
	  * gst/matroska/matroska-demux.h:
	  Close the current segment if we're doing a non-flushing seek and send
	  the close-segment and the new segment of the seek from the streaming
	  thread.

2008-08-08 15:20:24 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacenc.*: Handle non-zero start timestamps correctly, mark header packets as
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_write_callback),
	  (gst_flac_enc_check_discont), (gst_flac_enc_chain),
	  (gst_flac_enc_change_state):
	  * ext/flac/gstflacenc.h:
	  Handle non-zero start timestamps correctly, mark header packets as
	  IN_CAPS and print a warning and suggest using audiorate if stream
	  discontinuities are detected. When FLAC supports flushing the encoder
	  somehow this should be done for discontinuities instead.
	  Remove some unused variables from the instance struct.

2008-08-07 17:14:39 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add pulseaudio to plugins list in spec file
	  Original commit message from CVS:
	  add pulseaudio to plugins list in spec file

2008-08-07 16:14:42 +0000  Frederic Crozat <fcrozat@mandriva.org>

	  Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
	  Original commit message from CVS:
	  Patch by: Frederic Crozat <fcrozat@mandriva.org>
	  * ext/dvdread/dvdreadsrc.c: (plugin_init):
	  * ext/lame/gstlame.c: (plugin_init):
	  * gst/asfdemux/gstasf.c: (plugin_init):
	  Make sure gettext returns translations in UTF-8 encoding rather
	  than in the current locale encoding (#546822).

2008-08-07 16:13:41 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacenc.c: If seeking failed return the appropiate return value to FLAC.
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback):
	  If seeking failed return the appropiate return value to FLAC.
	  Otherwise it thinks seeking was successfull and tries to rewrite
	  parts of the headers which then get appended to the output.

2008-08-07 16:11:00 +0000  Frederic Crozat <fcrozat@mandriva.org>

	  Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
	  Original commit message from CVS:
	  Patch by: Frederic Crozat <fcrozat@mandriva.org>
	  * ext/esd/gstesd.c: (plugin_init):
	  * ext/flac/gstflac.c: (plugin_init):
	  * ext/shout2/gstshout2.c: (plugin_init):
	  * ext/wavpack/gstwavpack.c: (plugin_init):
	  * sys/oss/gstossaudio.c: (plugin_init):
	  * sys/v4l2/gstv4l2.c: (plugin_init):
	  Make sure gettext returns translations in UTF-8 encoding rather
	  than in the current locale encoding (#546822).

2008-08-07 14:40:13 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacdec.c: Add FIXME for 0.11 to simply output everything with width=32 as given by FLAC and let audiocon...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c:
	  Add FIXME for 0.11 to simply output everything with width=32 as given
	  by FLAC and let audioconvert handle the conversions instead of doing
	  them in flacdec.

2008-08-07 10:22:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/v4l2/v4l2src_calls.c: When outputting a pad template range for the size, include a framerate range too, to avoid ...
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format):
	  When outputting a pad template range for the size, include a framerate
	  range too, to avoid 'not a real subset of template caps' errors.

2008-08-06 15:34:55 +0000  Jonathan Matthew <notverysmart@gmail.com>

	  ext/flac/: Port flactag to 0.10, add documentation for it and clean it up a bit.
	  Original commit message from CVS:
	  Based on a patch by: Jonathan Matthew <notverysmart at gmail dot com>
	  * ext/flac/Makefile.am:
	  * ext/flac/gstflac.c: (plugin_init):
	  * ext/flac/gstflactag.c: (gst_flac_tag_setup_interfaces),
	  (gst_flac_tag_base_init), (gst_flac_tag_class_init),
	  (gst_flac_tag_dispose), (gst_flac_tag_init),
	  (gst_flac_tag_sink_setcaps), (gst_flac_tag_chain),
	  (gst_flac_tag_change_state):
	  * ext/flac/gstflactag.h:
	  Port flactag to 0.10, add documentation for it and clean it up a bit.
	  Fixes bug #413841.
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/gst-plugins-good-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_base_init):
	  * ext/flac/gstflacdec.h:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_base_init):
	  * ext/flac/gstflacenc.h:
	  Add flactag and flacenc to the documentation and mark
	  the private parts of the flacdec instance structure as private.
	  Also use gst_element_class_set_details_simple() in flacdec and
	  flacenc.

2008-08-06 13:12:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/qtdemux.c: Use audio/x-qdm for caps. Collect some info - mplayer has a decoder for it but ffmpeg does not.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c:
	  Use audio/x-qdm for caps. Collect some info - mplayer has a decoder
	  for it but ffmpeg does not.

2008-08-05 15:05:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: Handle the list chunk and use gst_riff_parse_info() to parse the info sub-chunk.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Handle the list chunk and use gst_riff_parse_info() to parse the info
	  sub-chunk.

2008-08-05 14:22:12 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: Handle the acid chunk and send tempo as part of tags. Other fields are interesting too, b...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Handle the acid chunk and send tempo as part of tags. Other fields are
	  interesting too, but need more tag-definitions. Fixes #545433.

2008-08-05 14:16:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: Refactor wavparse. Call _reset() from dispose() and move old code from dispose into reset...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Refactor wavparse. Call _reset() from dispose() and move old code from
	  dispose into reset. This way we don't leak taglists when we abort
	  parsing. Fix some comments. Move code for skipping a chunk into extra
	  function. Replace chunk sizes with a const to ease readability.

2008-08-05 13:57:57 +0000  Aurelien Grimaud <gstelzz@yahoo.fr>

	  gst/rtsp/gstrtspsrc.c: Improve udp port setup. Fixes #545710.
	  Original commit message from CVS:
	  Patch by: Aurelien Grimaud <gstelzz at yahoo dot fr>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_alloc_udp_ports):
	  Improve udp port setup. Fixes #545710.

2008-08-05 13:54:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Add MP1S depayloader.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_base_init),
	  (gst_rtp_mp1s_depay_class_init), (gst_rtp_mp1s_depay_init),
	  (gst_rtp_mp1s_depay_setcaps), (gst_rtp_mp1s_depay_process),
	  (gst_rtp_mp1s_depay_set_property),
	  (gst_rtp_mp1s_depay_get_property),
	  (gst_rtp_mp1s_depay_change_state),
	  (gst_rtp_mp1s_depay_plugin_init):
	  * gst/rtp/gstrtpmp1sdepay.h:
	  Add MP1S depayloader.
	  * gst/rtsp/URLS:
	  Some more sample rtsp streams.

2008-08-05 08:43:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/URLS: Add another URL.
	  Original commit message from CVS:
	  * gst/rtsp/URLS:
	  Add another URL.
	  * tests/check/elements/id3v2mux.c: (test_taglib_id3mux_with_tags):
	  * tests/check/elements/rglimiter.c: (GST_START_TEST):
	  Add some more debug info.

2008-08-04 09:16:40 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/gstavimux.c: Provide cbSize field for audio extra_data size, and take care to pad extra_data.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
	  Provide cbSize field for audio extra_data size, and take care to
	  pad extra_data.

2008-08-04 07:23:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/qtdemux.c: Return the result of gst_pad_{start,stop}_task instead of hard-coded
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c:
	  Return the result of gst_pad_{start,stop}_task instead of hard-coded
	  TRUE.

2008-08-04 07:17:38 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/: Add keyword tag support. Fixes #520694 for qtdemux.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c:
	  * gst/qtdemux/qtdemux_fourcc.h:
	  Add keyword tag support. Fixes #520694 for qtdemux.

2008-08-04 07:05:33 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/qtdemux.c: Add support for tmpo tag (BPM).
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c:
	  Add support for tmpo tag (BPM).

2008-08-03 12:23:49 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacenc.c: Set an estimate for the total number of samples that will be encoded if possible to help decod...
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_query_peer_total_samples),
	  (gst_flac_enc_sink_setcaps), (gst_flac_enc_write_callback):
	  Set an estimate for the total number of samples that will be encoded
	  if possible to help decoders if the streaminfo can't be rewritten
	  later (like when muxing into Ogg containers).
	  Add a warning if we get header packets after data packets as those
	  will get lost when muxing into Ogg, i.e. rewriting the headers doesn't
	  work.

2008-08-03 11:38:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacdec.c: Support decoding of all depths between 4 and 32 bits and read the depth from the streaminfo he...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_metadata_callback),
	  (gst_flac_dec_write):
	  Support decoding of all depths between 4 and 32 bits and read the
	  depth from the streaminfo header if needed. Also support all sampling
	  rates between 1 and 655350 Hz.
	  * ext/flac/gstflacenc.c:
	  (gst_flac_enc_caps_append_structure_with_widths),
	  (gst_flac_enc_sink_getcaps), (gst_flac_enc_sink_setcaps),
	  (gst_flac_enc_chain):
	  * ext/flac/gstflacenc.h:
	  Support encoding in all bit depths supported by the streamable
	  subformat (i.e. 8, 12, 16, 20 and 24 bits) and all sampling rates
	  between 1 Hz and 655350 Hz.

2008-08-03 09:23:14 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacenc.c: Support encoding of up to 8 channels.
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_init),
	  (gst_flac_enc_sink_getcaps):
	  Support encoding of up to 8 channels.

2008-08-02 21:39:01 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.*: Fix seeking race condition in #540300
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c:
	  * ext/soup/gstsouphttpsrc.h:
	  Fix seeking race condition in #540300
	  Patch By: Wouter Cloetens  <wouter at mind be>

2008-08-02 18:35:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: When receiving a SEEK event on a specific pad first search for a seek table entry for ...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek),
	  (gst_matroska_demux_element_send_event),
	  (gst_matroska_demux_handle_seek_event),
	  (gst_matroska_demux_handle_src_event):
	  When receiving a SEEK event on a specific pad first search for a seek
	  table entry for the stream of the pad and then fall back to an entry
	  for a different stream.

2008-08-02 18:20:44 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Build depend on core CVS for the attachment tag.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/matroska/matroska-ids.c: (gst_matroska_register_tags):
	  * gst/matroska/matroska-ids.h:
	  Build depend on core CVS for the attachment tag.

2008-08-02 18:18:05 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Decode the codec private data and following ContentEncoding if necessary.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/matroska/Makefile.am:
	  * gst/matroska/lzo.c: (get_byte), (get_len), (copy),
	  (copy_backptr), (lzo1x_decode), (main):
	  * gst/matroska/lzo.h:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_read_track_encoding),
	  (gst_matroska_decompress_data), (gst_matroska_decode_data),
	  (gst_matroska_decode_buffer),
	  (gst_matroska_decode_content_encodings),
	  (gst_matroska_demux_read_track_encodings),
	  (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock):
	  * gst/matroska/matroska-ids.h:
	  Decode the codec private data and following ContentEncoding if
	  necessary.
	  Support bzip2, lzo and header stripped compression. For lzo use the
	  ffmpeg lzo implementation as liblzo is GPL licensed.
	  Fix zlib decompression.

2008-08-02 18:11:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.c: Fix muxing of MP3/MP2 with different MPEG versions by calculating the duration of a fram...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_audio_pad_setcaps):
	  Fix muxing of MP3/MP2 with different MPEG versions by calculating the
	  duration of a frame with the new mpegaudioversion caps field.

2008-08-02 18:06:20 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.*: Allow an infinite number of stream inside Matroska containers and use a GPtrArray for ...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_finalize),
	  (gst_matroska_demux_class_init), (gst_matroska_demux_init),
	  (gst_matroska_demux_combine_flows), (gst_matroska_demux_reset),
	  (gst_matroska_demux_stream_from_num),
	  (gst_matroska_demux_tracknumber_unique),
	  (gst_matroska_demux_add_stream), (gst_matroska_demux_send_event),
	  (gst_matroska_demux_handle_seek_event),
	  (gst_matroska_demux_sync_streams),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_loop):
	  * gst/matroska/matroska-demux.h:
	  Allow an infinite number of stream inside Matroska containers and use
	  a GPtrArray for storing them instead of allowing "only" 127 streams.

2008-08-02 18:01:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Fix indention everywhere. A broken indent version has added newlines after every single declaration so...
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_read_class_init),
	  (gst_ebml_read_change_state), (gst_ebml_read_element_level_up),
	  (gst_ebml_read_peek_bytes), (gst_ebml_read_element_id),
	  (gst_ebml_read_element_length), (gst_ebml_peek_id),
	  (gst_ebml_read_get_length), (gst_ebml_read_skip),
	  (gst_ebml_read_buffer), (gst_ebml_read_bytes),
	  (gst_ebml_read_uint), (gst_ebml_read_sint), (_ext2dbl),
	  (gst_ebml_read_float), (gst_ebml_read_ascii), (gst_ebml_read_date),
	  (gst_ebml_read_master), (gst_ebml_read_binary),
	  (gst_ebml_read_header):
	  * gst/matroska/ebml-write.c: (gst_ebml_write_element_id),
	  (gst_ebml_write_element_size), (gst_ebml_write_uint),
	  (gst_ebml_write_sint), (gst_ebml_write_ascii),
	  (gst_ebml_write_master_start), (gst_ebml_write_master_finish),
	  (gst_ebml_replace_uint):
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
	  (gst_matroska_demux_read_track_encoding),
	  (gst_matroska_demux_read_track_encodings),
	  (gst_matroska_demux_add_stream), (gst_matroskademux_do_index_seek),
	  (gst_matroska_demux_send_event),
	  (gst_matroska_demux_element_send_event),
	  (gst_matroska_demux_handle_seek_event),
	  (gst_matroska_demux_handle_src_event),
	  (gst_matroska_demux_init_stream),
	  (gst_matroska_demux_parse_tracks),
	  (gst_matroska_demux_parse_index_cuetrack),
	  (gst_matroska_demux_parse_index_pointentry),
	  (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_parse_metadata_id_simple_tag),
	  (gst_matroska_demux_parse_metadata_id_tag),
	  (gst_matroska_demux_parse_metadata),
	  (gst_matroska_demux_parse_attached_file),
	  (gst_matroska_demux_parse_attachments),
	  (gst_matroska_demux_parse_chapters), (gst_matroska_ebmlnum_uint),
	  (gst_matroska_ebmlnum_sint), (gst_matroska_demux_push_hdr_buf),
	  (gst_matroska_demux_push_flac_codec_priv_data),
	  (gst_matroska_demux_push_xiph_codec_priv_data),
	  (gst_matroska_demux_push_dvd_clut_change_event),
	  (gst_matroska_demux_add_mpeg_seq_header),
	  (gst_matroska_demux_add_wvpk_header),
	  (gst_matroska_demux_check_subtitle_buffer),
	  (gst_matroska_decode_buffer),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_parse_cluster),
	  (gst_matroska_demux_parse_contents_seekentry),
	  (gst_matroska_demux_parse_contents),
	  (gst_matroska_demux_loop_stream_parse_id),
	  (gst_matroska_demux_loop_stream), (gst_matroska_demux_loop),
	  (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
	  (gst_matroska_demux_subtitle_caps),
	  (gst_matroska_demux_change_state):
	  * gst/matroska/matroska-ids.c:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
	  (gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event),
	  (gst_matroska_mux_video_pad_setcaps),
	  (xiph3_streamheader_to_codecdata),
	  (vorbis_streamheader_to_codecdata),
	  (theora_streamheader_to_codecdata),
	  (gst_matroska_mux_audio_pad_setcaps),
	  (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
	  (gst_matroska_mux_track_header), (gst_matroska_mux_start),
	  (gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish),
	  (gst_matroska_mux_best_pad), (gst_matroska_mux_write_data),
	  (gst_matroska_mux_collected), (gst_matroska_mux_change_state):
	  Fix indention everywhere. A broken indent version has added newlines
	  after every single declaration some time ago.

2008-08-02 17:59:05 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: If no Tracks are found error out instead of trying it again until the end of time.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_loop_stream_parse_id):
	  If no Tracks are found error out instead of trying it again until the
	  end of time.

2008-08-02 17:57:31 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: Fix demuxing of raw integer audio. The samples are unsigned only for 8 bit and signed ...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps):
	  Fix demuxing of raw integer audio. The samples are unsigned only for 8
	  bit and signed otherwise, not the other way around.

2008-08-02 17:54:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.c: Add more raw YUV formats to the list of supported formats.
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  Add more raw YUV formats to the list of supported formats.

2008-08-02 17:52:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.c: Add support for muxing raw float audio now that the spec defines the endianness and add ...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_audio_pad_setcaps):
	  Add support for muxing raw float audio now that the spec defines the
	  endianness and add support for muxing raw integer audio with 24 and
	  32 bits.
	  Allow muxing of more than 8 audio channels.

2008-08-02 17:47:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.c: Add locking to the global array of used track UIDs to prevent random crashes if more tha...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_create_uid),
	  (gst_matroska_mux_reset), (gst_matroska_mux_start):
	  Add locking to the global array of used track UIDs to prevent random
	  crashes if more than a single matrosmux instance is used.
	  Use 64 bit values for the track UIDs.
	  Use the global GRandom of GLib instead of creating our own one
	  for the few random numbers we need every single time.

2008-08-02 17:18:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacdec.c: Always post the audio-codec tag, not only if other tags are present.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_setup_seekable_decoder),
	  (gst_flac_dec_setup_stream_decoder),
	  (gst_flac_dec_update_metadata):
	  Always post the audio-codec tag, not only if other tags are present.

2008-08-01 23:26:50 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to development -> 0.10.9.1
	  Original commit message from CVS:
	  * configure.ac:
	  Back to development -> 0.10.9.1

2008-08-01 15:58:47 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add missing gstreamer plugins to spec file
	  Original commit message from CVS:
	  add missing gstreamer plugins to spec file

=== release 0.10.9 ===

2008-07-31 22:10:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 0.10.9
	  Original commit message from CVS:
	  Release 0.10.9

2008-07-31 21:50:44 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2008-07-31 21:26:48 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/soup/gstsouphttpsrc.c: Don't throw an error when soup completes a msg with status 'cancelled', as that indicates ...
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c:
	  Don't throw an error when soup completes a msg with status
	  'cancelled', as that indicates we cancelled a request while
	  shutting down or seeking, and it's not an error.
	  Fixes: #540300 again.

2008-07-31 14:24:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/lame/gstlame.c: Use the default for the strict-iso property too.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_class_init),
	  (gst_lame_get_default_settings):
	  Use the default for the strict-iso property too.
	  Allow a bitrate setting of 0, which lets lame choose the default value
	  and which makes it possible to set the compression-ratio property.

2008-07-29 16:57:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/lame/gstlame.*: Get the defaults settings of LAME in the plugin initialization function and return FALSE here if ...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init),
	  (gst_lame_chain), (gst_lame_get_default_settings), (plugin_init):
	  * ext/lame/gstlame.h:
	  Get the defaults settings of LAME in the plugin initialization
	  function and return FALSE here if something goes wrong. This removes
	  the hacky failing instance init function.
	  Use LAMEs default value for all settings instead of overwriting some
	  of them. Overwriting some of them gives unexpected results if one only
	  sets a preset. Fixes bug #498004.

2008-07-28 20:17:46 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: 0.10.8.4 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  0.10.8.4 pre-release

2008-07-27 15:56:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/lame/gstlame.c: Use LAME's default for the min/max/mean VBR bitrate. Setting our own defaults will restrict the b...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_init):
	  Use LAME's default for the min/max/mean VBR bitrate. Setting our own
	  defaults will restrict the bitrate when using the presets in a bad way.
	  Fixes bug #498004.

2008-07-27 11:01:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Put the MPEG audio version into the caps as "mpegaudioversion".
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_setcaps):
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  (mp3_type_frame_length_from_header), (mp3_caps_create),
	  (gst_mp3parse_chain):
	  Put the MPEG audio version into the caps as "mpegaudioversion".
	  This is different from "mpegversion".

2008-07-25 14:50:03 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Fix segment-stop regression.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment):
	  Fix segment-stop regression.
	  Add documentation regarding segments in quicktime files by Wim Taymans.
	  Fixes #544509

2008-07-24 23:55:58 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: 0.10.8.3 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  0.10.8.3 pre-release
	  * po/LINGUAS:
	  * po/pt_BR.po:
	  Add pt_BR translation

2008-07-23 22:01:20 +0000  Michael Smith <msmith@xiph.org>

	  gst/goom/: Fix build with MSVC: include glib.h to define inline appropriately, use header guards where needed.
	  Original commit message from CVS:
	  * gst/goom/convolve_fx.c:
	  * gst/goom/filters.c:
	  * gst/goom/goom_config.h:
	  * gst/goom/goom_core.c:
	  * gst/goom/goom_tools.h:
	  Fix build with MSVC: include glib.h to define inline appropriately,
	  use header guards where needed.
	  * gst/udp/gstudpnetutils.c:
	  * gst/udp/gstudpsrc.c:
	  Fix build with MSVC: use WSA* constants/functions where appropriate, use
	  g_snprintf rather than snprintf.
	  Fixes #544433.

2008-07-22 18:25:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/lame/gstlame.*: Fix build with lame >= 3.97. The padding type and cwlimit settings are deprecated now and the fun...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init),
	  (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup):
	  * ext/lame/gstlame.h:
	  Fix build with lame >= 3.97. The padding type and cwlimit settings
	  are deprecated now and the function declarations are hidden in the
	  headers so deprecate the GObject properties for them and remove them
	  in 0.11. Fixes bug #544039.

2008-07-22 06:32:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/debug/gsttaginject.*: Sent tags in _transform_ip() instead of _start(). Fixes #543404 partially.
	  Original commit message from CVS:
	  * gst/debug/gsttaginject.c:
	  * gst/debug/gsttaginject.h:
	  Sent tags in _transform_ip() instead of _start(). Fixes #543404
	  partially.

2008-07-19 14:12:39 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: 0.10.8.2 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  0.10.8.2 pre-release

2008-07-19 13:50:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/Makefile.am: Finish hooking up pulseaudio plugin to the build.
	  Original commit message from CVS:
	  * ext/Makefile.am:
	  Finish hooking up pulseaudio plugin to the build.
	  * ext/pulse/pulsemixerctrl.c:
	  Fix compilation error.

2008-07-19 13:23:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  po/: Add new lithunian translation, and add french to the LINGUAS file.
	  Original commit message from CVS:
	  * po/LINGUAS:
	  * po/lt.po:
	  Add new lithunian translation, and add french to the LINGUAS
	  file.

2008-07-19 13:08:42 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.c: Fix Soup HTTP source seeking.
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c:
	  Fix Soup HTTP source seeking.
	  Patch By: Wouter Cloetens  <wouter at mind be>
	  Fixes: #540300
	  * tests/check/elements/.cvsignore:
	  Ignore new check programs.

2008-07-19 01:01:13 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move replaygain and interleave plugins from -bad.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/gst-plugins-good-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-interleave.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * tests/check/Makefile.am:
	  Move replaygain and interleave plugins from -bad.
	  Fixes: #543406
	  Fixes: #536228

2008-07-18 20:03:07 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/qtdemux/qtdemux.c: Revert ISO base media spec based pixel-aspect-ratio calculation.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
	  (qtdemux_parse_trak):
	  Revert ISO base media spec based pixel-aspect-ratio calculation.
	  Fixes #543300.

2008-07-17 16:42:53 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/osxvideo/osxvideosink.m: Fix minor build issues on macosx.
	  Original commit message from CVS:
	  * sys/osxvideo/osxvideosink.m:
	  Fix minor build issues on macosx.
	  Fixes #543054

2008-07-17 14:40:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Only use -Wno-attributes (which is there to work around a bug in the taglib 1.5 headers) if the c++ compiler actually...
	  Original commit message from CVS:
	  * configure.ac::
	  * ext/taglib/Makefile.am::
	  Only use -Wno-attributes (which is there to work around a
	  bug in the taglib 1.5 headers) if the c++ compiler actually
	  supports it (#543255).

2008-07-17 13:54:38 +0000  Benoit Fouet <benoit.fouet@purplelabs.com>

	  sys/v4l2/gstv4l2src.c: Avoid compiler warning by initialising variable to NULL (#543259).
	  Original commit message from CVS:
	  Patch by: Benoit Fouet <benoit.fouet purplelabs com>
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_negotiate):
	  Avoid compiler warning by initialising variable to NULL (#543259).

2008-07-14 17:17:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/debug/gsttaginject.c: Don't pass NULL taglists to gst_tag_list_is_empty().
	  Original commit message from CVS:
	  * gst/debug/gsttaginject.c: (gst_tag_inject_start):
	  Don't pass NULL taglists to gst_tag_list_is_empty().

2008-07-14 17:15:42 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/: Don't use declarations after statements.
	  Original commit message from CVS:
	  * tests/check/elements/cmmldec.c: (GST_START_TEST):
	  * tests/check/elements/rtp-payloading.c: (rtp_pipeline_create),
	  (rtp_pipeline_run):
	  * tests/check/elements/souphttpsrc.c: (souphttpsrc_suite):
	  Don't use declarations after statements.

2008-07-14 16:28:25 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  ext/jpeg/gstjpegdec.c: Align documentation with reality.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c:
	  Align documentation with reality.

2008-07-14 13:11:14 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/udp/gstudpnetutils.c: EAI_ADDRFAMILY was obsoleted in BSD at some point. Define it to the old value (1) if it's n...
	  Original commit message from CVS:
	  * gst/udp/gstudpnetutils.c:
	  EAI_ADDRFAMILY was obsoleted in BSD at some point. Define it to the
	  old value (1) if it's not defined which should not cause any problems
	  as we're using it internal only anyway.

2008-07-14 13:02:48 +0000  Alessandro Decina <alessandro@nnva.org>

	  gst/avi/gstavidemux.c: Fix build of avidemux on big endian architectures.
	  Original commit message from CVS:
	  Patch by: Alessandro Decina <alessandro at nnva dot org>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_riff_parse_vprp):
	  Fix build of avidemux on big endian architectures.

2008-07-10 20:47:56 +0000  Thiago Sousa Santos <thiagoss@lcc.ufcg.edu.br>

	  gst/qtdemux/qtdemux.c: Correctly distinguish 8bit vs 16bit raw audio.  Fixes #542410.
	  Original commit message from CVS:
	  Patch by: Thiago Sousa Santos <thiagoss at lcc dot ufcg dot edu dot br>
	  * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
	  Correctly distinguish 8bit vs 16bit raw audio.  Fixes #542410.

2008-07-10 18:51:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Document one more.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.prerequisites:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-stereo.xml:
	  * gst/stereo/gststereo.c:
	  Document one more.

2008-07-08 21:05:18 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/qtdemux/qtdemux.c: Set pixel-aspect-ratio in caps using display width and height provided in track.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
	  (qtdemux_parse_trak):
	  Set pixel-aspect-ratio in caps using display width and height
	  provided in track.

2008-07-08 13:59:51 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Don't include ERROR_CFLAGS in GST_CXXFLAGS as it might include flags that are invalid for C++. Fixes bu...
	  Original commit message from CVS:
	  * configure.ac:
	  Don't include ERROR_CFLAGS in GST_CXXFLAGS as it might include
	  flags that are invalid for C++. Fixes bug #516509.

2008-07-08 12:51:34 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Don't use declarations after statements and variable length arrays.
	  Original commit message from CVS:
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri):
	  * ext/speex/gstspeexenc.c: (gst_speex_enc_sink_getcaps):
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_set_wp_config):
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_fixate):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format):
	  * tests/examples/equalizer/demo.c: (message_handler):
	  * tests/examples/spectrum/demo-audiotest.c: (message_handler):
	  * tests/examples/spectrum/demo-osssrc.c: (message_handler):
	  Don't use declarations after statements and variable length arrays.

2008-07-07 21:28:58 +0000  Daniel Drake <dsd@gentoo.org>

	  sys/v4l2/v4l2src_calls.c: Try progressive video if interlaced fails. Fixes bug #541956 and the usage of v4l2src on OLPC.
	  Original commit message from CVS:
	  Patch by: Daniel Drake <dsd at gentoo dot org>
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture),
	  (gst_v4l2src_get_nearest_size):
	  Try progressive video if interlaced fails. Fixes bug #541956
	  and the usage of v4l2src on OLPC.

2008-07-07 15:34:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/rtp/gstrtpspeexdepay.*: Revert last change: Only the jitterbuffer is able to convert RTP to
	  Original commit message from CVS:
	  * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init),
	  (gst_rtp_speex_depay_process):
	  * gst/rtp/gstrtpspeexdepay.h:
	  Revert last change: Only the jitterbuffer is able to convert RTP to
	  Gstreamer timestamps and normal (de)payloaders should simply copy it.
	  Reopens bug #541787.

2008-07-07 10:30:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/rtp/gstrtpvrawdepay.c: Include stdlib.h for atoi().
	  Original commit message from CVS:
	  * gst/rtp/gstrtpvrawdepay.c:
	  Include stdlib.h for atoi().
	  * gst/rtsp/gstrtspsrc.c:
	  Use floating point math for latencies < 0 sec in log output.

2008-07-07 10:16:07 +0000  Tomasz Grobelny <tomasz@grobelny.oswiecenia.net>

	  gst/rtp/gstrtpspeexdepay.*: Take timestamp from the RTP packet as a first step to fix problems with transmission over...
	  Original commit message from CVS:
	  Patch by: Tomasz Grobelny <tomasz at grobelny dot oswiecenia dot net>
	  * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init),
	  (gst_rtp_speex_depay_process):
	  * gst/rtp/gstrtpspeexdepay.h:
	  Take timestamp from the RTP packet as a first step to fix problems
	  with transmission over RTP when the network is not reliable.
	  Fixes bug #541787.

2008-07-05 19:01:28 +0000  Tero Saarni <tero.saarni@gmail.com>

	  gst/udp/gstudpsrc.c: Fix parsing of udp:// URIs containing IPv6 addresses.
	  Original commit message from CVS:
	  Patch by: Tero Saarni <tero dot saarni at gmail dot com>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_set_uri):
	  Fix parsing of udp:// URIs containing IPv6 addresses.
	  Fixes bug #541650.

2008-07-04 20:43:07 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  ext/gdk_pixbuf/gstgdkpixbuf.c: Do not leak incoming buffers.
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain):
	  Do not leak incoming buffers.

2008-07-03 19:27:53 +0000  Damien Lespiau <damien.lespiau@gmail.com>

	  configure.ac: Fix build of the RTP plugin with mingw32 by linking to ws2_32 for htons() and htonl(). Fixes bug #541412.
	  Original commit message from CVS:
	  Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
	  * configure.ac:
	  Fix build of the RTP plugin with mingw32 by linking to ws2_32
	  for htons() and htonl(). Fixes bug #541412.

2008-07-02 09:51:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: Handle position and duration query in DEFAULT format if the pad's track has a default ...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
	  (gst_matroska_demux_add_stream), (gst_matroska_demux_query),
	  (gst_matroska_demux_element_query),
	  (gst_matroska_demux_handle_src_query),
	  (gst_matroska_demux_handle_seek_event):
	  Handle position and duration query in DEFAULT format if the
	  pad's track has a default frame duration set.
	  Fix seeking now that the segment's duration doesn't contain the
	  (possibly wrong or inaccurate) duration of the Matroska file.

2008-07-02 09:04:50 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/ebml-read.c: Use NAN constant instead of 0.0/0.0 if possible. NAN is defined in math.h except on MSVC wh...
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (_ext2dbl):
	  Use NAN constant instead of 0.0/0.0 if possible. NAN is defined
	  in math.h except on MSVC where it is defined in xmath.h.
	  Fixes compilation with MSVC.

2008-07-02 08:57:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.*: Don't set the segment duration to the duration from the Matroska header as this value ...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
	  (gst_matroska_demux_handle_src_query),
	  (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_loop_stream_parse_id):
	  * gst/matroska/matroska-demux.h:
	  Don't set the segment duration to the duration from the Matroska
	  header as this value could be wrong and is just informational.

2008-07-02 08:47:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: If no Tracks element is found until the first Cluster is found search it and error out...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_loop_stream_parse_id):
	  If no Tracks element is found until the first Cluster is found
	  search it and error out if none is found in the complete file.

2008-07-02 08:14:35 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: Resync non-subtitle tracks too if a too large gap compared to other tracks is detected.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams):
	  Resync non-subtitle tracks too if a too large gap compared to other
	  tracks is detected.

2008-07-01 13:28:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Add raw video pay and depayloaders, see RFC4175.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_base_init),
	  (gst_rtp_vraw_depay_class_init), (gst_rtp_vraw_depay_init),
	  (gst_rtp_vraw_depay_setcaps), (gst_rtp_vraw_depay_process),
	  (gst_rtp_vraw_depay_change_state),
	  (gst_rtp_vraw_depay_plugin_init):
	  * gst/rtp/gstrtpvrawdepay.h:
	  * gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_get_type),
	  (gst_rtp_vraw_pay_base_init), (gst_rtp_vraw_pay_class_init),
	  (gst_rtp_vraw_pay_init), (gst_rtp_vraw_pay_finalize),
	  (gst_rtp_vraw_pay_setcaps), (gst_rtp_vraw_pay_handle_buffer),
	  (gst_rtp_vraw_pay_plugin_init):
	  * gst/rtp/gstrtpvrawpay.h:
	  Add raw video pay and depayloaders, see RFC4175.

2008-06-30 22:53:39 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/libpng/gstpngdec.c: Don't return GST_FLOW_ERROR when buffer_alloc fails - return whatever it returned.
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c:
	  Don't return GST_FLOW_ERROR when buffer_alloc fails - return
	  whatever it returned.

2008-06-29 19:52:51 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/avi-ids.h: Add vprp chunk related structures.
	  Original commit message from CVS:
	  * gst/avi/avi-ids.h:
	  Add vprp chunk related structures.
	  * gst/avi/gstavidemux.c: (gst_avi_demux_riff_parse_vprp),
	  (gst_avi_demux_parse_stream):
	  Parse optional vprp chunk and add calculated pixel-aspect-ratio
	  to caps.  Fixes #539482.
	  * gst/avi/gstavimux.h:
	  * gst/avi/gstavimux.c: (gst_avi_mux_pad_reset),
	  (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_riff_get_avi_header):
	  Add a vprp chunk if non-trival pixel-aspect-ratio provided in caps.

2008-06-28 19:31:46 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  tests/check/elements/avimux.c: Adjust avimux unit test according to increased streamheader size.
	  Original commit message from CVS:
	  * tests/check/elements/avimux.c: (check_avimux_pad):
	  Adjust avimux unit test according to increased streamheader size.

2008-06-27 18:11:01 +0000  David Schleef <ds@schleef.org>

	  gst/qtdemux/qtdemux.c: Add Dirac stream type
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: Add Dirac stream type

2008-06-27 15:25:00 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/gstavimux.*: Add 8 bytes to current streamheader to make for a complete one and to make more players happy.  ...
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
	  * gst/avi/gstavimux.h:
	  Add 8 bytes to current streamheader to make for a complete one
	  and to make more players happy.  Fixes #519460.

2008-06-26 16:36:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/v4l2_calls.c: Don't include unused gstv4l2xoverlay.h. Fixes build in case where X11 headers are not installed.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2_calls.c::
	  Don't include unused gstv4l2xoverlay.h. Fixes build
	  in case where X11 headers are not installed.

2008-06-26 10:07:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/dv/gstdv.c: Fix compilation.
	  Original commit message from CVS:
	  * ext/dv/gstdv.c: (plugin_init):
	  Fix compilation.

2008-06-26 09:37:23 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/dv/gstdv.c: Marking rank of dvdec as GST_RANK_MARGINAL since it's the slowest
	  Original commit message from CVS:
	  * ext/dv/gstdv.c: (plugin_init):
	  Marking rank of dvdec as GST_RANK_MARGINAL since it's the slowest
	  DV decoder available.
	  Fixes #532393

2008-06-25 08:12:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/udp/gstudpsrc.c: Call getsockname() after the call to bind() to get updated values for the port, etc. This fixes ...
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  Call getsockname() after the call to bind() to get updated values
	  for the port, etc. This fixes the usage of udpsrc on anonymous
	  binding and it's usage by rtspsrc. Fixes bugs #539372, #539548.
	  Thanks to Aurelien Grimaud for pointing out the obvious fix.

2008-06-25 07:57:26 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/pipelines/wavpack.c: Remove workaround for a bug in identity that is fixed in 0.10.20.
	  Original commit message from CVS:
	  * tests/check/pipelines/wavpack.c: (bus_handler):
	  Remove workaround for a bug in identity that is fixed in 0.10.20.

2008-06-25 06:36:58 +0000  Jason Donenfeld <BugZilla@zx2c4.com>

	  ext/soup/gstsouphttpsrc.c: Fix HTTP auth support with user/password passed via the URI.
	  Original commit message from CVS:
	  Patch by: Jason Donenfeld <BugZilla at zx2c4 dot com>
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_headers_cb):
	  Fix HTTP auth support with user/password passed via the URI.
	  Fixes bug #540067.

2008-06-24 15:42:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Depend on released versions of core and -base.
	  Original commit message from CVS:
	  * configure.ac:
	  Depend on released versions of core and -base.

2008-06-23 16:13:40 +0000  Julien Moutte <julien@moutte.net>

	  gst/matroska/matroska-demux.c: Fix buggy format strings in macros. (makes it build on OS X again...)
	  Original commit message from CVS:
	  2008-06-23  Julien Moutte  <julien@fluendo.com>
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_read_track_encoding),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock): Fix buggy
	  format strings in macros. (makes it build on OS X again...)

2008-06-20 16:24:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/: Added debug.
	  Original commit message from CVS:
	  * gst/rtp/gstrtptheorapay.c:
	  * gst/udp/gstmultiudpsink.c:
	  Added debug.

2008-06-20 15:21:59 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	* common:
	* configure.ac:
	  switch v4l2src from experimental to normal build. Fixes #536831
	  Original commit message from CVS:
	  switch v4l2src from experimental to normal build. Fixes #536831

2008-06-19 11:24:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpg726pay.c: Remove unused variable so that we can compile again.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_setcaps):
	  Remove unused variable so that we can compile again.

2008-06-19 11:06:29 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtp/gstrtpg726pay.c: No need to check for audio/G723 and audio/32KADPCM here as they are no longer supported.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_setcaps):
	  No need to check for audio/G723 and audio/32KADPCM here as they are
	  no longer supported.

2008-06-19 10:58:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Use G_GINT64_CONSTANT, this fixes the duration query on files without known length.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
	  (gst_wavpack_parse_src_query), (gst_wavpack_parse_create_src_pad):
	  Use G_GINT64_CONSTANT, this fixes the duration query on files without
	  known length.

2008-06-19 10:48:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Fix demuxing of WavPack files. Muxing is still broken.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_add_wvpk_header),
	  (gst_matroska_demux_audio_caps):
	  * gst/matroska/matroska-ids.h:
	  Fix demuxing of WavPack files. Muxing is still broken.

2008-06-19 09:12:55 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Add a "vfunc" to the track context for postprocessing frames and convert the wavpack and subtitle post...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_track_free),
	  (gst_matroska_demux_add_mpeg_seq_header),
	  (gst_matroska_demux_add_wvpk_header),
	  (gst_matroska_demux_check_subtitle_buffer),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
	  (gst_matroska_demux_subtitle_caps):
	  * gst/matroska/matroska-ids.h:
	  Add a "vfunc" to the track context for postprocessing frames and
	  convert the wavpack and subtitle postprocessing to this vfunc.
	  Copy buffer flags in those functions to the new buffers too.
	  Parse CodecState elements of Blocks.
	  Add a postprocessing function for MPEG video that adds the sequence
	  header from the codec private data or codec state to the frames if
	  it's not already there.

2008-06-19 08:22:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: If a gap of more than 1/2 second is found in one stream send a
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock):
	  If a gap of more than 1/2 second is found in one stream send a
	  NEWSEGMENT event to not stall the pipeline if the gap is too large.
	  This also fixes Matroska files where the first buffer doesn't start
	  at timestamp 0. Fixes bug #429322.
	  The duration of a block is the default duration multiplied with the
	  number of laces. Every lace is one frame and the default duration
	  is the duration of one frame. This fixes playback of files that use
	  lacing for some tracks.

2008-06-18 20:09:28 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: Update FIXME/TODOs and only ignore EOS at the central, important place instead of seve...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_contents_seekentry):
	  Update FIXME/TODOs and only ignore EOS at the central, important place
	  instead of several places.

2008-06-18 16:55:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpg726pay.c: Fix caps, See #538891.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpg726pay.c:
	  Fix caps, See #538891.

2008-06-18 10:28:20 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: Improve debug output everywhere and fix the EOS logic.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
	  (gst_matroska_demux_stream_from_num),
	  (gst_matroska_demux_encoding_cmp),
	  (gst_matroska_demux_encoding_order_unique),
	  (gst_matroska_demux_read_track_encoding),
	  (gst_matroska_demux_read_track_encodings),
	  (gst_matroska_demux_tracknumber_unique),
	  (gst_matroska_demux_add_stream), (gst_matroska_demux_init_stream),
	  (gst_matroska_demux_parse_tracks),
	  (gst_matroska_demux_parse_index_cuetrack),
	  (gst_matroska_demux_parse_index_pointentry),
	  (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_parse_metadata_id_simple_tag),
	  (gst_matroska_demux_parse_metadata_id_tag),
	  (gst_matroska_demux_parse_metadata),
	  (gst_matroska_demux_parse_attached_file),
	  (gst_matroska_demux_parse_attachments),
	  (gst_matroska_demux_parse_chapters),
	  (gst_matroska_demux_sync_streams), (gst_matroska_decode_buffer),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_parse_cluster),
	  (gst_matroska_demux_parse_contents_seekentry),
	  (gst_matroska_demux_parse_contents),
	  (gst_matroska_demux_loop_stream_parse_id),
	  (gst_matroska_demux_loop):
	  Improve debug output everywhere and fix the EOS logic.
	  Check the values of the ContentEncoding elements more strictly and
	  don't use tracks for which it's invalid.
	  Check that the track number is unique for this stream.
	  Check that seek positions are below G_MAXINT64 as our seeks are
	  int64-based and overflows will fail badly.
	  After seeks also don't push SimpleBlocks until the first one
	  containing a keyframe is found. Before this was done only for normal
	  Blocks.
	  Update some FIXME/TODOs.
	  * gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes),
	  (gst_ebml_read_utf8), (gst_ebml_read_header):
	  Improve debug output.
	  * gst/matroska/matroska-ids.c:
	  (gst_matroska_track_init_video_context):
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_video_pad_setcaps):
	  Remove eye mode and don't parse it anymore. We can't use that
	  information in GStreamer yet so it's useless.

2008-06-18 10:12:57 +0000  mersad <mersad@axis.com>

	  gst/rtp/: Added G726 pay/depayloaders. Fixes #538891.
	  Original commit message from CVS:
	  Patch by: mersad <mersad at axis dot com>
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_base_init),
	  (gst_rtp_g726_depay_class_init), (gst_rtp_g726_depay_init),
	  (gst_rtp_g726_depay_setcaps), (gst_rtp_g726_depay_process),
	  (gst_rtp_g726_depay_plugin_init):
	  * gst/rtp/gstrtpg726depay.h:
	  * gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_base_init),
	  (gst_rtp_g726_pay_class_init), (gst_rtp_g726_pay_init),
	  (gst_rtp_g726_pay_setcaps), (gst_rtp_g726_pay_plugin_init):
	  * gst/rtp/gstrtpg726pay.h:
	  Added G726 pay/depayloaders. Fixes #538891.

2008-06-17 10:14:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/URLS: Some more urls.
	  Original commit message from CVS:
	  * gst/rtsp/URLS:
	  Some more urls.
	  * gst/smpte/barboxwipes.c:
	  Add a comment
	  * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  Fix typo, add audioresample to the pipeline.

2008-06-17 10:05:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libmng/: Somewhat port mngenc and mngdec to 0.10. Does not work yet and has many bits ifdeffed out still.
	  Original commit message from CVS:
	  * ext/libmng/Makefile.am:
	  * ext/libmng/gstmng.c: (plugin_init):
	  * ext/libmng/gstmngdec.c: (gst_mng_dec_base_init),
	  (gst_mng_dec_class_init), (gst_mng_dec_sink_setcaps),
	  (gst_mng_dec_init), (gst_mng_dec_src_getcaps), (gst_mng_dec_loop),
	  (gst_mng_dec_get_property), (gst_mng_dec_set_property),
	  (mngdec_error), (mngdec_openstream), (mngdec_closestream),
	  (gst_mng_dec_sink_event), (mngdec_readdata), (mngdec_settimer),
	  (mngdec_processheader), (mngdec_getcanvasline), (mngdec_refresh),
	  (gst_mng_dec_change_state):
	  * ext/libmng/gstmngdec.h:
	  * ext/libmng/gstmngenc.c: (gst_mng_enc_base_init),
	  (gst_mng_enc_class_init), (gst_mng_enc_sink_setcaps),
	  (gst_mng_enc_init), (gst_mng_enc_chain),
	  (gst_mng_enc_get_property), (gst_mng_enc_set_property):
	  * ext/libmng/gstmngenc.h:
	  Somewhat port mngenc and mngdec to 0.10. Does not work yet and has many
	  bits ifdeffed out still.

2008-06-16 11:34:54 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.c: When comparing index elements with the same time compare their block number.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_index_compare):
	  When comparing index elements with the same time compare their
	  block number.

2008-06-16 11:31:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_attached_file)
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_attached_file)
	  Init variable to NULL to avoid compiler warning.

2008-06-16 10:59:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Parse Attachments and post them as GST_TAG_IMAGE if we detect it as image and otherwise as GST_TAG_ATT...
	  Original commit message from CVS:
	  * gst/matroska/Makefile.am:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
	  (gst_matroska_demux_parse_attached_file),
	  (gst_matroska_demux_parse_attachments),
	  (gst_matroska_demux_parse_contents_seekentry),
	  (gst_matroska_demux_loop_stream_parse_id):
	  * gst/matroska/matroska-demux.h:
	  * gst/matroska/matroska-ids.c: (gst_matroska_register_tags):
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska.c: (plugin_init):
	  Parse Attachments and post them as GST_TAG_IMAGE if we detect
	  it as image and otherwise as GST_TAG_ATTACHMENT. Include filename
	  and description of the attachments in the caps. Fixes bug #537622.

2008-06-16 10:09:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/speex/gstspeexenc.c: Add mode property.
	  Original commit message from CVS:
	  * ext/speex/gstspeexenc.c: (gst_speex_enc_mode_get_type),
	  (gst_speex_enc_class_init), (gst_speex_enc_sink_getcaps),
	  (gst_speex_enc_get_latency), (gst_speex_enc_get_query_types),
	  (gst_speex_enc_src_query), (gst_speex_enc_init),
	  (gst_speex_enc_setup), (gst_speex_enc_push_buffer),
	  (gst_speex_enc_chain), (gst_speex_enc_get_property),
	  (gst_speex_enc_set_property):
	  Add mode property.
	  Some cleanups, add more debug info.
	  Add latency query.

2008-06-16 09:54:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/ebml-read.c: Return GST_FLOW_UNEXPECTED instead of GST_FLOW_ERROR on short reads.
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes):
	  Return GST_FLOW_UNEXPECTED instead of GST_FLOW_ERROR on short reads.
	  If we get less bytes than requested we can't do anything except doing
	  our EOS logic.

2008-06-15 19:09:54 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Use a GArray for storing the Cue (i.e. seek) information, store the CueTrackPositions for every track,...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
	  (gst_matroskademux_do_index_seek),
	  (gst_matroska_demux_parse_index_cuetrack),
	  (gst_matroska_demux_parse_index_pointentry),
	  (gst_matroska_index_compare), (gst_matroska_demux_parse_index),
	  (gst_matroska_demux_parse_metadata):
	  * gst/matroska/matroska-demux.h:
	  * gst/matroska/matroska-ids.h:
	  Use a GArray for storing the Cue (i.e. seek) information, store
	  the CueTrackPositions for every track, store the block number
	  and optimize searching in the array by sorting it after the last
	  element was added.
	  Fix a small memory leak when trying to parse a tags element that was
	  already parsed.

2008-06-15 15:29:29 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.*: Don't write another SeekHead which indexes all Clusters to the end of the file. This isn...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
	  (gst_matroska_mux_start), (gst_matroska_mux_finish),
	  (gst_matroska_mux_write_data):
	  * gst/matroska/matroska-mux.h:
	  Don't write another SeekHead which indexes all Clusters to the end of
	  the file. This isn't useful for anything and just increases filesize.

2008-06-15 15:01:30 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/ebml-read.c: Prevent unaligned memory access when reading floats.
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (_ext2dbl), (gst_ebml_read_float):
	  Prevent unaligned memory access when reading floats.

2008-06-15 14:08:41 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Make sure that every Tags element is only parsed once and it's containing tags are only posted once.
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c:
	  * gst/matroska/ebml-read.h:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
	  (gst_matroska_demux_parse_metadata):
	  * gst/matroska/matroska-demux.h:
	  Make sure that every Tags element is only parsed once and it's
	  containing tags are only posted once.

2008-06-15 09:43:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Handle EBML elements like Void or CRC32 in the EbmlRead base class already. They're not useful in the ...
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_peek_id),
	  (gst_ebml_read_header):
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_parse_tracks),
	  (gst_matroska_demux_parse_index_cuetrack),
	  (gst_matroska_demux_parse_index_pointentry),
	  (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_parse_metadata_id_simple_tag),
	  (gst_matroska_demux_parse_metadata_id_tag),
	  (gst_matroska_demux_parse_metadata),
	  (gst_matroska_demux_parse_attachments),
	  (gst_matroska_demux_parse_chapters),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_parse_cluster),
	  (gst_matroska_demux_parse_contents_seekentry),
	  (gst_matroska_demux_parse_contents),
	  (gst_matroska_demux_loop_stream_parse_id):
	  Handle EBML elements like Void or CRC32 in the EbmlRead base class
	  already. They're not useful in the matroska parser and only cause
	  additional code.

2008-06-14 15:51:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Reverse the level list as we usually are only interested in the first element or want to add a new fir...
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_level_free),
	  (gst_ebml_finalize), (gst_ebml_read_change_state),
	  (gst_ebml_read_element_level_up), (gst_ebml_read_master):
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_contents_seekentry):
	  Reverse the level list as we usually are only interested in the
	  first element or want to add a new first element. Having the
	  first element stored at the end and calling g_list_last() and
	  g_list_append() is more expensive.
	  Also use GSlice for allocating the GstEbmlLevel structs.

2008-06-13 21:13:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/debug/gsttaginject.c: Don't unref NULL taglist in finalize. Don't use c++ style comments.
	  Original commit message from CVS:
	  * gst/debug/gsttaginject.c: (gst_tag_inject_finalize),
	  (gst_tag_inject_class_init), (gst_tag_inject_init):
	  Don't unref NULL taglist in finalize. Don't use c++ style
	  comments.

2008-06-13 19:14:41 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Use gst_value_serialize() and gst_value_deserialize() for transforming tags from some GType to a strin...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_metadata_id_simple_tag):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_write_simple_tag),
	  (gst_matroska_mux_write_data):
	  Use gst_value_serialize() and gst_value_deserialize() for transforming
	  tags from some GType to a string and the other way around. The default
	  transformations in GLib don't include transformations from string to
	  number types.

2008-06-13 19:07:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-demux.*: Only parse Tracks, SeekHead and SegmentInfo elements once but allow
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
	  (gst_matroska_demux_parse_tracks),
	  (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_parse_attachments),
	  (gst_matroska_demux_parse_chapters),
	  (gst_matroska_demux_parse_contents_seekentry),
	  (gst_matroska_demux_loop_stream_parse_id):
	  * gst/matroska/matroska-demux.h:
	  Only parse Tracks, SeekHead and SegmentInfo elements once but allow
	  Tags multiple times. The first ones can appear more than once but must
	  contain the same content as the first for backup purposes so we ignore
	  all but the first one. Tags can appear multiple times with different
	  content.
	  Jump to all elements except Clusters that are available from a
	  SeekHead to make it more likely to have all required informations
	  before getting to the first Clusters.
	  Add dummy functions for parsing Attachments and Chapters.

2008-06-13 14:33:52 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/replaygain/: More doc updates.
	  Original commit message from CVS:
	  * gst/replaygain/gstrganalysis.c:
	  * gst/replaygain/gstrglimiter.c:
	  * gst/replaygain/gstrgvolume.c:
	  More doc updates.

2008-06-13 11:59:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.prerequisites:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-alsaspdif.xml:
	  * docs/plugins/inspect/plugin-amrwb.xml:
	  * docs/plugins/inspect/plugin-app.xml:
	  * docs/plugins/inspect/plugin-bayer.xml:
	  * docs/plugins/inspect/plugin-bz2.xml:
	  * docs/plugins/inspect/plugin-cdaudio.xml:
	  * docs/plugins/inspect/plugin-cdxaparse.xml:
	  * docs/plugins/inspect/plugin-dtsdec.xml:
	  * docs/plugins/inspect/plugin-dvb.xml:
	  * docs/plugins/inspect/plugin-dvdspu.xml:
	  * docs/plugins/inspect/plugin-faac.xml:
	  * docs/plugins/inspect/plugin-faad.xml:
	  * docs/plugins/inspect/plugin-fbdevsink.xml:
	  * docs/plugins/inspect/plugin-festival.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-flvdemux.xml:
	  * docs/plugins/inspect/plugin-freeze.xml:
	  * docs/plugins/inspect/plugin-gsm.xml:
	  * docs/plugins/inspect/plugin-gstinterlace.xml:
	  * docs/plugins/inspect/plugin-gstrtpmanager.xml:
	  * docs/plugins/inspect/plugin-h264parse.xml:
	  * docs/plugins/inspect/plugin-interleave.xml:
	  * docs/plugins/inspect/plugin-jack.xml:
	  * docs/plugins/inspect/plugin-ladspa.xml:
	  * docs/plugins/inspect/plugin-metadata.xml:
	  * docs/plugins/inspect/plugin-mms.xml:
	  * docs/plugins/inspect/plugin-modplug.xml:
	  * docs/plugins/inspect/plugin-mpeg2enc.xml:
	  * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
	  * docs/plugins/inspect/plugin-mpegtsparse.xml:
	  * docs/plugins/inspect/plugin-mpegvideoparse.xml:
	  * docs/plugins/inspect/plugin-musepack.xml:
	  * docs/plugins/inspect/plugin-musicbrainz.xml:
	  * docs/plugins/inspect/plugin-mve.xml:
	  * docs/plugins/inspect/plugin-mythtv.xml
	  * docs/plugins/inspect/plugin-nas.xml:
	  * docs/plugins/inspect/plugin-neon.xml:
	  * docs/plugins/inspect/plugin-nsfdec.xml:
	  * docs/plugins/inspect/plugin-nuvdemux.xml:
	  * docs/plugins/inspect/plugin-oss4.xml
	  * docs/plugins/inspect/plugin-rawparse.xml:
	  * docs/plugins/inspect/plugin-real.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-rfbsrc.xml:
	  * docs/plugins/inspect/plugin-sdl.xml:
	  * docs/plugins/inspect/plugin-sdp.xml:
	  * docs/plugins/inspect/plugin-selector.xml:
	  * docs/plugins/inspect/plugin-sndfile.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-spcdec.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-speexresample.xml:
	  * docs/plugins/inspect/plugin-stereo.xml:
	  * docs/plugins/inspect/plugin-subenc.xml
	  * docs/plugins/inspect/plugin-timidity.xml:
	  * docs/plugins/inspect/plugin-tta.xml:
	  * docs/plugins/inspect/plugin-vcdsrc.xml:
	  * docs/plugins/inspect/plugin-videosignal.xml:
	  * docs/plugins/inspect/plugin-vmnc.xml:
	  * docs/plugins/inspect/plugin-wildmidi.xml:
	  * docs/plugins/inspect/plugin-x264.xml:
	  * docs/plugins/inspect/plugin-xvid.xml:
	  * docs/plugins/inspect/plugin-y4menc.xml:
	  * ext/amrwb/gstamrwbdec.c:
	  * ext/amrwb/gstamrwbenc.c:
	  * ext/amrwb/gstamrwbparse.c:
	  * ext/dc1394/gstdc1394.c:
	  * ext/directfb/dfbvideosink.c:
	  * ext/ivorbis/vorbisdec.c:
	  * ext/jack/gstjackaudiosink.c:
	  * ext/mpeg2enc/gstmpeg2enc.cc:
	  * ext/mplex/gstmplex.cc:
	  * ext/musicbrainz/gsttrm.c:
	  * ext/mythtv/gstmythtvsrc.c:
	  * ext/theora/theoradec.c:
	  * ext/timidity/gsttimidity.c:
	  * ext/timidity/gstwildmidi.c:
	  * gst-libs/gst/app/gstappsink.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/dvdspu/gstdvdspu.c:
	  * gst/festival/gstfestival.c:
	  * gst/freeze/gstfreeze.c:
	  * gst/interleave/deinterleave.c:
	  * gst/interleave/interleave.c:
	  * gst/modplug/gstmodplug.cc:
	  * gst/nuvdemux/gstnuvdemux.c:
	  Add missing elements to docs. Fix doc-markup: use convinience syntax
	  for examples (produces valid docbook), add several refsec2 when we
	  have several titles. Fix some types.

2008-06-13 11:54:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.*: Add property to control automatic join/leave of multicast groups.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	  (gst_udpsrc_create), (gst_udpsrc_set_property),
	  (gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
	  * gst/udp/gstudpsrc.h:
	  Add property to control automatic join/leave of multicast groups.
	  Add G_LIKELY.
	  Remove setting caps on buffers explicitly, basesrc does that for us now.
	  Improve debug info.
	  Convert some non-fatal error into warnings.
	  Use g_ntohs for better portability.
	  Leave multicast groups when stopping.
	  When using external sockets, use getsockname() on them to fill up the
	  addr structure before calling methods that use the structure.
	  Should all fix #536903.
	  API: GstUDPSrc::auto-multicast property

2008-06-13 11:47:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpnetutils.c: Use g_ntohl for better portability.
	  Original commit message from CVS:
	  * gst/udp/gstudpnetutils.c: (gst_udp_is_multicast):
	  Use g_ntohl for better portability.

2008-06-13 11:45:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.c: Fix a typo and do some small cleanups.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send),
	  (gst_multiudpsink_remove):
	  Fix a typo and do some small cleanups.

2008-06-13 09:39:41 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/gstrtptheoradepay.c: Make the delivery-method mandatory on the caps and only accept inline for now.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps):
	  Make the delivery-method mandatory on the caps and only accept inline
	  for now.
	  Reverse strcmp checks for delivery-method.
	  * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps):
	  Make delivery method optional when parsing caps and note this in the
	  caps.
	  Reverse strcmp checks for delivery-method.
	  * gst/rtp/gstrtpvorbispay.c:
	  Update a comment to note that the delivery-method is optional,
	  Fixes #537675.

2008-06-13 06:57:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add missing elements to docs. Restore alphabetical order in section file. Document mad (it was included in docs alrea...
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-ugly-plugins-sections.txt:
	  * ext/a52dec/gsta52dec.c:
	  * ext/amrnb/amrnbdec.c:
	  * ext/amrnb/amrnbenc.c:
	  * ext/amrnb/amrnbparse.c:
	  * ext/lame/gstlame.c:
	  * ext/mad/gstmad.c:
	  * ext/sidplay/gstsiddec.cc:
	  * gst/asfdemux/gstrtspwms.c:
	  * gst/mpegaudioparse/gstxingmux.c:
	  * gst/realmedia/rademux.c:
	  * gst/realmedia/rdtmanager.c:
	  * gst/realmedia/rtspreal.c:
	  * gst/synaesthesia/gstsynaesthesia.c:
	  Add missing elements to docs. Restore alphabetical order in section
	  file. Document mad (it was included in docs already).
	  Fix doc-markup: use convinience syntax for examples
	  (produces valid docbook), add several refsec2 when we have several
	  titles. Fix some types.

2008-06-13 05:52:17 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Do not use short_description in section docs for elements. We extract them from element details and there will be war...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c:
	  * ext/sidplay/gstsiddec.cc:
	  * gst/mpegaudioparse/gstxingmux.c:
	  Do not use short_description in section docs for elements. We extract
	  them from element details and there will be warnings if they differ.

2008-06-12 17:30:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Set udpsrc for receiving data from multicast groups to PAUSED instead of leaving them in READY...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_mcast):
	  Set udpsrc for receiving data from multicast groups to PAUSED instead of
	  leaving them in READY. Fixes #537832.

2008-06-12 12:14:38 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: Simplify code. gst_tag_list_merge() does the NULL checks. Add a FIXME for a random constant in t...
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Simplify code. gst_tag_list_merge() does the NULL checks. Add a FIXME
	  for a random constant in tagmuxing code.

2008-06-11 14:28:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/debug/gsttaginject.*: Now actually adding the new element.
	  Original commit message from CVS:
	  * gst/debug/gsttaginject.c:
	  * gst/debug/gsttaginject.h:
	  Now actually adding the new element.

2008-06-11 14:11:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Remove dummy plugin_init. Remove some undefined entries from doc- section file. Add taginject element and rebuild doc...
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/gst-plugins-good-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gamma.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-goom2k1.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-monoscope.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multifile.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-soup.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-video4linux2.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  * gst/debug/Makefile.am:
	  * gst/debug/breakmydata.c:
	  * gst/debug/efence.c:
	  * gst/debug/gstdebug.c:
	  * gst/debug/gstnavseek.c:
	  * gst/debug/gstpushfilesrc.c:
	  * gst/debug/gstpushfilesrc.h:
	  * gst/debug/negotiation.c:
	  * gst/debug/progressreport.c:
	  * gst/debug/progressreport.h:
	  * gst/debug/rndbuffersize.c:
	  * gst/debug/testplugin.c:
	  Remove dummy plugin_init. Remove some undefined entries from doc-
	  section file. Add taginject element and rebuild docs for it.

2008-06-11 11:27:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/matroska-mux.c: Update the counter for the number of streams when pads are added or removed. This will m...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_request_new_pad),
	  (gst_matroska_mux_release_pad), (gst_matroska_mux_write_data):
	  Update the counter for the number of streams when pads are added or
	  removed. This will make sure that a seek table is generated for
	  files with just one audio stream.

2008-06-11 11:18:23 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/: Add some more tags, improve debugging a bit and make sure that
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_metadata_id_simple_tag):
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_write_simple_tag):
	  Add some more tags, improve debugging a bit and make sure that
	  GValue transformation has succeeded before using the result
	  as a tag.

2008-06-11 08:56:16 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/gstrtptheorapay.c: The Theora RTP payloader only supports the "inline" delievery method so let's declare this...
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtptheorapay.c:
	  The Theora RTP payloader only supports the "inline" delievery method
	  so let's declare this on the caps of the static pad template.
	  Fixes bug #537675.

2008-06-10 17:20:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videomixer/videomixer.c: Remove bogus check.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_fill_queues),
	  (gst_videomixer_blend_buffers), (gst_videomixer_update_queues):
	  Remove bogus check.

2008-06-10 16:25:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videomixer/videomixer.c: Use stream_time to synchronize the object properties.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_fill_queues),
	  (gst_videomixer_blend_buffers):
	  Use stream_time to synchronize the object properties.
	  Use running_time of the master pad to timestamp outgoing buffers.
	  Fix the initial segment event to extend an unknown amount of time.
	  Fixes #537361.

2008-06-10 11:05:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Try to ignore unparsable/unknown streams and give a warning instead of erroring out. Fixes #53...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_parse_index), (gst_avi_demux_massage_index),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_stream_header_push),
	  (gst_avi_demux_stream_header_pull):
	  Try to ignore unparsable/unknown streams and give a warning instead of
	  erroring out. Fixes #537377.

2008-06-10 10:44:53 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/ebml-write.c: Use GDOUBLE_TO_BE() instead of (probably slower) custom code.
	  Original commit message from CVS:
	  * gst/matroska/ebml-write.c: (gst_ebml_write_float):
	  Use GDOUBLE_TO_BE() instead of (probably slower) custom code.
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init),
	  (gst_matroska_demux_class_init), (gst_matroska_demux_init),
	  (gst_matroska_track_free), (gst_matroska_demux_encoding_cmp),
	  (gst_matroska_demux_read_track_encodings),
	  (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_handle_src_query),
	  (gst_matroska_demux_init_stream),
	  (gst_matroska_demux_parse_index_cuetrack),
	  (gst_matroska_demux_parse_index_pointentry),
	  (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_parse_metadata_id_simple_tag),
	  (gst_matroska_demux_parse_metadata),
	  (gst_matroska_demux_add_wvpk_header), (gst_matroska_decode_buffer),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_parse_cluster),
	  (gst_matroska_demux_parse_contents_seekentry),
	  (gst_matroska_demux_loop_stream_parse_id),
	  (gst_matroska_demux_loop), (gst_matroska_demux_video_caps),
	  (gst_matroska_demux_audio_caps),
	  (gst_matroska_demux_subtitle_caps):
	  * gst/matroska/matroska-demux.h:
	  * gst/matroska/matroska-ids.c:
	  (gst_matroska_track_init_subtitle_context):
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init),
	  (gst_matroska_mux_class_init), (gst_matroska_mux_init),
	  (gst_matroska_mux_create_uid), (gst_matroska_mux_reset),
	  (gst_matroska_mux_video_pad_setcaps),
	  (gst_matroska_mux_audio_pad_setcaps),
	  (gst_matroska_mux_subtitle_pad_setcaps),
	  (gst_matroska_mux_request_new_pad),
	  (gst_matroska_mux_track_header), (gst_matroska_mux_start),
	  (gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish),
	  (gst_matroska_mux_write_data), (gst_matroska_mux_collected),
	  (gst_matroska_mux_set_property):
	  Add many FIXMEs/TODOs all over the matroska muxer and demuxer
	  elements, do some checks for valid values in the demuxer, handle
	  tracktimecodescale in the demuxer, set correct default values for all
	  settings in the demuxer, review and add all missing matroska
	  IDs and some more raw YUV formats, and some trivial cleanup.

2008-06-10 08:59:17 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/pulse/: Some smaller cleanup. Use G_PARAM_STATIC_STRINGS, gst_element_class_set_details_simple() and fix coding s...
	  Original commit message from CVS:
	  * ext/pulse/pulsemixer.c: (gst_pulsemixer_base_init),
	  (gst_pulsemixer_class_init):
	  * ext/pulse/pulsesink.c: (gst_pulsesink_base_init),
	  (gst_pulsesink_class_init), (gst_pulsesink_prepare):
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported),
	  (gst_pulsesrc_base_init), (gst_pulsesrc_class_init),
	  (gst_pulsesrc_prepare):
	  Some smaller cleanup. Use G_PARAM_STATIC_STRINGS,
	  gst_element_class_set_details_simple() and fix coding style a bit
	  more.

2008-06-10 08:22:17 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add documentation to the pulseaudio plugin and run make update in docs/plugins.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/gst-plugins-good-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gamma.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-goom2k1.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-monoscope.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multifile.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-pulseaudio.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-soup.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-video4linux2.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  * ext/pulse/plugin.c:
	  * ext/pulse/pulsemixer.c:
	  * ext/pulse/pulsesink.c:
	  * ext/pulse/pulsesrc.c:
	  Add documentation to the pulseaudio plugin and run make update
	  in docs/plugins.

2008-06-10 06:52:44 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/gstsunaudiomixerctrl.c: Improvements for the SunAudio mixer by handling mute as no gain for tracks that ...
	  Original commit message from CVS:
	  Patch by: Brian Cameron <brian.cameron at sun dot com>
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_get_volume),
	  (gst_sunaudiomixer_ctrl_set_volume):
	  Improvements for the SunAudio mixer by handling mute as no gain
	  for tracks that have a gain property but no mute property.
	  Fixes bug #536067.

2008-06-10 06:45:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add pulseaudio GStreamer element from gst-pulse. Development will continue here instead of pulseaudio SVN. Fixes bug ...
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/pulse/Makefile.am:
	  * ext/pulse/plugin.c: (plugin_init):
	  * ext/pulse/pulsemixer.c: (gst_pulsemixer_interface_supported),
	  (gst_pulsemixer_implements_interface_init),
	  (gst_pulsemixer_init_interfaces), (gst_pulsemixer_base_init),
	  (gst_pulsemixer_class_init), (gst_pulsemixer_init),
	  (gst_pulsemixer_finalize), (gst_pulsemixer_set_property),
	  (gst_pulsemixer_get_property), (gst_pulsemixer_change_state):
	  * ext/pulse/pulsemixer.h:
	  * ext/pulse/pulsemixerctrl.c:
	  (gst_pulsemixer_ctrl_context_state_cb),
	  (gst_pulsemixer_ctrl_sink_info_cb),
	  (gst_pulsemixer_ctrl_source_info_cb),
	  (gst_pulsemixer_ctrl_subscribe_cb),
	  (gst_pulsemixer_ctrl_success_cb), (gst_pulsemixer_ctrl_open),
	  (gst_pulsemixer_ctrl_close), (gst_pulsemixer_ctrl_new),
	  (gst_pulsemixer_ctrl_free), (gst_pulsemixer_ctrl_list_tracks),
	  (gst_pulsemixer_ctrl_timeout_event), (restart_time_event),
	  (gst_pulsemixer_ctrl_set_volume), (gst_pulsemixer_ctrl_get_volume),
	  (gst_pulsemixer_ctrl_set_record), (gst_pulsemixer_ctrl_set_mute):
	  * ext/pulse/pulsemixerctrl.h:
	  * ext/pulse/pulsemixertrack.c: (gst_pulsemixer_track_class_init),
	  (gst_pulsemixer_track_init), (gst_pulsemixer_track_new):
	  * ext/pulse/pulsemixertrack.h:
	  * ext/pulse/pulseprobe.c: (gst_pulseprobe_context_state_cb),
	  (gst_pulseprobe_sink_info_cb), (gst_pulseprobe_source_info_cb),
	  (gst_pulseprobe_invalidate), (gst_pulseprobe_open),
	  (gst_pulseprobe_enumerate), (gst_pulseprobe_close),
	  (gst_pulseprobe_new), (gst_pulseprobe_free),
	  (gst_pulseprobe_get_properties), (gst_pulseprobe_needs_probe),
	  (gst_pulseprobe_probe_property), (gst_pulseprobe_get_values),
	  (gst_pulseprobe_set_server):
	  * ext/pulse/pulseprobe.h:
	  * ext/pulse/pulsesink.c: (gst_pulsesink_base_init),
	  (gst_pulsesink_class_init), (gst_pulsesink_init),
	  (gst_pulsesink_destroy_stream), (gst_pulsesink_destroy_context),
	  (gst_pulsesink_finalize), (gst_pulsesink_dispose),
	  (gst_pulsesink_set_property), (gst_pulsesink_get_property),
	  (gst_pulsesink_context_state_cb), (gst_pulsesink_stream_state_cb),
	  (gst_pulsesink_stream_request_cb),
	  (gst_pulsesink_stream_latency_update_cb), (gst_pulsesink_open),
	  (gst_pulsesink_close), (gst_pulsesink_prepare),
	  (gst_pulsesink_unprepare), (gst_pulsesink_write),
	  (gst_pulsesink_delay), (gst_pulsesink_success_cb),
	  (gst_pulsesink_reset), (gst_pulsesink_change_title),
	  (gst_pulsesink_event), (gst_pulsesink_get_type):
	  * ext/pulse/pulsesink.h:
	  * ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported),
	  (gst_pulsesrc_implements_interface_init),
	  (gst_pulsesrc_init_interfaces), (gst_pulsesrc_base_init),
	  (gst_pulsesrc_class_init), (gst_pulsesrc_init),
	  (gst_pulsesrc_destroy_stream), (gst_pulsesrc_destroy_context),
	  (gst_pulsesrc_finalize), (gst_pulsesrc_dispose),
	  (gst_pulsesrc_set_property), (gst_pulsesrc_get_property),
	  (gst_pulsesrc_context_state_cb), (gst_pulsesrc_stream_state_cb),
	  (gst_pulsesrc_stream_request_cb), (gst_pulsesrc_open),
	  (gst_pulsesrc_close), (gst_pulsesrc_prepare),
	  (gst_pulsesrc_unprepare), (gst_pulsesrc_read),
	  (gst_pulsesrc_delay), (gst_pulsesrc_change_state),
	  (gst_pulsesrc_get_type):
	  * ext/pulse/pulsesrc.h:
	  * ext/pulse/pulseutil.c: (gst_pulse_fill_sample_spec),
	  (gst_pulse_client_name), (gst_pulse_gst_to_channel_map):
	  * ext/pulse/pulseutil.h:
	  Add pulseaudio GStreamer element from gst-pulse. Development will
	  continue here instead of pulseaudio SVN. Fixes bug #400679.
	  Only changes over gst-pulse SVN are added copyright to the top of
	  files and coding style changes.

2008-06-09 20:02:05 +0000  Benjamin Kampmann <benjamin@fluendo.com>

	  ext/cdio/: Also extract album title and album genre from CD-TEXT if available (#537021).
	  Original commit message from CVS:
	  Patch by: Benjamin Kampmann  <benjamin at fluendo dot com>
	  * ext/cdio/gstcdio.c: (gst_cdio_get_cdtext),
	  (gst_cdio_add_cdtext_album_tags):
	  * ext/cdio/gstcdio.h:
	  * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open):
	  Also extract album title and album genre from CD-TEXT if
	  available (#537021).

2008-06-09 08:52:04 +0000  Sjoerd Simons <sjoerd@luon.net>

	  sys/v4l2/gstv4l2src.c: Improve negotiation a bit more by picking the smallest possible resolution that is larger than...
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_negotiate):
	  Improve negotiation a bit more by picking the smallest possible
	  resolution that is larger than the resolution specified in the
	  first caps entry of the peer caps. Fixes bug #536994.

2008-06-09 08:42:49 +0000  Bastien Nocera <hadess@hadess.net>

	  sys/v4l2/: Fix compilation with newer GIT kernels that deprecated
	  Original commit message from CVS:
	  Patch by: Bastien Nocera <hadess at hadess dot net>
	  * sys/v4l2/gstv4l2vidorient.c:
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  Fix compilation with newer GIT kernels that deprecated
	  V4L2_CID_HCENTER and V4L2_CID_VCENTER. Fixes bug #536317.

2008-06-07 18:48:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Require libcdio >= 0.76.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/cdio/gstcdio.c:
	  * ext/cdio/gstcdio.h:
	  * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open):
	  Require libcdio >= 0.76.

2008-06-05 11:07:17 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/: Properly implement duration and position queries in bytes format. We have to take the upstream reply...
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
	  (gst_deinterleave_src_query):
	  * gst/interleave/interleave.c: (gst_interleave_src_query_duration),
	  (gst_interleave_src_query):
	  Properly implement duration and position queries in bytes format. We
	  have to take the upstream reply and divide/multiply it by the number
	  of channels to get the correct result.

2008-06-05 09:45:00 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/avi/gstavidemux.c: Catch UNEXPECTED when downstream has reached end of segment in reverse mode.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Catch UNEXPECTED when downstream has reached end of
	  segment in reverse mode.

2008-06-04 18:08:35 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/avi/gstavidemux.c: Fix typo in comment
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Fix typo in comment

2008-06-04 18:03:24 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/avi/gstavidemux.c: Because we don't know the frame order we need to push till the next keyframe
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Because we don't know the frame order we need to push till
	  the next keyframe

2008-06-04 17:39:31 +0000  Sjoerd Simons <sjoerd@luon.net>

	  sys/v4l2/gstv4l2src.c: Provide a custom negotiation function to make sure to pick the highest possible framerate and ...
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	  (gst_v4l2src_fixate), (gst_v4l2src_negotiate):
	  Provide a custom negotiation function to make sure to pick the highest
	  possible framerate and resolution. Fixes bug #536646.

2008-06-04 16:49:26 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/avi/gstavidemux.c: Set EOS when going out of the segment in reverse playback
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Set EOS when going out of the segment in reverse playback

2008-06-04 15:19:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/Makefile.am: Add -Wno-attributes to CXXFLAGS to suppress warning caused by taglib headers (with gcc 4.3.1).
	  Original commit message from CVS:
	  * ext/taglib/Makefile.am::
	  Add -Wno-attributes to CXXFLAGS to suppress warning caused by
	  taglib headers (with gcc 4.3.1).

2008-06-04 11:59:18 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtsp/gstrtspsrc.c: Use the new gst_rtsp_connection_get_ip() to access the IP address of a GstRTSPConnection since...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
	  Use the new gst_rtsp_connection_get_ip() to access the IP address
	  of a GstRTSPConnection since it is a private member.

2008-06-04 10:42:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use new utility functions in libgsttag to process coverart (#512333).
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
	  * gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Use new utility functions in libgsttag to process coverart (#512333).

2008-06-04 08:54:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacdec.c: We actually support left/side, right/side and mid/side files. The conversion to normal, interl...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_write):
	  We actually support left/side, right/side and mid/side files. The
	  conversion to normal, interleaved stereo is done by libflac.

2008-06-04 07:36:07 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/ebml-write.c: Unref the write cache in finalize if it was set and add add "FIXME" to a comment that need...
	  Original commit message from CVS:
	  * gst/matroska/ebml-write.c: (gst_ebml_write_finalize),
	  (gst_ebml_write_set_cache):
	  Unref the write cache in finalize if it was set and add add "FIXME"
	  to a comment that needs it.

2008-06-04 06:48:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/interleave.*: Use an always increasing integer for the number in the name of the requested sink pads t...
	  Original commit message from CVS:
	  * gst/interleave/interleave.c: (gst_interleave_pad_get_type),
	  (gst_interleave_pad_get_property), (gst_interleave_pad_class_init),
	  (gst_interleave_request_new_pad), (gst_interleave_release_pad):
	  * gst/interleave/interleave.h:
	  Use an always increasing integer for the number in the name of the
	  requested sink pads to guarantuee a unique name. Add a "channel"
	  property to GstInterleavePad to make it possible for applications
	  to retrieve the channel number in the output for every pad.
	  Use g_type_register_static_simple() instead of
	  g_type_register_static() to save some relocations.

2008-06-03 14:35:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/interleave.c: Stop GstCollectPads before calling the parent's state change function when going from PA...
	  Original commit message from CVS:
	  * gst/interleave/interleave.c: (gst_interleave_pad_get_type),
	  (gst_interleave_change_state):
	  Stop GstCollectPads before calling the parent's state change function
	  when going from PAUSED to READY as we otherwise deadlock.
	  Fixes bug #536258.

2008-06-03 09:03:19 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/interleave.c: Use new gst_audio_check_channel_positions() function and register the GstInterleavePad t...
	  Original commit message from CVS:
	  * gst/interleave/interleave.c:
	  (gst_interleave_check_channel_positions),
	  (gst_interleave_set_channel_positions),
	  (gst_interleave_class_init):
	  Use new gst_audio_check_channel_positions() function and register
	  the GstInterleavePad type from a threadsafe context.

2008-06-02 16:10:00 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/avi/gstavidemux.*: Implement reverse playback. Fixes #535300.
	  Original commit message from CVS:
	  Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_index_next),
	  (gst_avi_demux_index_prev), (gst_avi_demux_index_entry_for_time),
	  (gst_avi_demux_do_seek), (gst_avi_demux_handle_seek),
	  (gst_avi_demux_process_next_entry):
	  * gst/avi/gstavidemux.h:
	  Implement reverse playback. Fixes #535300.
	  Small cleanups.

2008-06-02 12:42:14 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/interleave.*: Allow setting channel positions via a property and allow using the channel positions on ...
	  Original commit message from CVS:
	  * gst/interleave/interleave.c: (gst_interleave_pad_get_type),
	  (gst_interleave_finalize), (gst_audio_check_channel_positions),
	  (gst_interleave_set_channel_positions),
	  (gst_interleave_class_init), (gst_interleave_init),
	  (gst_interleave_set_property), (gst_interleave_get_property),
	  (gst_interleave_request_new_pad), (gst_interleave_release_pad),
	  (gst_interleave_sink_setcaps), (gst_interleave_src_query_duration),
	  (gst_interleave_src_query_latency), (gst_interleave_collected):
	  * gst/interleave/interleave.h:
	  Allow setting channel positions via a property and allow using the
	  channel positions on the input as the channel positions of the output.
	  Fix some broken logic and memory leaks.
	  * tests/check/Makefile.am:
	  * tests/check/elements/interleave.c: (src_handoff_float32),
	  (sink_handoff_float32), (GST_START_TEST), (interleave_suite):
	  Add unit tests for checking correct handling of channel positions.

2008-06-02 12:22:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/videomixer/videomixer.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_query_duration),
	  (gst_videomixer_query_latency):
	  When using gst_element_iterate_pads() one has to unref every pad
	  after usage.

2008-05-31 16:53:23 +0000  Bastien Nocera <hadess@hadess.net>

	  gst/qtdemux/: Improve meta-data handling, add 'comment', 'description' and 'copyright' tag handling.
	  Original commit message from CVS:
	  Patch by: Bastien Nocera <hadess at hadess dot net>
	  * gst/qtdemux/qtdemux.c: (qtdemux_tag_add_str),
	  (qtdemux_parse_udta):
	  * gst/qtdemux/qtdemux_fourcc.h:
	  Improve meta-data handling, add 'comment', 'description' and
	  'copyright' tag handling.
	  Fixes #535935

2008-05-31 15:30:41 +0000  Julien Moutte <julien@moutte.net>

	  gst/qtdemux/qtdemux.c: Make sure we we don't clip the segment's stop using the main segment duration as that could cr...
	  Original commit message from CVS:
	  2008-05-31  Julien Moutte  <julien@fluendo.com>
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_find_keyframe),
	  (gst_qtdemux_find_segment), (gst_qtdemux_perform_seek),
	  (gst_qtdemux_seek_to_previous_keyframe),
	  (gst_qtdemux_activate_segment), (gst_qtdemux_loop): Make sure we
	  we don't clip the segment's stop using the main segment duration
	  as
	  that could crop quite some video frames. Make reverse playback
	  support
	  more robust and support edit lists. Support seeking to the last
	  frame,
	  and fix reverse looping playback. Add some debugging.
	  * win32/common/config.h: Updated.

2008-05-31 08:37:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.c: Don't clip float/double samples, correctly unset passthrough mode and use better rou...
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_transform_ip):
	  Don't clip float/double samples, correctly unset passthrough mode
	  and use better rounding for integer samples.

2008-05-30 11:03:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.*: Update the filter coefficients only when needed in the transform_ip function and cor...
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_init),
	  (setup_filter), (set_passthrough), (update_coefficients),
	  (gst_iir_equalizer_compute_frequencies),
	  (gst_iir_equalizer_transform_ip):
	  * gst/equalizer/gstiirequalizer.h:
	  Update the filter coefficients only when needed in the transform_ip
	  function and correctly set the element into passthrough mode if the
	  gain of all bands is 0.

2008-05-29 11:30:16 +0000  Sebastian Keller <sebastian-keller@gmx.de>

	  gst/alpha/gstalpha.c: Try to skip pixels or areas that are too dark or too bright for us to do meaningfull color dete...
	  Original commit message from CVS:
	  Based on patch by: Sebastian Keller <sebastian-keller at gmx dot de>
	  * gst/alpha/gstalpha.c: (gst_alpha_class_init), (gst_alpha_init),
	  (gst_alpha_set_property), (gst_alpha_get_property),
	  (gst_alpha_chroma_key_ayuv), (gst_alpha_chromakey_row_i420):
	  Try to skip pixels or areas that are too dark or too bright for us to do
	  meaningfull color detection.
	  Added properties to control the sensitivity to light and darkness.
	  Added some small cleanups. Fixes #512345.

2008-05-28 20:01:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Ignore some more generated things
	  Original commit message from CVS:
	  * docs/plugins/.cvsignore:
	  * tests/check/elements/.cvsignore:
	  Ignore some more generated things
	  * tests/check/Makefile.am:
	  Ignore OSS elements in the state changes test too.

2008-05-28 16:22:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Add SMPTE effect elements to docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Add SMPTE effect elements to docs.

2008-05-28 14:31:05 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Document whats first shown on the fdo plugin docs page :)
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * ext/raw1394/gstdv1394src.c:
	  Document whats first shown on the fdo plugin docs page :)

2008-05-28 14:07:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Rename audiovoice to audiokaraoke and add it to the docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofx.c:
	  * gst/audiofx/audiokaraoke.c:
	  * gst/audiofx/audiokaraoke.h:
	  * gst/audiofx/audiovoice.c:
	  * gst/audiofx/audiovoice.h:
	  Rename audiovoice to audiokaraoke and add it to the docs.

2008-05-28 13:28:20 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Document aasink and cacasink.
	  Original commit message from CVS:
	  * REQUIREMENTS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/gst-plugins-good-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gamma.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-goom2k1.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-monoscope.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multifile.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-soup.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-video4linux2.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  * ext/aalib/gstaasink.c:
	  * ext/libcaca/gstcacasink.c:
	  Document aasink and cacasink.

2008-05-28 08:36:44 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/videomixer/videomixer.*: duration and latency queries.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_reset),
	  (gst_videomixer_init), (gst_videomixer_query_duration),
	  (gst_videomixer_query_latency), (gst_videomixer_query),
	  (gst_videomixer_blend_buffers):
	  * gst/videomixer/videomixer.h:
	  Implement position (in time), duration and latency queries.

2008-05-28 08:14:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/interleave.c: Implement latency query.
	  Original commit message from CVS:
	  * gst/interleave/interleave.c: (gst_interleave_src_query_duration),
	  (gst_interleave_src_query_latency), (gst_interleave_src_query):
	  Implement latency query.

2008-05-27 17:55:30 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/videomixer/videomixer.*: Implement proper seek/newsegment handling.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_reset),
	  (gst_videomixer_init), (gst_videomixer_request_new_pad),
	  (gst_videomixer_fill_queues), (forward_event_func),
	  (forward_event), (gst_videomixer_src_event),
	  (gst_videomixer_sink_event):
	  * gst/videomixer/videomixer.h:
	  Implement proper seek/newsegment handling.
	  Based on adder's implementation.
	  Fixes #535121

2008-05-26 16:25:15 +0000  j^ <j@oil21.org>

	  gst/qtdemux/qtdemux.c: Add caps for DVCPRO50 and DVCPRO HD PAL/NTSC. See #526481.
	  Original commit message from CVS:
	  Patch by: j^ <j at oil21 dot org>
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add caps for DVCPRO50 and DVCPRO HD PAL/NTSC. See #526481.

2008-05-26 15:51:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audiofx/: Add simple voice removal element. Yay karaoke.
	  Original commit message from CVS:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofx.c: (plugin_init):
	  * gst/audiofx/audiovoice.c: (gst_audio_voice_base_init),
	  (gst_audio_voice_class_init), (gst_audio_voice_init),
	  (update_filter), (gst_audio_voice_set_property),
	  (gst_audio_voice_get_property), (gst_audio_voice_setup),
	  (gst_audio_voice_transform_int), (gst_audio_voice_transform_float),
	  (gst_audio_voice_transform_ip):
	  * gst/audiofx/audiovoice.h:
	  Add simple voice removal element. Yay karaoke.

2008-05-26 15:39:26 +0000  William M. Brack <wbrack@mmm.com.hk>

	  sys/v4l2/v4l2src_calls.c: Fix potential caps leak.
	  Original commit message from CVS:
	  Patch by: William M. Brack <wbrack at mmm dot com dot hk>
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format):
	  Fix potential caps leak.
	  If we can't get the framerate with an ioctl, try to get it with the
	  current norm. Fixes #520092.

2008-05-26 15:14:55 +0000  William M. Brack <wbrack@mmm.com.hk>

	  sys/v4l2/v4l2src_calls.c: If we fail to get the frame intervals, simply don't touch the framerates on the template ca...
	  Original commit message from CVS:
	  Patch by: William M. Brack <wbrack at mmm dot com dot hk>
	  * sys/v4l2/v4l2src_calls.c:
	  (gst_v4l2src_probe_caps_for_format_and_size):
	  If we fail to get the frame intervals, simply don't touch the framerates
	  on the template caps instead of discarding the format. See #520092.

2008-05-26 14:52:51 +0000  William M. Brack <wbrack@mmm.com.hk>

	  sys/v4l2/gstv4l2src.c: Add NV12, NV21 and bayer support. See #520092.
	  Original commit message from CVS:
	  Patch by: William M. Brack <wbrack at mmm dot com dot hk>
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure),
	  (gst_v4l2_get_caps_info):
	  Add NV12, NV21 and bayer support. See #520092.

2008-05-26 13:51:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Unbreak segment activation again. Fixes #531672.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_find_segment),
	  (gst_qtdemux_activate_segment):
	  Unbreak segment activation again. Fixes #531672.

2008-05-26 10:28:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/deinterleave.c: Add another example launch line.
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c:
	  Add another example launch line.
	  * gst/interleave/interleave.c: (interleave_24),
	  (gst_interleave_finalize), (gst_interleave_base_init),
	  (gst_interleave_class_init), (gst_interleave_init),
	  (gst_interleave_request_new_pad), (gst_interleave_release_pad),
	  (gst_interleave_change_state), (__remove_channels),
	  (__set_channels), (gst_interleave_sink_getcaps),
	  (gst_interleave_set_process_function),
	  (gst_interleave_sink_setcaps), (gst_interleave_sink_event),
	  (gst_interleave_src_query_duration), (gst_interleave_src_query),
	  (forward_event_func), (forward_event), (gst_interleave_src_event),
	  (gst_interleave_collected):
	  * gst/interleave/interleave.h:
	  Major rewrite of interleave using GstCollectpads. This new version
	  also supports almost all raw audio formats and has better caps
	  negotiation. Fixes bug #506594.
	  Also update docs and add some more examples.
	  * tests/check/elements/interleave.c: (interleave_chain_func),
	  (GST_START_TEST), (src_handoff_float32), (sink_handoff_float32),
	  (interleave_suite):
	  Add some more extensive unit tests for interleave.

2008-05-26 09:57:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Don't use _gst_pad().
	  Original commit message from CVS:
	  * examples/switch/switcher.c: (switch_timer):
	  * gst/replaygain/gstrgvolume.c: (gst_rg_volume_init):
	  * gst/rtpmanager/gstrtpclient.c: (create_stream):
	  * gst/sdp/gstsdpdemux.c: (gst_sdp_demux_stream_configure_udp),
	  (gst_sdp_demux_stream_configure_udp_sink):
	  * tests/check/elements/deinterleave.c: (GST_START_TEST),
	  (pad_added_setup_data_check_float32_8ch_cb):
	  * tests/check/elements/rganalysis.c: (send_eos_event),
	  (send_tag_event):
	  Don't use _gst_pad().

2008-05-25 16:09:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/: Set the channel layout when decoding FLAC files with more than 2 channels as defined by the FLAC spec. Fix...
	  Original commit message from CVS:
	  * ext/flac/Makefile.am:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_write):
	  Set the channel layout when decoding FLAC files with more than 2
	  channels as defined by the FLAC spec. Fixes bug #534570.
	  Also don't try to decode left/side, right/side and mid/side files
	  as we don't support this at all.

2008-05-24 12:55:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: We need -base CVS (rtsp).
	  Original commit message from CVS:
	  * configure.ac:
	  We need -base CVS (rtsp).

2008-05-22 19:47:53 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  docs/plugins/: Add interleave/deinterleave to the docs and while at that run make update in docs/plugins.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.prerequisites:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-alsaspdif.xml:
	  * docs/plugins/inspect/plugin-amrwb.xml:
	  * docs/plugins/inspect/plugin-app.xml:
	  * docs/plugins/inspect/plugin-bayer.xml:
	  * docs/plugins/inspect/plugin-bz2.xml:
	  * docs/plugins/inspect/plugin-cdaudio.xml:
	  * docs/plugins/inspect/plugin-cdxaparse.xml:
	  * docs/plugins/inspect/plugin-dfbvideosink.xml:
	  * docs/plugins/inspect/plugin-dtsdec.xml:
	  * docs/plugins/inspect/plugin-dvb.xml:
	  * docs/plugins/inspect/plugin-dvdspu.xml:
	  * docs/plugins/inspect/plugin-faac.xml:
	  * docs/plugins/inspect/plugin-faad.xml:
	  * docs/plugins/inspect/plugin-fbdevsink.xml:
	  * docs/plugins/inspect/plugin-festival.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-flvdemux.xml:
	  * docs/plugins/inspect/plugin-freeze.xml:
	  * docs/plugins/inspect/plugin-gsm.xml:
	  * docs/plugins/inspect/plugin-gstrtpmanager.xml:
	  * docs/plugins/inspect/plugin-h264parse.xml:
	  * docs/plugins/inspect/plugin-interleave.xml:
	  * docs/plugins/inspect/plugin-jack.xml:
	  * docs/plugins/inspect/plugin-ladspa.xml:
	  * docs/plugins/inspect/plugin-metadata.xml:
	  * docs/plugins/inspect/plugin-mms.xml:
	  * docs/plugins/inspect/plugin-modplug.xml:
	  * docs/plugins/inspect/plugin-mpeg2enc.xml:
	  * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
	  * docs/plugins/inspect/plugin-mpegtsparse.xml:
	  * docs/plugins/inspect/plugin-mpegvideoparse.xml:
	  * docs/plugins/inspect/plugin-musepack.xml:
	  * docs/plugins/inspect/plugin-musicbrainz.xml:
	  * docs/plugins/inspect/plugin-mve.xml:
	  * docs/plugins/inspect/plugin-nas.xml:
	  * docs/plugins/inspect/plugin-neon.xml:
	  * docs/plugins/inspect/plugin-nsfdec.xml:
	  * docs/plugins/inspect/plugin-nuvdemux.xml:
	  * docs/plugins/inspect/plugin-rawparse.xml:
	  * docs/plugins/inspect/plugin-real.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-rfbsrc.xml:
	  * docs/plugins/inspect/plugin-sdl.xml:
	  * docs/plugins/inspect/plugin-sdp.xml:
	  * docs/plugins/inspect/plugin-selector.xml:
	  * docs/plugins/inspect/plugin-sndfile.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-spcdec.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-speexresample.xml:
	  * docs/plugins/inspect/plugin-stereo.xml:
	  * docs/plugins/inspect/plugin-tta.xml:
	  * docs/plugins/inspect/plugin-vcdsrc.xml:
	  * docs/plugins/inspect/plugin-videosignal.xml:
	  * docs/plugins/inspect/plugin-vmnc.xml:
	  * docs/plugins/inspect/plugin-wildmidi.xml:
	  * docs/plugins/inspect/plugin-x264.xml:
	  * docs/plugins/inspect/plugin-xvid.xml:
	  * docs/plugins/inspect/plugin-y4menc.xml:
	  Add interleave/deinterleave to the docs and while at that
	  run make update in docs/plugins.
	  * gst/interleave/deinterleave.c:
	  Add a parapraph about using a queue and audioconvert after the source
	  pads to the docs.

2008-05-22 18:55:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/deinterleave.*: Don't set a getcaps() function on the src pads as it's not required and the default ge...
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
	  (gst_deinterleave_class_init), (gst_deinterleave_init),
	  (gst_deinterleave_add_new_pads), (gst_deinterleave_sink_getcaps):
	  * gst/interleave/deinterleave.h:
	  Don't set a getcaps() function on the src pads as it's not required
	  and the default getcaps() function returns the correct results for
	  our src pads.
	  Complete documentation and add myself to the authors of the element.

2008-05-22 14:49:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/udp/Makefile.am: Add -D_GNU_SOURCE to CFLAGS so we get things like EAI_ADDRFAMILY when including netdb.h when bui...
	  Original commit message from CVS:
	  * gst/udp/Makefile.am:
	  Add -D_GNU_SOURCE to CFLAGS so we get things like EAI_ADDRFAMILY
	  when including netdb.h when building against glibc >= 2.8.

2008-05-22 11:19:03 +0000  Julien Moutte <julien@moutte.net>

	  gst/smpte/gstsmptealpha.c: Fix debug statement arguments.
	  Original commit message from CVS:
	  2008-05-22  Julien Moutte  <julien@fluendo.com>
	  * gst/smpte/gstsmptealpha.c: (gst_smpte_alpha_setcaps): Fix
	  debug statement arguments.
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_setup_qos_dscp):
	  * gst/udp/gstudpnetutils.c: (gst_udp_join_group),
	  (gst_udp_leave_group): Fix IP and IPV6 options to make it work
	  on more platforms.

2008-05-21 17:51:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/elements/: Don't use gst_element_get_pad(), it's a bad, bad method.
	  Original commit message from CVS:
	  * tests/check/elements/avimux.c: (setup_src_pad),
	  (teardown_src_pad):
	  * tests/check/elements/icydemux.c: (icydemux_found_pad),
	  (GST_START_TEST):
	  * tests/check/elements/matroskamux.c: (setup_src_pad),
	  (teardown_src_pad), (setup_sink_pad), (teardown_sink_pad):
	  * tests/check/elements/videocrop.c: (video_crop_get_test_caps),
	  (GST_START_TEST):
	  * tests/check/elements/wavpackparse.c: (wavpackparse_found_pad),
	  (setup_wavpackparse), (cleanup_wavpackparse):
	  Don't use gst_element_get_pad(), it's a bad, bad method.

2008-05-21 17:39:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Don't use gst_element_get_pad(), it's a bad method.
	  Original commit message from CVS:
	  * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
	  (do_toggle_element):
	  * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
	  (do_toggle_element):
	  * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
	  (do_toggle_element):
	  * ext/gconf/gstswitchsink.c: (gst_switch_commit_new_kid):
	  * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_reset),
	  (do_toggle_element):
	  * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_reset),
	  (do_toggle_element):
	  * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset),
	  (gst_auto_audio_sink_detect):
	  * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
	  (gst_auto_video_sink_detect):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_udp),
	  (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_skip_lws),
	  (gst_rtspsrc_unskip_lws), (gst_rtspsrc_skip_commas),
	  (gst_rtspsrc_skip_item), (gst_rtsp_decode_quoted_string),
	  (gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr):
	  * tests/icles/videocrop-test.c: (test_with_caps),
	  (video_crop_get_test_caps):
	  Don't use gst_element_get_pad(), it's a bad method.

2008-05-21 17:35:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/: Joining a multicast group and setting the loop/ttl properties are totally unrelated tasks are must be separ...
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send),
	  (gst_multiudpsink_add_internal):
	  * gst/udp/gstudpnetutils.c: (gst_udp_set_loop_ttl),
	  (gst_udp_join_group):
	  * gst/udp/gstudpnetutils.h:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  Joining a multicast group and setting the loop/ttl properties are
	  totally unrelated tasks are must be separated.

2008-05-21 14:09:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: Also support alaw/mulaw.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Also support alaw/mulaw.

2008-05-21 13:47:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.*: Add a fixme for the auto-multicast property.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	  (gst_multiudpsink_setup_qos_dscp), (gst_multiudpsink_add_internal):
	  * gst/udp/gstmultiudpsink.h:
	  Add a fixme for the auto-multicast property.
	  Fix some confusing debug messages.
	  Disable setting a qos value by default.

2008-05-21 11:38:17 +0000  Gustaf Räntilä <g.rantila@gmail.com>

	  gst/udp/gstmultiudpsink.c: Ignore EPERM errors from sendto. Fixes #533619.
	  Original commit message from CVS:
	  Patch by: Gustaf Räntilä <g dot rantila at gmail dot com>
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
	  Ignore EPERM errors from sendto. Fixes #533619.

2008-05-21 10:51:52 +0000  Henrik Eriksson <henriken@axis.com>

	  gst/udp/gstmultiudpsink.*: Add qos-dscp property to manage the Quality of service.
	  Original commit message from CVS:
	  Patch by: Henrik Eriksson <henriken at axis dot com>
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	  (gst_multiudpsink_init), (gst_multiudpsink_setup_qos_dscp),
	  (gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
	  (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal):
	  * gst/udp/gstmultiudpsink.h:
	  Add qos-dscp property to manage the Quality of service.

2008-05-21 10:09:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtptheoradepay.c: Improve debugging of the ident.
	  Original commit message from CVS:
	  * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_process):
	  Improve debugging of the ident.

2008-05-21 09:56:02 +0000  Bruno Santos <brunof@ua.pt>

	  gst/udp/gstudpnetutils.*: Provide a bunch of helper methods to deal with IPv4 and IPv6 transparently.
	  Original commit message from CVS:
	  Patch by: Bruno Santos <brunof at ua dot pt>
	  * gst/udp/gstudpnetutils.c: (gst_udp_get_addr),
	  (gst_udp_join_group), (gst_udp_leave_group),
	  (gst_udp_is_multicast):
	  * gst/udp/gstudpnetutils.h:
	  Provide a bunch of helper methods to deal with IPv4 and IPv6
	  transparently.
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	  (gst_multiudpsink_init), (gst_multiudpsink_set_property),
	  (gst_multiudpsink_get_property), (join_multicast),
	  (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
	  (gst_multiudpsink_remove):
	  * gst/udp/gstmultiudpsink.h:
	  Add multicast TTL and loopback properties.
	  Use the helper methods to implement ip4 and ip6.
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create), (gst_udpsrc_start):
	  * gst/udp/gstudpsrc.h:
	  Use the helper methods to implement ip4 and ip6.
	  Fixes #515962.

2008-05-21 09:38:48 +0000  Patrick Radizi <patrick.radizi@axis.com>

	  gst/multipart/multipartdemux.*: Don't blindly copy the mime-type as the caps name because they not always map directl...
	  Original commit message from CVS:
	  Patch by: Patrick Radizi <patrick dot radizi at axis dot com>
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_class_init),
	  (gst_multipart_demux_get_gstname),
	  (gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain):
	  * gst/multipart/multipartdemux.h:
	  Don't blindly copy the mime-type as the caps name because they not
	  always map directly. Instead use a hashtable with common mappings.
	  Fixes #533287.

2008-05-20 17:27:35 +0000  Michael Meeks <mmeeks@ximian.org>

	  ext/esd/esdsink.c: When we post an error, we must return -1 to let the parent know that we cannot write the segment e...
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_write):
	  When we post an error, we must return -1 to let the parent know that we
	  cannot write the segment else it will loop and continue to call us again
	  forever. Patch by Michael Meeks.

2008-05-20 14:24:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/videomixer/videomixer.c: Add missing incudes.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c:
	  Add missing incudes.

2008-05-20 13:57:44 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtp/gstrtph264pay.*: Correct a typo (sinle -> single).
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264pay.c: (gst_h264_scan_mode_get_type),
	  (gst_rtp_h264_pay_handle_buffer):
	  * gst/rtp/gstrtph264pay.h:
	  Correct a typo (sinle -> single).

2008-05-20 11:33:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph264depay.*: Add experimental support for outputting quicktime-like AVC output in addition to the exist...
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	  (gst_rtp_h264_depay_init), (gst_rtp_h264_depay_set_property),
	  (gst_rtp_h264_depay_get_property), (gst_rtp_h264_depay_setcaps),
	  (gst_rtp_h264_depay_process):
	  * gst/rtp/gstrtph264depay.h:
	  Add experimental support for outputting quicktime-like AVC output in
	  addition to the existing bytestream output.
	  * gst/rtp/gstrtph264pay.c: (gst_h264_scan_mode_get_type),
	  (gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
	  (gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_payload_nal),
	  (gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
	  (gst_rtp_h264_pay_get_property):
	  * gst/rtp/gstrtph264pay.h:
	  Make the parsing mode configurable, for some inputs we don't need to
	  scan every byte for start codes.
	  Only set the marker bit on ACCESS units.

2008-05-20 10:47:10 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.c: Use a bigger type in integer mode for the intermediate results to prevent overflows....
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  Use a bigger type in integer mode for the intermediate results to
	  prevent overflows. This fixes the crippled sound when using the
	  equalizer in integer mode. Fixes bug #510865.

2008-05-20 10:42:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/videomixer/videomixer.*: Instead of a random number for the request pad id's, use a counter.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c:
	  * gst/videomixer/videomixer.h:
	  Instead of a random number for the request pad id's,
	  use a counter.
	  Register the videomixerpad class from the element's class_init
	  where it's safer, and allows the docs generator to scan it.

2008-05-20 09:29:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/smpte/: Add new plugin that adds the SMPTE transition in the alpha channel of
	  Original commit message from CVS:
	  * gst/smpte/Makefile.am:
	  * gst/smpte/gstsmpte.c: (gst_smpte_plugin_init):
	  * gst/smpte/gstsmpte.h:
	  * gst/smpte/gstsmptealpha.c:
	  (gst_smpte_alpha_transition_type_get_type),
	  (gst_smpte_alpha_get_type), (gst_smpte_alpha_base_init),
	  (gst_smpte_alpha_class_init), (gst_smpte_alpha_update_mask),
	  (gst_smpte_alpha_setcaps), (gst_smpte_alpha_get_unit_size),
	  (gst_smpte_alpha_init), (gst_smpte_alpha_finalize),
	  (gst_smpte_alpha_do_ayuv), (gst_smpte_alpha_do_i420),
	  (gst_smpte_alpha_transform), (gst_smpte_alpha_set_property),
	  (gst_smpte_alpha_get_property), (gst_smpte_alpha_plugin_init):
	  * gst/smpte/gstsmptealpha.h:
	  * gst/smpte/plugin.c: (plugin_init):
	  Add new plugin that adds the SMPTE transition in the alpha channel of
	  I420 and AYUV frames so that they can be blended with videomixer later
	  on. Uses all niceties such as using base transform for efficient alloc
	  and negotiation. It currently requires GstController to control the
	  position in the transition effect.

2008-05-19 21:05:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Try using thaytans new mechanism to get extra classes into plugin docs. Aparently works for the Eq. For VideoMixer th...
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/gst-plugins-good-plugins.types:
	  * gst/videomixer/videomixer.c:
	  Try using thaytans new mechanism to get extra classes into plugin
	  docs. Aparently works for the Eq. For VideoMixer the GObject stuff is
	  missing still.

2008-05-19 12:32:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/deinterleave.c: Set keep-positions property to TRUE for the 8 channel test to ensure that the or...
	  Original commit message from CVS:
	  * tests/check/elements/deinterleave.c: (GST_START_TEST):
	  Set keep-positions property to TRUE for the 8 channel test to ensure
	  that the original channel position is set on the output.

2008-05-19 07:46:05 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/deinterleave.*: Add a property to select whether channel positions should be kept on the mono output b...
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c: (gst_deinterleave_class_init),
	  (gst_deinterleave_init), (gst_deinterleave_add_new_pads),
	  (gst_deinterleave_set_pads_caps), (gst_deinterleave_set_property),
	  (gst_deinterleave_get_property):
	  * gst/interleave/deinterleave.h:
	  Add a property to select whether channel positions should be kept on
	  the mono output buffers or should be dropped.

2008-05-18 19:27:59 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/gstavimux.c: Set proper rate in avi stream header for PCM audio, and also do some more sanity checks on caps ...
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_audsink_set_caps):
	  Set proper rate in avi stream header for PCM audio, and also do some
	  more sanity checks on caps in this case.  Fixes #511489.

2008-05-17 19:39:53 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/deinterleave.*: Queue events until src pads were added and they can be sent. Otherwise downstream will...
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c: (gst_deinterleave_finalize),
	  (gst_deinterleave_init), (gst_deinterleave_sink_event),
	  (gst_deinterleave_process), (gst_deinterleave_sink_activate_push):
	  * gst/interleave/deinterleave.h:
	  Queue events until src pads were added and they can be sent. Otherwise
	  downstream will never get the first newsegment event.

2008-05-17 14:05:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/deinterleave.c: Always set the channel positions when gst_audio_get_channel_positions() returns someth...
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps),
	  (gst_deinterleave_getcaps):
	  Always set the channel positions when gst_audio_get_channel_positions()
	  returns something, even if they're not set in the caps. This makes
	  sure that the output channels can be interleaved again correctly
	  in the mono/stereo cases too.
	  Don't ask for the peercaps of the current pad in getcaps() as this
	  might call getcaps() again and deadlock.

2008-05-17 10:38:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  sys/v4l2/gstv4l2src.c: Don't include the gstv4l2xoverlay.h header as the XOverlay support isn't implemented at all ye...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c:
	  Don't include the gstv4l2xoverlay.h header as the XOverlay support
	  isn't implemented at all yet and this requires X headers to be
	  installed. Fixes bug #533264.

2008-05-16 21:56:24 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/: Add support for all raw audio formats and provide better negotiation if the caps are changing.
	  Original commit message from CVS:
	  * gst/interleave/Makefile.am:
	  * gst/interleave/deinterleave.c: (deinterleave_24),
	  (gst_deinterleave_finalize), (gst_deinterleave_base_init),
	  (gst_deinterleave_class_init), (gst_deinterleave_init),
	  (gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps),
	  (gst_deinterleave_set_process_function),
	  (gst_deinterleave_sink_setcaps), (__remove_channels),
	  (__set_channels), (gst_deinterleave_getcaps),
	  (gst_deinterleave_process), (gst_deinterleave_chain),
	  (gst_deinterleave_sink_activate_push):
	  * gst/interleave/deinterleave.h:
	  Add support for all raw audio formats and provide better negotiation
	  if the caps are changing.
	  Don't allow changes of the channel positions and set the position of
	  the corresponding channel on the src pad caps.
	  General cleanup and smaller bugfixes.
	  * tests/check/elements/deinterleave.c: (float_buffer_check_probe):
	  Check the channel positions on the output buffer caps.

2008-05-16 17:50:20 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Fix some compiler warnings.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackstreamreader.c:
	  * tests/examples/spectrum/demo-audiotest.c:
	  * tests/examples/spectrum/demo-osssrc.c:
	  Fix some compiler warnings.

2008-05-14 18:28:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph264depay.c: Small comment added.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
	  Small comment added.
	  * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
	  (gst_rtp_h264_pay_decode_nal), (gst_rtp_h264_pay_parse_sps_pps),
	  (gst_rtp_h264_pay_payload_nal), (gst_rtp_h264_pay_handle_buffer):
	  Debug string cleanups (remove trailing \n)
	  Refactor and clean up the payloader a bit and make sure that we only
	  put one NAL unit in an RTP packet even if the input buffer contains
	  multiple NAL units.
	  Add suport for AVC format input.

2008-05-14 17:58:50 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtp/gstrtph264pay.*: Make it possible to specify profile-level-id and sprop-parameter-sets using properties in ca...
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
	  (gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_handle_buffer),
	  (gst_rtp_h264_pay_set_property), (gst_rtp_h264_pay_get_property):
	  * gst/rtp/gstrtph264pay.h:
	  Make it possible to specify profile-level-id and sprop-parameter-sets
	  using properties in case they are not available in-stream.

2008-05-14 14:19:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, since it causes weird invalid free errors in ...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Add deinterleave unit test to VALGRIND_TO_FIX, since it causes
	  weird invalid free errors in valgrind/libc after _exit for some
	  reason.
	  * tests/check/elements/deinterleave.c: (pads_created),
	  (set_channel_positions), (src_handoff_float32_8ch),
	  (float_buffer_check_probe),
	  (pad_added_setup_data_check_float32_8ch_cb),
	  (make_fake_src_8chans_float32), (GST_START_TEST),
	  (deinterleave_suite):
	  Add some more deinterleave unit test bits I had locally.

2008-05-14 12:52:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: Remove ladspa fro plugin-docs, its in gst-plugins-bad.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-ladspa.xml:
	  Remove ladspa fro plugin-docs, its in gst-plugins-bad.

2008-05-14 07:32:44 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/interleave/: Split definitions into separate header files for better documentation generation.
	  Original commit message from CVS:
	  * gst/interleave/Makefile.am:
	  * gst/interleave/deinterleave.h:
	  * gst/interleave/interleave.h:
	  * gst/interleave/plugin.h:
	  Split definitions into separate header files for better documentation
	  generation.
	  * gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
	  (gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps),
	  (gst_deinterleave_process):
	  Don't use alloca, allow caps changes as long as the number of channels
	  does not change, don't use g_warning, return NOT_NEGOTIATED as early
	  as possible and some other cleanup.
	  * gst/interleave/interleave.c: (gst_interleave_base_init),
	  (gst_interleave_class_init):
	  Do some random cleanup.
	  * tests/check/Makefile.am:
	  * tests/check/elements/deinterleave.c: (GST_START_TEST),
	  (deinterleave_chain_func), (deinterleave_pad_added),
	  (deinterleave_suite):
	  Add unit tests for the deinterleave element.

2008-05-13 20:25:20 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/gstavimux.c: Send an initial BYTE segment to inform downstream of later seeking, and to forego sync attempts.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_start_file):
	  Send an initial BYTE segment to inform downstream of later seeking,
	  and to forego sync attempts.

2008-05-13 08:59:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpg729depay.c: Fix wrong caps string.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_setcaps):
	  Fix wrong caps string.

2008-05-13 08:35:55 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/: Added G729 pay and depayloaders. Fixes #532409.
	  Original commit message from CVS:
	  Based on patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_base_init),
	  (gst_rtp_g729_depay_class_init), (gst_rtp_g729_depay_init),
	  (gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process),
	  (gst_rtp_g729_depay_plugin_init):
	  * gst/rtp/gstrtpg729depay.h:
	  * gst/rtp/gstrtpg729pay.c: (gst_rtpg729pay_base_init),
	  (gst_rtpg729pay_class_init), (gst_rtpg729pay_init),
	  (gst_rtpg729pay_setcaps), (gst_rtp_g729_pay_plugin_init):
	  * gst/rtp/gstrtpg729pay.h:
	  Added G729 pay and depayloaders. Fixes #532409.

2008-05-13 08:21:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/speex/gstspeexdec.c: Fix the calculation of the duration of the concealment packets.
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_sink_event):
	  Fix the calculation of the duration of the concealment packets.

2008-05-12 18:27:24 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/: Add DV pay and depayloaders. Fixes #532423.
	  Original commit message from CVS:
	  Based on patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_base_init),
	  (gst_rtp_dv_depay_class_init), (gst_rtp_dv_depay_init),
	  (parse_encode), (gst_rtp_dv_depay_setcaps),
	  (calculate_difblock_location), (gst_rtp_dv_depay_process),
	  (gst_rtp_dv_depay_reset), (gst_rtp_dv_depay_change_state),
	  (gst_rtp_dv_depay_plugin_init):
	  * gst/rtp/gstrtpdvdepay.h:
	  * gst/rtp/gstrtpdvpay.c: (gst_dv_pay_mode_get_type),
	  (gst_rtp_dv_pay_base_init), (gst_rtp_dv_pay_class_init),
	  (gst_rtp_dv_pay_init), (gst_dv_pay_set_property),
	  (gst_dv_pay_get_property), (gst_rtp_dv_pay_setcaps),
	  (gst_dv_pay_negotiate), (include_dif),
	  (gst_rtp_dv_pay_handle_buffer), (gst_rtp_dv_pay_plugin_init):
	  * gst/rtp/gstrtpdvpay.h:
	  Add DV pay and depayloaders. Fixes #532423.

2008-05-12 16:35:39 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/matroska/matroska-demux.c: Convert subtitle palette info in VobSub private data from VobSub's (buggy) RGB to YUV.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_push_dvd_clut_change_event):
	  Convert subtitle palette info in VobSub private data from VobSub's
	  (buggy) RGB to YUV.

2008-05-12 15:26:01 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	  gst/avi/gstavimux.c: Do not leave fourcc stream header field empty upon reset.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_pad_reset):
	  Do not leave fourcc stream header field empty upon reset.
	  Fixes #519301.

2008-05-11 14:43:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Add goom2k1 into the docs.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-goom2k1.xml:
	  * gst/goom/gstgoom.c:
	  * gst/goom2k1/gstgoom.c:
	  Add goom2k1 into the docs.

2008-05-08 16:58:02 +0000  Wouter Cloetens <wouter@mind.be>

	  gst/rtsp/gstrtspsrc.c: Support Digest authentication. Fixes #532065.
	  Original commit message from CVS:
	  Based on patch by: Wouter Cloetens  <wouter at mind be>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_skip_lws), (gst_rtspsrc_unskip_lws),
	  (gst_rtspsrc_skip_commas), (gst_rtspsrc_skip_item),
	  (gst_rtsp_decode_quoted_string),
	  (gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr),
	  (gst_rtspsrc_setup_auth):
	  Support Digest authentication. Fixes #532065.

2008-05-08 10:20:52 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/level/gstlevel.c: Also support 32bit (e.g. whe having it after 'mad'). Add more notes about whats needed for libo...
	  Original commit message from CVS:
	  * gst/level/gstlevel.c:
	  Also support 32bit (e.g. whe having it after 'mad'). Add more notes
	  about whats needed for liboil acceleration. Simplify docs a bit.

2008-05-08 08:15:34 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/matroska/matroska-mux.c: Update the track duration if the old one was invalid.
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_collected):
	  Update the track duration if the old one was invalid.
	  Fixes bug #532117.

2008-05-07 16:36:04 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  gst/rtp/gstrtph264pay.c (gst_rtp_h264_pay_parse_sps_pps): Use GST_STR_NULL when trying to print sps and pps strings t...
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264pay.c (gst_rtp_h264_pay_parse_sps_pps):
	  Use GST_STR_NULL when trying to print sps and pps strings that could
	  be NULL, as this might crash on some platforms.

2008-05-07 15:33:52 +0000  Haakon Sporsheim <haakon.sporsheim@tandberg.com>

	  sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_setup_ddraw): Do IDirectDrawClipper_SetHWnd() if the window I...
	  Original commit message from CVS:
	  patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
	  * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_setup_ddraw):
	  Do IDirectDrawClipper_SetHWnd() if the window ID has already been
	  set after creating the clipper.

2008-05-07 15:28:06 +0000  Haakon Sporsheim <haakon.sporsheim@tandberg.com>

	  sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame): Added checking of surface lost case after an uns...
	  Original commit message from CVS:
	  patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
	  * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame):
	  Added checking of surface lost case after an unsuccessful
	  IDirectDrawSurface7_Lock() call.
	  If surface is lost, return GST_FLOW_OK.

2008-05-07 15:19:47 +0000  Haakon Sporsheim <haakon.sporsheim@tandberg.com>

	* ChangeLog:
	* sys/directdraw/gstdirectdrawsink.c:
	  sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame,
	  Original commit message from CVS:
	  patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
	  * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame,
	  WndProc, gst_directdraw_sink_window_thread):
	  Improved Windows message loop and fixed window destruction issue.
	  When the window which DirectDraw is rendering to is destroyed, the
	  render/show_frame function will return GST_FLOW_ERROR.
	  Partially fixes #520885.

2008-05-07 15:09:10 +0000  Haakon Sporsheim <haakon.sporsheim@tandberg.com>

	  sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_set_caps): Fixed mid stream resolution change bug, the offscr...
	  Original commit message from CVS:
	  patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
	  * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_set_caps):
	  Fixed mid stream resolution change bug, the offscreen surface is now
	  released when set_caps is called.
	  Partially fixes #520885.

2008-05-07 14:56:22 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	* ChangeLog:
	* sys/directdraw/gstdirectdrawsink.c:
	  sys/directdraw/gstdirectdrawsink.c
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c
	  (gst_directdraw_sink_buffer_alloc):
	  Make it so that gst_directdraw_sink_buffer_alloc uses the right
	  width/height.
	  Especially when looking through the pool of buffers, make sure that
	  the width/height of caps is used instead of the already negotiated
	  dimensions.
	  For example if a buffer with different caps is requested, i.e.
	  higher resolution, the caller would get a buffer with the old
	  dimensions and thus corrupt the heap.

2008-05-07 14:43:39 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	* sys/directdraw/gstdirectdrawsink.c:
	  sys/directdraw/gstdirectdrawsink.c
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c
	  (gst_directdraw_sink_buffer_alloc):
	  Clear the flags on recycled buffers from buffer_alloc.
	  Partially fixes #520885.
	  The right fix this time.

2008-05-07 14:39:45 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	* sys/directdraw/gstdirectdrawsink.c:
	  sys/directdraw/gstdirectdrawsink.c
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c
	  (gst_directdraw_sink_buffer_alloc):
	  Reverting previous commit, it had it all mixed up, was for a different
	  patch (major automation screw-up). Sorry!

2008-05-07 13:48:28 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	* ChangeLog:
	* sys/directdraw/gstdirectdrawsink.c:
	  sys/directdraw/gstdirectdrawsink.c
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c
	  (gst_directdraw_sink_buffer_alloc):
	  Clear the flags on recycled buffers from buffer_alloc.
	  Partially fixes #520885.

2008-05-07 11:22:51 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  gst/rtp/gstrtpilbcpay.c: Added missing stdlib.h include for strtol(), and made include ordering and style consistent ...
	  Original commit message from CVS:
	  * gst/rtp/gstrtpilbcpay.c:
	  Added missing stdlib.h include for strtol(), and made include ordering and
	  style consistent with the corresponding depayloader.

2008-05-07 09:52:34 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  gst/rtp/gstrtpilbcpay.c: Added missing stdlib.h include for strtol(), and made include ordering and style consistent ...
	  Original commit message from CVS:
	  * gst/rtp/gstrtpilbcpay.c:
	  Added missing stdlib.h include for strtol(), and made include ordering and
	  style consistent with the corresponding depayloader.

2008-05-07 08:03:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Error out if we don't have the required core/base versions.
	  Original commit message from CVS:
	  * configure.ac:
	  Error out if we don't have the required core/base versions.

2008-05-06 09:33:46 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  sys/osxvideo/cocoawindow.m: Fix compiler warnings on PPC64. Fixes bug #499318.
	  Original commit message from CVS:
	  Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
	  * sys/osxvideo/cocoawindow.m:
	  Fix compiler warnings on PPC64. Fixes bug #499318.

2008-05-05 11:19:13 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/rtsp/gstrtspsrc.c: Don't leak file descriptors on error. Fixes #531532.
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_open):
	  Don't leak file descriptors on error. Fixes #531532.

2008-05-03 09:18:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/gconf/: When we can't create a fakesink/fakesrc complain instead of unreffing
	  Original commit message from CVS:
	  * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
	  (gst_gconf_audio_src_change_state):
	  * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
	  (gst_gconf_video_sink_change_state):
	  * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
	  (gst_gconf_video_src_change_state):
	  * ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
	  (gst_switch_commit_new_kid), (gst_switch_sink_change_state):
	  When we can't create a fakesink/fakesrc complain instead of unreffing
	  NULL pointers and crashing later. See bug #530535.

2008-05-02 12:44:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph263pdepay.c: Add some more debug info and guard against small payloads.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process):
	  Add some more debug info and guard against small payloads.
	  * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
	  Set duration on outgoing buffers because we can.

2008-05-02 12:39:03 +0000  Olivier Crete <tester@tester.ca>

	  ext/speex/gstspeexenc.c: Add negotiation for the speex channels and rate. Fixes #465146.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * ext/speex/gstspeexenc.c: (gst_speex_enc_sink_getcaps),
	  (gst_speex_enc_init), (gst_speex_enc_chain):
	  Add negotiation for the speex channels and rate. Fixes #465146.

2008-05-02 12:34:22 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/gstrtpspeexpay.c: Add negotiation for the speec channels and rate. See #465146.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init),
	  (gst_rtp_speex_pay_getcaps):
	  Add negotiation for the speec channels and rate. See #465146.

2008-05-02 12:24:55 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/gstrtpilbcpay.c: Add negotiation for the ILBC mode. See #465146.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_class_init),
	  (gst_rtpilbcpay_sink_setcaps), (gst_rtpilbcpay_sink_getcaps):
	  Add negotiation for the ILBC mode. See #465146.

2008-05-02 11:32:31 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/soup/gstsouphttpsrc.c: Include stdlib to fix the build. Use g_free instead of free, libsoup uses glib.
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c:
	  Include stdlib to fix the build. Use g_free instead of free, libsoup
	  uses glib.

2008-05-02 09:09:58 +0000  j^ <j@bootlab.org>

	  gst/qtdemux/qtdemux.c: Add more mpeg2 variants. Fixes #530886.
	  Original commit message from CVS:
	  Patch by: j^ <j@bootlab.org>
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add more mpeg2 variants. Fixes #530886.

2008-05-01 10:52:11 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	  gst/udp/gstudpsrc.c: Don't error out if we get an ICMP destination-unreachable message when trying to read packets on...
	  Original commit message from CVS:
	  Patch by: Youness Alaoui <youness.alaoui at collabora co uk>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Don't error out if we get an ICMP destination-unreachable
	  message when trying to read packets on win32 (#529454).

2008-04-30 12:18:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use new error code for encrypted streams (which requires core CVS).
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  Use new error code for encrypted streams (which requires core CVS).

2008-04-30 12:10:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Fix swapped pad template names, spotted by Thiago Sousa Santos.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_videosrc_template),
	  (gst_qtdemux_audiosrc_template):
	  Fix swapped pad template names, spotted by Thiago Sousa Santos.

2008-04-30 09:48:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/speex/gstspeexdec.c: Produce concealment data when time progresses in a segment update.
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_sink_event),
	  (speex_dec_chain_parse_data):
	  Produce concealment data when time progresses in a segment update.

2008-04-29 14:11:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/speex/gstspeexdec.c: Try to preserve input timestamps when we can.
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data),
	  (speex_dec_chain):
	  Try to preserve input timestamps when we can.
	  Do beginnings of error concealment.

2008-04-28 22:38:11 +0000  Michael Smith <msmith@xiph.org>

	  gst/debug/gstnavigationtest.c: MSVC doesn't provide rint(), define an adequate replacement locally as elsewhere.
	  Original commit message from CVS:
	  * gst/debug/gstnavigationtest.c:
	  MSVC doesn't provide rint(), define an adequate replacement locally as
	  elsewhere.

2008-04-28 11:16:32 +0000  Julien Moutte <julien@moutte.net>

	  gst/debug/rndbuffersize.c: Fix printf format to pacify Mac OSX's gcc.
	  Original commit message from CVS:
	  2008-04-28  Julien Moutte  <julien@fluendo.com>
	  * gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop): Fix printf
	  format to pacify Mac OSX's gcc.

2008-04-25 19:34:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/debug/rndbuffersize.c: Bring rndbuffersize element into a state that doesn't require us to move it to -bad immedi...
	  Original commit message from CVS:
	  * gst/debug/rndbuffersize.c: (DEFAULT_SEED), (DEFAULT_MIN),
	  (DEFAULT_MAX), (src_template), (sink_template),
	  (gst_rnd_buffer_size_base_init), (gst_rnd_buffer_size_class_init),
	  (gst_rnd_buffer_size_init), (gst_rnd_buffer_size_activate),
	  (gst_rnd_buffer_size_loop), (gst_rnd_buffer_size_plugin_init):
	  Bring rndbuffersize element into a state that doesn't require us
	  to move it to -bad immediately. For one, fix up default min/max
	  values so that the element actuall works using the default values.
	  Also, don't ignore flow return values and do some kind of minimal
	  eos logic. Allow min=max to pull fixed-sized buffers. Bunch of
	  other gratuitious clean-ups.

2008-04-25 19:24:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Add docs for gdkpixbufsink; update docs to CVS version.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/gst-plugins-good-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-1394.xml:
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gamma.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-monoscope.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multifile.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-shout2send.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-video4linux2.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  Add docs for gdkpixbufsink; update docs to CVS version.

2008-04-25 18:45:33 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Remove test sync-offset by default.
	  Original commit message from CVS:
	  * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  Remove test sync-offset by default.

2008-04-25 13:31:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Use GLib versions of htonl, htons, ntohl and ntohs in order to avoid problems on win32 (#529707).
	  Original commit message from CVS:
	  * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_chain):
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add_internal):
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  Use GLib versions of htonl, htons, ntohl and ntohs in order
	  to avoid problems on win32 (#529707).

2008-04-25 12:52:44 +0000  Jesús Corrius <jesus@softcatala.org>

	  gst/goom/: Fix build with mingw32: use rand() instead of random() and replace bzero() with memset(). Fixes #529692.
	  Original commit message from CVS:
	  Patch by: Jesús Corrius <jesus at softcatala org>
	  * gst/goom/filters.c: (zoomVector):
	  * gst/goom/goom_core.c: (init_buffers):
	  Fix build with mingw32: use rand() instead of random() and
	  replace bzero() with memset(). Fixes #529692.

2008-04-25 07:56:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Fix typo in comments.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows):
	  Fix typo in comments.
	  * tests/examples/rtp/client-H263p-PCMA.sdp:
	  * tests/examples/rtp/client-H263p-PCMA.sh:
	  * tests/examples/rtp/client-H264-PCMA.sdp:
	  * tests/examples/rtp/client-H264-PCMA.sh:
	  * tests/examples/rtp/client-H264.sdp:
	  * tests/examples/rtp/client-H264.sh:
	  * tests/examples/rtp/client-PCMA.sdp:
	  * tests/examples/rtp/client-PCMA.sh:
	  * tests/examples/rtp/server-alsasrc-PCMA.sh:
	  * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	  * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  Add some more docs and fix examples.

2008-04-24 22:04:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/multifile.c: Include stdlib.h and unistd.h for mkdtemp. Some platforms have it declared in the f...
	  Original commit message from CVS:
	  * tests/check/elements/multifile.c:
	  Include stdlib.h and unistd.h for mkdtemp. Some platforms have it
	  declared in the former, some have it declared in the latter.

2008-04-24 22:01:52 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Stop using deprecated GLib functions.
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_set_property):
	  * gst/debug/tests.c: (md5_get_value):
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
	  * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	  * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps):
	  * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps):
	  Stop using deprecated GLib functions.

2008-04-24 21:17:42 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to development -> 0.10.8.1
	  Original commit message from CVS:
	  * configure.ac:
	  Back to development -> 0.10.8.1
	  === release 0.10.8 ===

=== release 0.10.8 ===

2008-04-23 23:40:48 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* NEWS:
	* RELEASE:
	  Release 0.10.8 a little harder (edited the release notes)
	  Original commit message from CVS:
	  Release 0.10.8 a little harder (edited the release notes)

2008-04-23 23:26:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* gst-plugins-good.doap:
	* po/LINGUAS:
	* win32/common/config.h:
	  Release 0.10.8
	  Original commit message from CVS:
	  Release 0.10.8

2008-04-23 23:18:44 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* common:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/it.po:
	* po/ja.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2008-04-22 00:29:00 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: 0.10.7.4 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  0.10.7.4 pre-release

2008-04-22 00:18:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/goom/: Free a bunch of stuff, and initialise things to fix leaks and valgrind warnings in the testsuite.
	  Original commit message from CVS:
	  * gst/goom/config_param.c: (goom_plugin_parameters_free):
	  * gst/goom/convolve_fx.c: (convolve_init), (convolve_free):
	  * gst/goom/filters.c: (zoomFilterVisualFXWrapper_free):
	  * gst/goom/flying_stars_fx.c: (fs_free):
	  * gst/goom/goom_config_param.h:
	  * gst/goom/goom_core.c: (goom_init), (goom_close):
	  * gst/goom/goom_plugin_info.h:
	  * gst/goom/gstgoom.c: (gst_goom_finalize):
	  * gst/goom/lines.c: (goom_lines_free):
	  * gst/goom/plugin_info.c: (plugin_info_init), (plugin_info_free):
	  * gst/goom/surf3d.c: (grid3d_free):
	  * gst/goom/surf3d.h:
	  * gst/goom/tentacle3d.c: (tentacle_free):
	  Free a bunch of stuff, and initialise things to fix leaks
	  and valgrind warnings in the testsuite.
	  Fixes: #529268

2008-04-21 21:54:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/rganalysis.c: Don't leak a tag list. Fixes bug #529285.
	  Original commit message from CVS:
	  * tests/check/elements/rganalysis.c: (GST_START_TEST):
	  Don't leak a tag list. Fixes bug #529285.

2008-04-21 08:21:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Ref caps as the return value for the request_pt_map signal.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (request_pt_map),
	  (gst_rtspsrc_configure_caps):
	  Ref caps as the return value for the request_pt_map signal.
	  Remove some caps weirdness when configuring a stream. See #528245.

2008-04-18 18:47:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/gdkpixbufsink-test.c: Add cast to placate gcc 4.1.2.
	  Original commit message from CVS:
	  * tests/icles/gdkpixbufsink-test.c:
	  Add cast to placate gcc 4.1.2.

2008-04-17 23:00:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: 0.10.7.3 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  0.10.7.3 pre-release

2008-04-17 22:32:16 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Disable some more elements in the state test.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Disable some more elements in the state test.
	  Add a define so the soup test can find the test files
	  it needs at runtime.
	  * tests/check/elements/souphttpsrc.c: (run_server):
	  Add a define so the soup test can find the test files
	  it needs at runtime.

2008-04-17 18:08:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/goom/convolve_fx.c: Don't ever draw the GOOM logo.
	  Original commit message from CVS:
	  * gst/goom/convolve_fx.c: (convolve_apply):
	  Don't ever draw the GOOM logo.
	  Fixes: #528615

2008-04-17 10:24:32 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/: gst_atomic_int_set ==> g_atomic_int_set
	  Original commit message from CVS:
	  * ext/cdio/gstcdiocddasrc.c:
	  * ext/dv/gstdvdemux.c:
	  gst_atomic_int_set ==> g_atomic_int_set

2008-04-16 10:31:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Strip out the config/script parsing stuff, we don't need it.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/goom/Makefile.am:
	  * gst/goom/convolve_fx.c:
	  * gst/goom/default_scripts.h:
	  * gst/goom/goom.h:
	  * gst/goom/goom_core.c: (choose_a_goom_line):
	  * gst/goom/goom_plugin_info.h:
	  * gst/goom/goomsl.c:
	  * gst/goom/goomsl.h:
	  * gst/goom/goomsl_hash.c:
	  * gst/goom/goomsl_hash.h:
	  * gst/goom/goomsl_heap.c:
	  * gst/goom/goomsl_heap.h:
	  * gst/goom/goomsl_private.h:
	  * gst/goom/plugin_info.c:
	  Strip out the config/script parsing stuff, we don't need it.
	  Fixes #527999.

2008-04-15 16:58:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/plugin_info.c: Disable altivec optimisations for 32-bit PPC as well to make things build properly on all PPC...
	  Original commit message from CVS:
	  * gst/goom/plugin_info.c: (setOptimizedMethods):
	  Disable altivec optimisations for 32-bit PPC as well to make
	  things build properly on all PPC systems. Fixes #528143

2008-04-14 20:01:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-plugins-good.spec.in: Update for souphttpsrc plugin which has moved to -good.
	  Original commit message from CVS:
	  * gst-plugins-good.spec.in:
	  Update for souphttpsrc plugin which has moved to -good.

2008-04-14 13:38:32 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/matroska/matroska-demux.c: Fix open-ended seeks in matroskademux
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_handle_seek_event):
	  Fix open-ended seeks in matroskademux
	  Patch by: Mark Nauwelaerts <manauw skynet be>
	  Fixes: #526557

2008-04-13 23:13:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Add soup test certificates to the dist.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Add soup test certificates to the dist.

2008-04-13 17:43:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/Makefile.am: Remove LADSPA reference I missed.
	  Original commit message from CVS:
	  * ext/Makefile.am:
	  Remove LADSPA reference I missed.

2008-04-13 13:06:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/soup/gstsouphttpsrc.c: Give souphttpsrc GST_RANK_PRIMARY to make it the default HTTP source over gnome-vfs and ev...
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c: (plugin_init):
	  Give souphttpsrc GST_RANK_PRIMARY to make it the default HTTP source
	  over gnome-vfs and everything else. Fixes bug #527848.

2008-04-12 23:47:23 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Remove LADSPA plugin. Fixes: #515978
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/Makefile.am:
	  Remove LADSPA plugin. Fixes: #515978

2008-04-12 23:30:54 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move soup plugin from -bad (Fixes: #523124)
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-soup.xml:
	  * ext/Makefile.am:
	  * tests/check/Makefile.am:
	  Move soup plugin from -bad (Fixes: #523124)

2008-04-11 11:08:35 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Fix the Changelog - actually speex <= 1.1.12 are vulnerable.
	  Original commit message from CVS:
	  Fix the Changelog - actually speex <= 1.1.12 are vulnerable.

2008-04-11 10:32:20 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/speex/gstspeexdec.c: Fix bounds checking of mode in Speex header, which may produce negative numbers in speex < 1...
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_header):
	  Fix bounds checking of mode in Speex header, which may
	  produce negative numbers in speex < 1.1.12

2008-04-10 07:11:51 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/souphttpsrc.c: Increase the timeout for the internet tests to 250 seconds and check for NULL cap...
	  Original commit message from CVS:
	  * tests/check/elements/souphttpsrc.c: (got_buffer),
	  (souphttpsrc_suite):
	  Increase the timeout for the internet tests to 250 seconds
	  and check for NULL caps instead of just crashing.
	  The real fix would be to implement an shoutcast server for the unit test
	  instead of relying on a working internet connection.
	  Fixes bug #521749.

2008-04-09 16:11:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/: Remove a bunch of font/text related code that we don't need.
	  Original commit message from CVS:
	  * gst/goom/Makefile.am:
	  * gst/goom/gfontlib.c:
	  * gst/goom/gfontlib.h:
	  * gst/goom/gfontrle.c:
	  * gst/goom/gfontrle.h:
	  * gst/goom/goom.h:
	  * gst/goom/goom_core.c: (goom_update):
	  * gst/goom/goom_plugin_info.h:
	  * gst/goom/gstgoom.c: (gst_goom_chain):
	  * gst/goom/plugin_info.c:
	  Remove a bunch of font/text related code that we don't need.

2008-04-09 14:02:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/: Change license of these files to LGPL, as permitted by the author, Guillaume Borios. See #515073.
	  Original commit message from CVS:
	  * gst/goom/ppc_drawings.s:
	  * gst/goom/ppc_zoom_ultimate.s:
	  Change license of these files to LGPL, as permitted by the
	  author, Guillaume Borios. See #515073.

2008-04-09 13:31:22 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/goom/: As hinted in Bug #518213, revert one change and fix warnings properly.
	  Original commit message from CVS:
	  * gst/goom/convolve_fx.c:
	  * gst/goom/motif_goom1.h:
	  * gst/goom/motif_goom2.h:
	  As hinted in Bug #518213, revert one change and fix warnings properly.
	  This fixes both #518213 and #520073 for me.

2008-04-09 12:02:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/matroska/: Fix the Forte build by making function declaration signatures match the implementations.
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_read_seek):
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_handle_seek_event),
	  (gst_matroska_demux_parse_contents_seekentry),
	  (gst_matroska_demux_loop):
	  Fix the Forte build by making function declaration signatures
	  match the implementations.

2008-04-08 19:49:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/: More logging when probing (see #518474), some comments in _reset().
	  Original commit message from CVS:
	  * sys/oss/gstosshelper.c: (gst_oss_helper_rate_check_rate):
	  * sys/oss/gstosssink.c: (gst_oss_sink_reset):
	  * sys/oss/gstosssrc.c: (gst_oss_src_reset):
	  More logging when probing (see #518474), some comments in _reset().

2008-04-07 17:18:48 +0000  Julien Moutte <julien@moutte.net>

	  gst/rtp/gstrtph264pay.c: Fix build because of a bad argument number.
	  Original commit message from CVS:
	  2008-04-07  Julien Moutte  <julien@fluendo.com>
	  * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps): Fix build
	  because of a bad argument number.

2008-04-06 18:28:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/: Interactive test app for gdkpixbufsink.
	  Original commit message from CVS:
	  * tests/icles/.cvsignore:
	  * tests/icles/Makefile.am:
	  * tests/icles/gdkpixbufsink-test.c:
	  Interactive test app for gdkpixbufsink.

2008-04-06 09:01:42 +0000  Sjoerd Simons <sjoerd@luon.net>

	  ext/soup/gstsouphttpsrc.c: Only ignore actual redirects not all responses when in state
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_response_cb):
	  Only ignore actual redirects not all responses when in state
	  GST_SOUP_HTTP_SRC_SESSION_IO_STATUS_RUNNING. Fixes bug #526337.

2008-04-06 08:57:59 +0000  Damien Lespiau <damien.lespiau@gmail.com>

	  configure.ac: Actually build dlls when cross-compiling with mingw32.
	  Original commit message from CVS:
	  Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
	  * configure.ac:
	  Actually build dlls when cross-compiling with mingw32.
	  Fixes bug #526247.

2008-04-05 12:00:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/hal/hal.c: Don't munge device string to 'default:x' for capture devices.
	  Original commit message from CVS:
	  * ext/hal/hal.c: (gst_hal_get_alsa_element):
	  Don't munge device string to 'default:x' for capture devices.
	  Fixes #525833.

2008-04-04 19:00:19 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Always use GSlice as we actually depend on GLib 2.12 already.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_index_entry_free):
	  Always use GSlice as we actually depend on GLib 2.12 already.

2008-04-04 11:26:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Require core/base 0.10.18 for ARGB caps parsing fixes in libgstvideo.
	  Original commit message from CVS:
	  * configure.ac:
	  Require core/base 0.10.18 for ARGB caps parsing fixes in libgstvideo.
	  Also bump the GLib requirement to the current de-facto requirement
	  (ie. 2.12).

2008-04-04 10:32:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph264pay.*: Parse codec_data for future AVC compatibility.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264pay.c: (encode_base64),
	  (gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_handle_buffer):
	  * gst/rtp/gstrtph264pay.h:
	  Parse codec_data for future AVC compatibility.
	  Fail when we encounter AVC data for now.

2008-04-04 09:50:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/spectrum/gstspectrum.c: Rename property enums and default defines for the properties to match the property names ...
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
	  (gst_spectrum_init), (gst_spectrum_set_property),
	  (gst_spectrum_get_property), (gst_spectrum_message_new):
	  Rename property enums and default defines for the properties to match
	  the property names and rephrase property descriptions to make them a
	  bit clearer (hopefully). See #518188.

2008-04-03 22:59:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add unit test for gdkpixbufsink element.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/gdkpixbufsink.c:
	  Add unit test for gdkpixbufsink element.

2008-04-03 22:50:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gdk_pixbuf/: Add gdkpixbufsink element for easy snapshotting (#525946).
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/Makefile.am:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init):
	  * ext/gdk_pixbuf/gstgdkpixbufsink.c:
	  (gst_gdk_pixbuf_sink_base_init),
	  (gst_gdk_pixbuf_sink_class_init), (gst_gdk_pixbuf_sink_init),
	  (gst_gdk_pixbuf_sink_start), (gst_gdk_pixbuf_sink_stop),
	  (gst_gdk_pixbuf_sink_set_caps),
	  (gst_gdk_pixbuf_sink_pixbuf_destroy_notify),
	  (gst_gdk_pixbuf_sink_get_pixbuf_from_buffer),
	  (gst_gdk_pixbuf_sink_handle_buffer), (gst_gdk_pixbuf_sink_preroll),
	  (gst_gdk_pixbuf_sink_render), (gst_gdk_pixbuf_sink_set_property),
	  (gst_gdk_pixbuf_sink_get_property):
	  * ext/gdk_pixbuf/gstgdkpixbufsink.h:
	  Add gdkpixbufsink element for easy snapshotting (#525946).

2008-04-03 20:25:34 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/pipelines/wavpack.c: Bump timeout from 3 to 60 seconds.
	  Original commit message from CVS:
	  * tests/check/pipelines/wavpack.c: (wavpack_suite):
	  Bump timeout from 3 to 60 seconds.

2008-04-03 20:21:15 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/pipelines/.cvignore: Remove useless file.
	  Original commit message from CVS:
	  * tests/check/pipelines/.cvignore:
	  Remove useless file.
	  * tests/check/pipelines/.cvsignore:
	  Add new test to .cvsignore.

2008-04-03 20:05:31 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/: Add unit test that encodes and decodes some data, checks that it is still the same and that all timesta...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/wavpack.c: (bus_handler),
	  (identity_handoff), (fakesink_handoff), (GST_START_TEST),
	  (wavpack_suite), (main):
	  Add unit test that encodes and decodes some data, checks that it
	  is still the same and that all timestamps/offsets are perfect.

2008-04-03 18:28:28 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: Use GSlice for allocating index entries and use gst_element_class_set_details_simple().
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_base_init):
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_index_entry_new),
	  (gst_wavpack_parse_index_entry_free),
	  (gst_wavpack_parse_base_init),
	  (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset):
	  Use GSlice for allocating index entries and use
	  gst_element_class_set_details_simple().

2008-04-02 22:37:29 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/: Fix up copyrights (#525860).
	  Original commit message from CVS:
	  Patch by: Brian Cameron <brian.cameron at sun dot com>
	  * sys/sunaudio/gstsunaudio.c:
	  * sys/sunaudio/gstsunaudiomixer.c:
	  * sys/sunaudio/gstsunaudiomixer.h:
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  * sys/sunaudio/gstsunaudiomixerctrl.h:
	  * sys/sunaudio/gstsunaudiomixertrack.c:
	  * sys/sunaudio/gstsunaudiomixertrack.h:
	  * sys/sunaudio/gstsunaudiosink.c:
	  * sys/sunaudio/gstsunaudiosink.h:
	  * sys/sunaudio/gstsunaudiosrc.c:
	  * sys/sunaudio/gstsunaudiosrc.h:
	  Fix up copyrights (#525860).

2008-04-02 16:10:33 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add new goom plugin to spec file
	  Original commit message from CVS:
	  add new goom plugin to spec file

2008-04-02 15:42:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/goomsl.c: Check return value of fread() to avoid compiler warnings.
	  Original commit message from CVS:
	  * gst/goom/goomsl.c: (gsl_read_file):
	  Check return value of fread() to avoid compiler warnings.

2008-04-01 11:00:43 +0000  mersad <mersad@axis.com>

	  gst/law/: Make negotiation a bit modern.
	  Original commit message from CVS:
	  Based on patch by: mersad <mersad at axis dot com>
	  * gst/law/alaw-decode.c: (gst_alaw_dec_sink_setcaps),
	  (gst_alaw_dec_chain), (gst_alaw_dec_change_state):
	  * gst/law/alaw-decode.h:
	  * gst/law/alaw-encode.c: (gst_alaw_enc_chain):
	  * gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
	  (gst_mulawdec_chain), (gst_mulawdec_change_state):
	  * gst/law/mulaw-decode.h:
	  * gst/law/mulaw-encode.c: (gst_mulawenc_chain):
	  Make negotiation a bit modern.
	  Use pad_alloc. Fixes #525359.

2008-03-31 22:06:14 +0000  David Schleef <ds@schleef.org>

	  gst/goom/xmmx.c: Fix constraints on asm code so that it compiles consistently.  Fixes #522278.
	  Original commit message from CVS:
	  * gst/goom/xmmx.c: Fix constraints on asm code so that it
	  compiles consistently.  Fixes #522278.

2008-03-27 09:36:58 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/: Fix up the mixer tracks to use a volume range of 0-255, which is what the sun audio API uses. This sim...
	  Original commit message from CVS:
	  Patch by: Brian Cameron <brian.cameron at sun dot com>
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_get_volume),
	  (gst_sunaudiomixer_ctrl_set_volume):
	  * sys/sunaudio/gstsunaudiomixertrack.c: (gst_sunaudiomixer_track_new):
	  Fix up the mixer tracks to use a volume range of 0-255, which is what
	  the sun audio API uses. This simplifies the code and avoids rounding
	  errors. Fixes #524593.

2008-03-26 15:10:08 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  Add device-fd property to make it possible to apps to call ioctl's.
	  Original commit message from CVS:
	  Add device-fd property to make it possible to apps to call ioctl's.

2008-03-25 16:44:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Unbreak streaming mode again.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (next_entry_size):
	  Unbreak streaming mode again.

2008-03-25 12:39:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/v4l2src_calls.c: Remove superfluous DEBUG macro.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture):
	  Remove superfluous DEBUG macro.

2008-03-25 12:33:09 +0000  William M. Brack <wbrack@mmm.com.hk>

	  sys/v4l2/v4l2src_calls.c: Check whether the device supports setting the framerate before trying to set it and then po...
	  Original commit message from CVS:
	  Based on patch by: William M. Brack <wbrack at mmm com hk>
	  * sys/v4l2/v4l2src_calls.c: (fractions_are_equal),
	  (gst_v4l2src_set_capture):
	  Check whether the device supports setting the framerate before
	  trying to set it and then posting a warning or error if it doesn't
	  work (#516649, #520092). Also compare fractions more correctly.

2008-03-24 12:32:59 +0000  Rene Stadler <mail@renestadler.de>

	  Make rganalysis and rglimiter elements GAP-flag aware.
	  Original commit message from CVS:
	  * gst/replaygain/gstrganalysis.c (gst_rg_analysis_init),
	  (gst_rg_analysis_transform_ip):
	  * gst/replaygain/gstrglimiter.c (gst_rg_limiter_init),
	  (gst_rg_limiter_transform_ip):
	  Make rganalysis and rglimiter elements GAP-flag aware.
	  * tests/check/elements/rganalysis.c: (test_gap_buffers),
	  (rganalysis_suite):
	  * tests/check/elements/rglimiter.c (test_gap), (rglimiter_suite):
	  Add tests to verify gap-awareness.

2008-03-23 13:31:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/Makefile.am: Remove ppc assembler optimisations from the build until they actually build (they also seem to ...
	  Original commit message from CVS:
	  * gst/goom/Makefile.am:
	  Remove ppc assembler optimisations from the build until they
	  actually build (they also seem to have GPL headers).

2008-03-23 12:48:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  m4/Makefile.am: Better not dist files that don't exist any longer (lrint*m4).
	  Original commit message from CVS:
	  * m4/Makefile.am:
	  Better not dist files that don't exist any longer (lrint*m4).

2008-03-22 19:26:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/soup/gstsouphttpsrc.c: Don't autoplug souphttpsrc for dav/davs. This is better handled by
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_headers_cb),
	  (gst_soup_http_src_chunk_allocator),
	  (gst_soup_http_src_got_chunk_cb),
	  (gst_soup_http_src_uri_get_protocols):
	  Don't autoplug souphttpsrc for dav/davs. This is better handled by
	  GIO and GnomeVFS as they provide authentication.
	  Don't leak the icy caps if we already set them and get a new
	  icy-metaint header.
	  Try harder to set the icy caps on the output buffer to have correct
	  caps for the first buffer already.
	  * tests/check/elements/souphttpsrc.c: (got_buffer),
	  (GST_START_TEST):
	  Check that we get a buffer with application/x-icy caps if iradio-mode
	  is enabled and we have an icecast URL.

2008-03-22 18:18:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/soup/gstsouphttpsrc.c: Actually set the icy caps on our src pad if we have icecast data.
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_chunk_allocator):
	  Actually set the icy caps on our src pad if we have icecast data.
	  Fixes bug #523854.

2008-03-21 13:36:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Remove lrint/lrintf checks. We don't use it anywhere.
	  Original commit message from CVS:
	  * configure.ac:
	  * m4/lrint.m4:
	  * m4/lrintf.m4:
	  Remove lrint/lrintf checks. We don't use it anywhere.

2008-03-19 19:56:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/freeze/: Add example to source code documentation blob and remove the 3 line
	  Original commit message from CVS:
	  * gst/freeze/FAQ:
	  * gst/freeze/Makefile.am:
	  * gst/freeze/gstfreeze.c:
	  Add example to source code documentation blob and remove the 3 line
	  FAQ.
	  * gst/interleave/interleave.c:
	  Add a source code documentation blob.

2008-03-18 15:03:06 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  sys/osxvideo/osxvideosink.m (gst_osx_video_sink_osxwindow_destroy)
	  Original commit message from CVS:
	  2008-03-18  Andy Wingo  <wingo@pobox.com>
	  * sys/osxvideo/osxvideosink.m
	  (gst_osx_video_sink_osxwindow_destroy)
	  (gst_osx_video_sink_osxwindow_new): Actually set a lock on the
	  task, whoopdee.
	  (cocoa_event_loop): Pacify the taymans by upping the usleepage to
	  2 ms.

2008-03-18 11:50:08 +0000  Andy Wingo <wingo@pobox.com>

	  sys/osxvideo/osxvideosink.m (gst_osx_video_sink_osxwindow_destroy)
	  Original commit message from CVS:
	  2008-03-18  Andy Wingo  <wingo@pobox.com>
	  * sys/osxvideo/osxvideosink.m (gst_osx_video_sink_osxwindow_destroy)
	  (gst_osx_video_sink_osxwindow_new, cocoa_event_loop):
	  * sys/osxvideo/osxvideosink.h (struct _GstOSXVideoSink): If we
	  need to run an event loop, do so in a task instead of assuming
	  that there will be a GMainLoop. Fixes #523134.

2008-03-17 19:50:58 +0000  William M. Brack <wbrack@mmm.com.hk>

	  sys/v4l2/v4l2src_calls.c: Make sure the probed frame sizes are reversed in the resulting caps also when using V4L2_FR...
	  Original commit message from CVS:
	  Patch by: William M. Brack <wbrack at mmm com hk>
	  * sys/v4l2/v4l2src_calls.c:
	  (gst_v4l2src_probe_caps_for_format_and_size),
	  (gst_v4l2src_probe_caps_for_format):
	  Make sure the probed frame sizes are reversed in the resulting
	  caps also when using V4L2_FRMSIZE_STEPWISE (so they end up
	  highest resolution first); also remove unused variable.
	  (Partly fixes #520092)

2008-03-17 15:56:01 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  gst/rtsp/gstrtspsrc.c: Call WSAStartup() and WSACleanup before using the Winsock API.
	  Original commit message from CVS:
	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize):
	  Call WSAStartup() and WSACleanup before using the Winsock API.
	  See #520808.

2008-03-16 15:01:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Erm, the buffer-size is just guint, no need for the special format specifier.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Erm, the buffer-size is just guint, no need for the special format
	  specifier.

2008-03-16 14:34:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/: Small fixes to build more on PPC: ifdef out code that uses unknown define; add newline at end of header fi...
	  Original commit message from CVS:
	  * gst/goom/plugin_info.c:
	  * gst/goom/ppc_zoom_ultimate.h:
	  Small fixes to build more on PPC: ifdef out code that uses unknown
	  define; add newline at end of header file to avoid compiler warning.
	  Assembler code still doesn't build though.

2008-03-16 14:04:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Fix up my last commit. Use G_GUINT32_FORMAT for the guint32 debug log.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Fix up my last commit. Use G_GUINT32_FORMAT for the guint32 debug log.
	  Also downgrade a GST_WARNING to GST_DEBUG and add a comment.

2008-03-15 22:10:38 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Chunksize is uint32. Fix format specifier.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Chunksize is uint32. Fix format specifier.

2008-03-14 15:53:01 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	* gst/rtsp/COPYING.MIT:
	  fix license file, remove extra line copied over by mistake
	  Original commit message from CVS:
	  fix license file, remove extra line copied over by mistake

2008-03-13 14:30:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/audiofx.c: Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values.
	  Original commit message from CVS:
	  * gst/audiofx/audiofx.c:
	  Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead
	  of hardcoding values.

2008-03-13 09:45:09 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.*: Try to resume on server disconnect. Fixes bug #522134.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_init),
	  (gst_soup_http_src_finished_cb), (gst_soup_http_src_response_cb),
	  (gst_soup_http_src_build_message), (gst_soup_http_src_create):
	  * ext/soup/gstsouphttpsrc.h:
	  Try to resume on server disconnect. Fixes bug #522134.

2008-03-11 23:12:04 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  sys/oss/gstosssrc.*: Cache probed caps, so _get_caps() during recording doesn't cause ioctl calls which may disrupt t...
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw skynet be>
	  * sys/oss/gstosssrc.c: (gst_oss_src_init), (gst_oss_src_getcaps),
	  (gst_oss_src_close):
	  * sys/oss/gstosssrc.h:
	  Cache probed caps, so _get_caps() during recording doesn't cause
	  ioctl calls which may disrupt the recording (fixes #521875).

2008-03-11 16:23:04 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Make sure we always send a DISCONT after a seek by setting the sample index to an undefined va...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
	  (gst_qtdemux_activate_segment),
	  (gst_qtdemux_prepare_current_sample),
	  (gst_qtdemux_loop_state_movie), (qtdemux_parse_trak):
	  Make sure we always send a DISCONT after a seek by setting the sample
	  index to an undefined value after a seek.

2008-03-11 15:18:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavisubtitle.h: Fix up IS_FOO macros, which makes gtk-doc much happier.
	  Original commit message from CVS:
	  * gst/avi/gstavisubtitle.h: (GST_IS_AVI_SUBTITLE),
	  (GST_IS_AVI_SUBTITLE_CLASS):
	  Fix up IS_FOO macros, which makes gtk-doc much happier.

2008-03-08 19:29:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/Makefile.am: Move the -lgstfoo where it belongs.
	  Original commit message from CVS:
	  * tests/icles/Makefile.am:
	  Move the -lgstfoo where it belongs.

2008-03-08 19:14:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery
	  Original commit message from CVS:
	  ChangeLog surgery

2008-03-08 04:40:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/matroska/ebml-ids.h: Add ID for EBML CRC32 elements.
	  Original commit message from CVS:
	  * gst/matroska/ebml-ids.h:
	  Add ID for EBML CRC32 elements.
	  * gst/matroska/Makefile.am:
	  * gst/matroska/ebml-read.c: (gst_ebml_finalize),
	  (gst_ebml_read_class_init), (gst_ebml_read_peek_bytes),
	  (gst_ebml_read_get_length), (_ext2dbl), (gst_ebml_read_float),
	  (gst_ebml_read_header):
	  Support reading 80bit floats, add finalize method to clean up
	  in any case, support reading length/id elements with any length
	  as long as it's smaller than our supported maximum, don't leak
	  buffers if reading as much data as we wanted failed and some
	  smaller cleanup.

2008-03-08 04:21:34 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/gstrtph263pdepay.c: Check that a buffer is large enough before reading from it.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process):
	  Check that a buffer is large enough before reading from it.
	  Fixes bug #521102.

2008-03-07 15:54:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Fix compilation after removing the GstPollMode from the constructor.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  Fix compilation after removing the GstPollMode from the
	  constructor.

2008-03-07 13:08:42 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Check for sinh(), cosh() and asinh() and define our own implementations if they're not available. Fixes bug #520880.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiochebband.c:
	  * gst/audiofx/audiocheblimit.c:
	  * gst/audiofx/math_compat.h:
	  Check for sinh(), cosh() and asinh() and define our own
	  implementations if they're not available. Fixes bug #520880.

2008-03-07 12:40:18 +0000  Olivier Crete <tester@tester.ca>

	  ext/speex/gstspeexenc.c: Unref the buffers only once when handling not-negotiated errors.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * ext/speex/gstspeexenc.c: (gst_speex_enc_chain):
	  Unref the buffers only once when handling not-negotiated errors.
	  Fixes bug #520764.

2008-03-07 10:01:40 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  gst/udp/gstudpsrc.c: Properly balance WSA_Cleanup with WSA_Startup.
	  Original commit message from CVS:
	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_finalize), (gst_udpsrc_start),
	  (gst_udpsrc_stop):
	  Properly balance WSA_Cleanup with WSA_Startup.
	  Also make the poll controllable on windows. Fixes #520888.

2008-03-06 19:47:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/matroska/: Handle return values from pull_range in a more granular way to properly shut down on seeks.
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes),
	  (gst_ebml_read_pull_bytes), (gst_ebml_read_element_id),
	  (gst_ebml_read_element_length), (gst_ebml_peek_id),
	  (gst_ebml_read_skip), (gst_ebml_read_buffer),
	  (gst_ebml_read_bytes), (gst_ebml_read_uint), (gst_ebml_read_sint),
	  (gst_ebml_read_float), (gst_ebml_read_ascii), (gst_ebml_read_utf8),
	  (gst_ebml_read_date), (gst_ebml_read_master),
	  (gst_ebml_read_binary), (gst_ebml_read_header):
	  * gst/matroska/ebml-read.h:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_combine_flows), (gst_matroska_demux_reset),
	  (gst_matroska_demux_read_track_encodings),
	  (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_handle_src_query),
	  (gst_matroska_demux_handle_seek_event),
	  (gst_matroska_demux_init_stream),
	  (gst_matroska_demux_parse_tracks),
	  (gst_matroska_demux_parse_index_cuetrack),
	  (gst_matroska_demux_parse_index_pointentry),
	  (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_parse_metadata_id_simple_tag),
	  (gst_matroska_demux_parse_metadata_id_tag),
	  (gst_matroska_demux_parse_metadata),
	  (gst_matroska_demux_sync_streams),
	  (gst_matroska_demux_push_hdr_buf),
	  (gst_matroska_demux_push_flac_codec_priv_data),
	  (gst_matroska_demux_push_xiph_codec_priv_data),
	  (gst_matroska_demux_add_wvpk_header),
	  (gst_matroska_demux_check_subtitle_buffer),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_parse_cluster),
	  (gst_matroska_demux_parse_contents_seekentry),
	  (gst_matroska_demux_parse_contents),
	  (gst_matroska_demux_loop_stream_parse_id),
	  (gst_matroska_demux_loop_stream), (gst_matroska_demux_loop):
	  * gst/matroska/matroska-demux.h:
	  * gst/matroska/matroska-ids.h:
	  Handle return values from pull_range in a more granular way to properly
	  shut down on seeks.
	  Combine return values from push.
	  Implement proper error handling.
	  Prepare for handling seeking correctly.

2008-03-03 22:01:56 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/matroska/ebml-read.c: Use GINT64 formatting constants from GLIB.
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c:
	  Use GINT64 formatting constants from GLIB.
	  * gst/matroska/matroska-demux.c:
	  Add some guards to avoid a possible division by 0 and crashing
	  with NULL events on some systems.
	  Use gst_gdouble_to_guint64 somewhere instead of an implicit
	  conversion.
	  * gst/matroska/matroska-mux.c:
	  Check for invalid timestamps in a bunch of places to avoid
	  writing bogus durations into the output file.
	  Fix some double<->gint64 conversions that weren't using
	  gst_guint64_to_gdouble

2008-03-03 13:03:43 +0000  Peter Kjellerstedt <pkj@axis.com>

	  configure.ac: Move the checks for bison, flex and as to the program section and the check for gcc inline asm to the c...
	  Original commit message from CVS:
	  * configure.ac:
	  Move the checks for bison, flex and as to the program section and the
	  check for gcc inline asm to the compiler characteristics section.

2008-03-03 12:10:55 +0000  Peter Kjellerstedt <pkj@axis.com>

	  configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
	  Original commit message from CVS:
	  * configure.ac:
	  Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
	  plug-ins are included/excluded. (#498222)

2008-02-29 12:35:24 +0000  Michael Smith <msmith@xiph.org>

	  gst/videomixer/videomixer.c: Don't call gst_object_sync_values() unless we have a valid timestamp.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_blend_buffers):
	  Don't call gst_object_sync_values() unless we have a valid timestamp.

2008-02-29 06:18:55 +0000  David Schleef <ds@schleef.org>

	  gst/matroska/: Fix Dirac mapping.  I had previously added a VfW-type mapping, but it looks like Dirac will get a nati...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska-mux.c:
	  Fix Dirac mapping.  I had previously added a VfW-type
	  mapping, but it looks like Dirac will get a native Matroska
	  mapping, and this is the most likely method.

2008-02-28 23:56:30 +0000  David Schleef <ds@schleef.org>

	  gst/avi/gstavimux.c: Add Dirac encoding
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: Add Dirac encoding

2008-02-28 11:51:24 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/udp/gstudpsrc.*: Port to GstPoll. See #505417.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt <pkj at axis com>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create),
	  (gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_unlock),
	  (gst_udpsrc_unlock_stop), (gst_udpsrc_stop):
	  * gst/udp/gstudpsrc.h:
	  Port to GstPoll. See #505417.

2008-02-28 08:37:44 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/law/mulaw-decode.c: Return GST_FLOW_NOT_NEGOTIATED when the caps are not set yet on the srcpad. We need rate and ...
	  Original commit message from CVS:
	  * gst/law/mulaw-decode.c: (gst_mulawdec_chain):
	  Return GST_FLOW_NOT_NEGOTIATED when the caps are not set
	  yet on the srcpad. We need rate and channels before we
	  can do any processing. Fixes bug #519088.

2008-02-26 10:09:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Detect and indicate if GCC inline assembly syntax is available.
	  Original commit message from CVS:
	  * configure.ac:
	  Detect and indicate if GCC inline assembly syntax is
	  available.
	  * gst/goom/Makefile.am:
	  * gst/goom/convolve_fx.c:
	  * gst/goom/flying_stars_fx.c:
	  * gst/goom/goom_config.h:
	  * gst/goom/goom_core.c:
	  * gst/goom/goomsl.c:
	  * gst/goom/ifs.c:
	  * gst/goom/mmx.c:
	  * gst/goom/plugin_info.c:
	  * gst/goom/xmmx.c:
	  Fix various GCC-isms, and only build the inline assembly
	  with compilers that support GCC inline assembly.
	  Fix a couple of other warnings shown with Forte.

2008-02-26 05:36:17 +0000  Wouter Cloetens <wouter@mind.be>

	  Add support for specifying a list of cookies to be passed in the HTTP request. Fixes bug #518722.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init),
	  (gst_soup_http_src_init), (gst_soup_http_src_dispose),
	  (gst_soup_http_src_set_property), (gst_soup_http_src_get_property),
	  (gst_soup_http_src_create):
	  * ext/soup/gstsouphttpsrc.h:
	  * tests/check/elements/souphttpsrc.c: (run_test), (GST_START_TEST),
	  (souphttpsrc_suite):
	  Add support for specifying a list of cookies to be passed in
	  the HTTP request. Fixes bug #518722.

2008-02-25 12:03:46 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/goom/xmmx.c: Use 'emms' instead of 'femms' to not crash on cpus that do not implement this 3dnow specific instruc...
	  Original commit message from CVS:
	  * gst/goom/xmmx.c:
	  Use 'emms' instead of 'femms' to not crash on cpus that do not
	  implement this 3dnow specific instruction.

2008-02-25 10:32:35 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/goom/plugin_info.c: Use extended MMX for draw_line() too if available, not only normal MMX.
	  Original commit message from CVS:
	  * gst/goom/plugin_info.c: (setOptimizedMethods):
	  Use extended MMX for draw_line() too if available, not only
	  normal MMX.

2008-02-25 06:50:31 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/jpeg/gstjpeg.c: Remove (commented out) smoke typefinder. This is in base now.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpeg.c: (plugin_init):
	  Remove (commented out) smoke typefinder. This is in base now.

2008-02-23 15:02:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/goom2k1/: Rename the installed library, and don't register the same
	  Original commit message from CVS:
	  * gst/goom2k1/Makefile.am:
	  * gst/goom2k1/gstgoom.c:
	  Rename the installed library, and don't register the same
	  GType name as the new goom.

2008-02-23 12:23:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Check for and define ERROR_CXXFLAGS and use them when building
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/taglib/Makefile.am:
	  Check for and define ERROR_CXXFLAGS and use them when building
	  C++ code (#516509).

2008-02-23 12:10:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/: Call oil_init(), otherwise oil_get_cpu_flags() won't return anything useful. Export goom debug category so...
	  Original commit message from CVS:
	  * gst/goom/gstgoom.c: (goom_debug), (plugin_init):
	  * gst/goom/plugin_info.c: (goom_debug), (GST_CAT_DEFAULT),
	  (setOptimizedMethods):
	  Call oil_init(), otherwise oil_get_cpu_flags() won't return
	  anything useful. Export goom debug category so we can get
	  rid of the VERBOSE define and the printfs.

2008-02-23 11:53:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/: Compile fixes for x86-64.
	  Original commit message from CVS:
	  * gst/goom/goomsl_heap.c: (align_it):
	  * gst/goom/plugin_info.c: (setOptimizedMethods):
	  Compile fixes for x86-64.

2008-02-23 03:10:55 +0000  Bastien Nocera <hadess@hadess.net>

	  gst/goom/Makefile.am: Don't compile lex or yacc outputs with warnings, but add other CFLAGS
	  Original commit message from CVS:
	  * gst/goom/Makefile.am: Don't compile lex or yacc outputs
	  with warnings, but add other CFLAGS
	  * gst/goom/goomsl.c (gsl_instr_set_namespace),
	  (gsl_instr_add_param), (iflow_execute), (gsl_enternamespace),
	  (calculate_labels), (gsl_read_file):
	  * gst/goom/goomsl_lex.l:
	  * gst/goom/goomsl_yacc.y:
	  * gst/goom/plugin_info.c: Remove a few live printf, and
	  fprintf, replace exit() calls with g_assert_not_reached()
	  if it not optimal for a library

2008-02-23 02:38:03 +0000  Bastien Nocera <hadess@hadess.net>

	  gst/goom/Makefile.am: Remove the warnings being disabled, fix linkage on x86, spotted by Sebastian Dröge
	  Original commit message from CVS:
	  * gst/goom/Makefile.am: Remove the warnings being disabled,
	  fix linkage on x86, spotted by Sebastian Dröge
	  <slomo@circular-chaos.org>
	  * gst/goom/convolve_fx.c (convolve_init),
	  (create_output_with_brightness), (convolve_apply):
	  * gst/goom/filters.c (zoomFilterVisualFXWrapper_create):
	  * gst/goom/goomsl.c:
	  * gst/goom/ifs.c (ifs_update), (ifs_visualfx_create):
	  * gst/goom/plugin_info.c:
	  * gst/goom/tentacle3d.c (tentacle_fx_create):
	  Fix warnings, and disable the motifs in the convolve_fx
	  plugin (they were causing warnings, and they were just
	  "Goom" in funny letterring)

2008-02-23 01:51:37 +0000  Bastien Nocera <hadess@hadess.net>

	  configure.ac: Add checks for Flex/Yacc/Bison and other furry animals, for the new goom 2k4 based plugin
	  Original commit message from CVS:
	  2008-02-23  Bastien Nocera  <hadess@hadess.net>
	  * configure.ac: Add checks for Flex/Yacc/Bison and other
	  furry animals, for the new goom 2k4 based plugin
	  * gst/goom/*: Update to use goom 2k4, uses liboil to detect
	  CPU optimisations (not working yet), move the old plugin to...
	  * gst/goom2k1/*: ... here, in case somebody is sick enough
	  Fixes #515073

2008-02-22 14:55:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.c: Fix broken GST_ELEMENT_ERROR macro, fixes compile with the Sun
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_setcaps):
	  Fix broken GST_ELEMENT_ERROR macro, fixes compile with the Sun
	  Workshop 12 compiler, but probably also crashes (#517985).

2008-02-22 09:56:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Post the server response code in an error message instead of a generic 'error' message. Fixes ...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	  Post the server response code in an error message instead of a generic
	  'error' message. Fixes #517237.

2008-02-22 07:20:03 +0000  Wouter Cloetens <wouter@mind.be>

	  Implement zero-copy and make the buffer size configurable.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * configure.ac:
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_cancel_message),
	  (gst_soup_http_src_finished_cb), (gst_soup_http_src_chunk_free),
	  (gst_soup_http_src_chunk_allocator),
	  (gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_create),
	  (gst_soup_http_src_start), (gst_soup_http_src_set_proxy):
	  * ext/soup/gstsouphttpsrc.h:
	  Implement zero-copy and make the buffer size configurable.
	  Prefix proxy URIs with "http://" if they don't start with it
	  already and catch errors earlier, fixes hanging in some situations.
	  Fixes bug #514948.

2008-02-22 06:22:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/Makefile.am: Ignore gconfaudiosrc for the states unit test too. It will fallback to alsasrc if the gconf ...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Ignore gconfaudiosrc for the states unit test too. It will fallback
	  to alsasrc if the gconf settings can't be read and not everybody has
	  alsa.

2008-02-22 06:06:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.*: Always report the duration if we know it in push mode and don't return 0 just to make ...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query),
	  (gst_wavpack_parse_create_src_pad):
	  * ext/wavpack/gstwavpackparse.h:
	  Always report the duration if we know it in push mode and don't
	  return 0 just to make totem believe we can't seek in push mode.
	  Newer totem version use the SEEKING query which properly reports
	  if we can seek or not.

2008-02-22 05:39:01 +0000  Jens Granseuer <jensgr@gmx.net>

	  tests/examples/equalizer/demo.c: C89 fix, moving variable declarations to the beginning of the block. Fixes bug #517933.
	  Original commit message from CVS:
	  Patch by: Jens Granseuer <jensgr at gmx dot net>
	  * tests/examples/equalizer/demo.c: (main):
	  C89 fix, moving variable declarations to the beginning of
	  the block. Fixes bug #517933.

2008-02-21 23:47:37 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to development...
	  Original commit message from CVS:
	  * configure.ac:
	  Back to development...

=== release 0.10.7 ===

2008-02-21 00:09:07 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gamma.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* gst-plugins-good.doap:
	* po/LINGUAS:
	* win32/common/config.h:
	  Release 0.10.7 - Red Door Black
	  Original commit message from CVS:
	  Release 0.10.7 - Red Door Black

2008-02-20 22:51:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/hu.po:
	* po/it.po:
	* po/ja.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2008-02-19 10:47:20 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/alpha/Makefile.am: Link alpha plugin with libgstbase. Fixes bug #517386.
	  Original commit message from CVS:
	  * gst/alpha/Makefile.am:
	  Link alpha plugin with libgstbase. Fixes bug #517386.

2008-02-18 11:13:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Init values to -1 instead of the default 0 value.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream):
	  Init values to -1 instead of the default 0 value.
	  Fixes #516524.

2008-02-14 14:50:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/examples/spectrum/spectrum-example.c: Add missing include to fix compilation when libxml usage is disabled.
	  Original commit message from CVS:
	  * tests/examples/spectrum/spectrum-example.c:
	  Add missing include to fix compilation when libxml usage is disabled.
	  Fixes: #516371

2008-02-12 23:38:19 +0000  Wim Taymans <wim.taymans@collabora.co.uk>

	  fixes: #514889
	  Original commit message from CVS:
	  patch by:  Wim Taymans  <wim.taymans@collabora.co.uk>
	  fixes: #514889
	  * gst/rtp/gstrtph264pay.c:
	  * gst/rtp/gstrtpmp4gdepay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmp4gpay.h:
	  * gst/rtp/gstrtptheorapay.c:
	  * gst/rtp/gstrtpvorbispay.c:
	  Fix various leaks shown up in valgrind
	  - free sprops and buffer in error cases in H264 payloader
	  - fix leak in mp4g depayloader when construction the caps
	  - don't leak config string in the mp4g payloader
	  - don't leak buffers and headers in theora and vorbis payloaders
	  * tests/check/elements/rtp-payloading.c:
	  Fix the RTP data test
	  - Actually send valid amr data to the payloader instead of 20
	  zero-bytes
	  - The mp4g payloader expects codec_data on the caps

2008-02-12 21:36:40 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/MANIFEST: Add libgstpng.dsp to MANIFEST.
	  Original commit message from CVS:
	  * win32/MANIFEST:
	  Add libgstpng.dsp to MANIFEST.
	  * win32/vs6/libgstaudiofx.dsp:
	  Add new source files to VS project file.

2008-02-12 13:34:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/ximage/gstximagesrc.c: Initialise variables when opening the X display rather than in _start(), as the display ca...
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c:
	  Initialise variables when opening the X display rather
	  than in _start(), as the display can be opened before that.
	  Fixes: #515985

2008-02-12 12:22:48 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  sys/directdraw/gstdirectdrawsink.c: Properly chain up finalize functions. Fixes bug #515980.
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c:
	  (gst_ddrawsurface_class_init), (gst_ddrawsurface_finalize),
	  (gst_directdraw_sink_finalize):
	  Properly chain up finalize functions. Fixes bug #515980.

2008-02-12 11:38:54 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  sys/v4l2/v4l2src_calls.c: Chain up the finalize functions. Fixes bug #515984.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize),
	  (gst_v4l2_buffer_class_init), (gst_v4l2_buffer_pool_finalize),
	  (gst_v4l2_buffer_pool_class_init):
	  Chain up the finalize functions. Fixes bug #515984.

2008-02-12 11:14:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  sys/ximage/ximageutil.c: Chain up in the finalize function for our custom buffer sub-class.
	  Original commit message from CVS:
	  * sys/ximage/ximageutil.c:
	  Chain up in the finalize function for our custom
	  buffer sub-class.
	  Patch by: Sebastian Dröge  <slomo@circular-chaos.org>
	  Fixes: #515706

2008-02-12 11:12:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/debug/efence.c: Properly chain up finalize method. Fixes bug #515979.
	  Original commit message from CVS:
	  * gst/debug/efence.c: (gst_fenced_buffer_finalize),
	  (gst_fenced_buffer_class_init):
	  Properly chain up finalize method. Fixes bug #515979.

2008-02-12 11:09:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/ximage/gstximagesrc.c: Free allocated Damage memory before closing our connection to the
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c:
	  Free allocated Damage memory before closing our connection to the
	  X server. Fixes: #515706

2008-02-12 05:21:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/souphttpsrc.c: Include glib/gprintf.h for g_vasprintf(). Fixes bug #515564.
	  Original commit message from CVS:
	  * tests/check/elements/souphttpsrc.c:
	  Include glib/gprintf.h for g_vasprintf(). Fixes bug #515564.

2008-02-12 05:14:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add a few libjpeg suppressions and initialize a variable to make smokeenc valgrind clean. Fixes bug #515701.
	  Original commit message from CVS:
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
	  * tests/check/Makefile.am:
	  * tests/check/gst-plugins-good.supp:
	  Add a few libjpeg suppressions and initialize a variable to
	  make smokeenc valgrind clean. Fixes bug #515701.

2008-02-11 21:24:30 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/avi/gstavidemux.c: Revert patch which sends timestamps only on keyframes, as it breaks playback with current gst-...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Revert patch which sends timestamps only on keyframes, as it
	  breaks playback with current gst-ffmpeg.
	  Fixes: #515562

2008-02-11 14:01:52 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Close some memory leaks spotted by the unit test. Fixes bug #515697.
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
	  * tests/check/elements/multifile.c: (GST_START_TEST):
	  Close some memory leaks spotted by the unit test. Fixes bug #515697.

2008-02-11 13:48:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/gconf/gconf.c: Use and unset the GError when pipeline creation fails instead of simply leaking it. Fixes bug #515...
	  Original commit message from CVS:
	  * ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	  Use and unset the GError when pipeline creation fails instead of
	  simply leaking it. Fixes bug #515704.

2008-02-11 09:13:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/lame/gstlame.c: Don't leak the allowed caps.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_setup):
	  Don't leak the allowed caps.
	  * tests/check/pipelines/lame.c: (GST_START_TEST):
	  Stop leaking all buffers. Fixes bug #515575.

2008-02-10 10:46:13 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Fix long description of audiofx elements. Fixes bug #515457.
	  Original commit message from CVS:
	  * gst/audiofx/audioamplify.c:
	  * gst/audiofx/audiochebband.c:
	  * gst/audiofx/audiocheblimit.c:
	  * gst/audiofx/audiodynamic.c:
	  * gst/audiofx/audioinvert.c:
	  * gst/audiofx/audiopanorama.c:
	  * gst/audiofx/audiowsincband.c:
	  * gst/audiofx/audiowsinclimit.c:
	  Fix long description of audiofx elements. Fixes bug #515457.

2008-02-09 01:45:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Add a simple example application for the spectrum element, include it in the docs, and fix some documentation ambigui...
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * gst/spectrum/gstspectrum.c:
	  * tests/examples/spectrum/.cvsignore:
	  * tests/examples/spectrum/Makefile.am:
	  * tests/examples/spectrum/spectrum-example.c:
	  Add a simple example application for the spectrum element, include it
	  in the docs, and fix some documentation ambiguities.
	  Fixes: #348085

2008-02-09 00:15:25 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/: Fix includes order
	  Original commit message from CVS:
	  * gst/equalizer/Makefile.am:
	  * gst/spectrum/Makefile.am:
	  Fix includes order
	  * tests/check/Makefile.am:
	  Exclude v4l2src from the states test - it takes too long to start.
	  * tests/check/elements/spectrum.c:
	  Make the test run properly with CK_FORK=no

2008-02-08 15:32:36 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add 3 new plugins to spec file
	  Original commit message from CVS:
	  add 3 new plugins to spec file

2008-02-08 15:27:51 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	* gst/audiofx/Makefile.am:
	  add missing header files for disting
	  Original commit message from CVS:
	  add missing header files for disting

2008-02-08 15:20:31 +0000  Julien Moutte <julien@moutte.net>

	  gst/matroska/matroska-demux.c: Flag keyframe and delta units correctly when dealign with a
	  Original commit message from CVS:
	  2008-02-08  Julien Moutte  <julien@fluendo.com>
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock): Flag
	  keyframe and delta units correctly when dealign with a
	  BlockGroup.
	  Fixes: #514397

2008-02-08 10:19:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/elements/.cvsignore: Spell the new tests correctly in .cvsignore
	  Original commit message from CVS:
	  * tests/check/elements/.cvsignore:
	  Spell the new tests correctly in .cvsignore

2008-02-08 10:09:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/multifile/gstmultifilesrc.c: Need to use gsize here for the size, fixes compiler warning.
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
	  Need to use gsize here for the size, fixes compiler warning.
	  * tests/examples/equalizer/.cvsignore:
	  * tests/examples/equalizer/Makefile.am:
	  * tests/examples/spectrum/.cvsignore:
	  * tests/examples/spectrum/Makefile.am:
	  Add missing files to fix the build.

2008-02-08 04:25:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move multifile plugin from -bad.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-multifile.xml:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  Move multifile plugin from -bad.
	  Fixes: #490283

2008-02-08 03:44:12 +0000  David Schleef <ds@schleef.org>

	  gst/multifile/: Use g_file_[sg]et_contents() instead of using stdio functions.
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesink.c:
	  * gst/multifile/gstmultifilesrc.c:
	  Use g_file_[sg]et_contents() instead of using stdio functions.
	  Should be less error prone.
	  * tests/check/elements/multifile.c:
	  Create a temporary directory using standard functions instead of
	  creating a directory in the current dir.

2008-02-08 03:28:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move spectrum plugin from -bad.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * gst/spectrum/Makefile.am:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/examples/Makefile.am:
	  Move spectrum plugin from -bad.
	  Move examples into tests/examples/spectrum.

2008-02-08 02:56:12 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Mention bug 415627 fixed with previous commit
	  Original commit message from CVS:
	  Mention bug 415627 fixed with previous commit

2008-02-08 02:49:20 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move the equalizer plugin across from -bad
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.interfaces:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * gst/equalizer/Makefile.am:
	  * tests/check/Makefile.am:
	  * tests/examples/Makefile.am:
	  Move the equalizer plugin across from -bad
	  * tests/check/elements/.cvsignore:
	  Add equalizer, audiosincwband and audiosincwlimit
	  * tests/check/elements/equalizer.c:
	  Fix compiler warnings

2008-02-08 02:48:54 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  docs/plugins/gst-plugins-bad-plugins.*: Remove equalizer plugin docs
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  Remove equalizer plugin docs
	  * tests/check/Makefile.am:
	  Add GST_OPTION_CFLAGS, to get -Werror -Wall into the tests as for
	  other modules.
	  * tests/check/elements/multifile.c:
	  * tests/check/elements/rganalysis.c:
	  * tests/check/elements/rglimiter.c:
	  Fix compiler warnings from -Wall -Werror

2008-02-08 01:07:02 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Only build with DISABLE_DEPRECATED during the CVS cycle. Pre-releases are treated like releases and bui...
	  Original commit message from CVS:
	  * configure.ac:
	  Only build with DISABLE_DEPRECATED during the CVS cycle. Pre-releases
	  are treated like releases and build without it.

2008-02-07 21:57:54 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move the lpwsinc and bpwsinc elements from gst-plugins-bad into the audiofx plugin, and rename to audiowsinclimit and...
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofx.c:
	  * gst/audiofx/audiowsincband.c:
	  * gst/audiofx/audiowsincband.h:
	  * gst/audiofx/audiowsinclimit.c:
	  * gst/audiofx/audiowsinclimit.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/audiowsincband.c:
	  * tests/check/elements/audiowsinclimit.c:
	  Move the lpwsinc and bpwsinc elements from gst-plugins-bad into
	  the audiofx plugin, and rename to audiowsinclimit and audiowsincband
	  respectively.
	  Fixes: #467666

2008-02-07 21:17:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Return GST_FLOW_NOT_NEGOTIATED if we get a buffer without caps, and add a somewhat useful debug message. Plus test.
	  Original commit message from CVS:
	  * gst/icydemux/gsticydemux.c: (gst_icydemux_chain):
	  * tests/check/elements/icydemux.c:
	  Return GST_FLOW_NOT_NEGOTIATED if we get a buffer without
	  caps, and add a somewhat useful debug message. Plus test.

2008-02-07 19:13:56 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/rtsp/gstrtspsrc.c: Include unistd.h only if HAVE_UNISTD_H is defined
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c:
	  Include unistd.h only if HAVE_UNISTD_H is defined
	  * win32/common/config.h.in:
	  * win32/common/config.h:
	  Define socklen_t as it seems it's not defined in default
	  Visual Studio headers.
	  * win32/vs6/libgstalpha.dsp:
	  * win32/vs6/libgstapetag.dsp:
	  * win32/vs6/libgstavi.dsp:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgstrtsp.dsp:
	  * win32/vs6/libgstvideomixer.dsp:
	  Update project file dependencies and add new source files

2008-02-07 16:38:55 +0000  Bjarne Rosengren <bjarne@axis.com>

	  gst/matroska/ebml-write.c: Don't leak buffers when we don't push them downstream.
	  Original commit message from CVS:
	  Patch by: Bjarne Rosengren <bjarne at axis dot com>
	  * gst/matroska/ebml-write.c: (gst_ebml_write_element_push):
	  Don't leak buffers when we don't push them downstream.
	  Fixes bug #514965.

2008-02-07 13:48:20 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/multifile/gstmultifilesink.c: Add a fixme comment.
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesink.c:
	  Add a fixme comment.
	  * gst/selector/gstoutputselector.c:
	  Fix same leak as in input-selector.
	  * tests/icles/output-selector-test.c:
	  Improve the test.

2008-02-07 13:41:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.c: Improve the docs.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c:
	  Improve the docs.

2008-02-07 10:17:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump requirements to (good) released versions to avoid confusion and make implicit core requirement exp...
	  Original commit message from CVS:
	  * configure.ac:
	  Bump requirements to (good) released versions to avoid
	  confusion and make implicit core requirement explicit.

2008-02-07 10:04:01 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstlpwsinc.c: Fix typo in the long description of the element.
	  Original commit message from CVS:
	  * gst/filter/gstlpwsinc.c:
	  Fix typo in the long description of the element.

2008-02-06 23:44:43 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Rename audiochebyshevfreqband -> audiochebband and audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS...
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiochebband.c:
	  * gst/audiofx/audiochebband.h:
	  * gst/audiofx/audiocheblimit.c:
	  * gst/audiofx/audiocheblimit.h:
	  * gst/audiofx/audiochebyshevfreqband.c:
	  * gst/audiofx/audiochebyshevfreqband.h:
	  * gst/audiofx/audiochebyshevfreqlimit.c:
	  * gst/audiofx/audiochebyshevfreqlimit.h:
	  * gst/audiofx/audiofx.c:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/audiochebband.c:
	  * tests/check/elements/audiocheblimit.c:
	  * tests/check/elements/audiochebyshevfreqband.c:
	  * tests/check/elements/audiochebyshevfreqlimit.c:
	  Rename audiochebyshevfreqband -> audiochebband and
	  audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS
	  surgery.
	  Closes: #491811

2008-02-06 11:07:47 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.c: Fix memory leak and improve debugging a bit.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_chunk_cb),
	  (gst_soup_http_src_create):
	  Fix memory leak and improve debugging a bit.

2008-02-05 17:59:24 +0000  orjan <orjanf@axis.com>

	  gst/multipart/multipartmux.c: Fix caps memory leak. Fixes #514573.
	  Original commit message from CVS:
	  Patch by: orjan <orjanf at axis dot com>
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	  Fix caps memory leak. Fixes #514573.

2008-02-04 12:07:14 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: If there's no entries in the subindex, don't try to do anything stupid, just return.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex):
	  If there's no entries in the subindex, don't try to do anything stupid,
	  just return.

2008-02-02 19:47:50 +0000  John Millikin <jmillikin@gmail.com>

	  ext/flac/gstflacdec.c: Fix extraction of picture blocks with newer libflac versions again:
	  Original commit message from CVS:
	  Patch by: John Millikin <jmillikin at gmail dot com>
	  * ext/flac/gstflacdec.c: (gst_flac_dec_scan_for_last_block),
	  (gst_flac_extract_picture_buffer), (gst_flac_dec_metadata_callback):
	  Fix extraction of picture blocks with newer libflac versions again:
	  FLAC__METADATA_TYPE_PICTURE is an enum, not a define (#513628).

2008-02-02 18:06:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Add rtp-payloading test to VALGRIND_TO_FIX.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Add rtp-payloading test to VALGRIND_TO_FIX.
	  * tests/check/elements/rtp-payloading.c:
	  Add semicolons after GST_TEST_END so gst-indent gets the
	  formatting right; make test less verbose in general, but
	  more verbose in the error case (which should probably
	  make the test fail anyway).

2008-02-01 18:29:21 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  Add documentation for avisubtitle and change class to
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * gst/avi/gstavisubtitle.c:
	  Add documentation for avisubtitle and change class to
	  Codec/Parser/Subtitle

2008-01-31 16:12:28 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/v4l2/v4l2_calls.c: Treat ENOTTY (driver does not implement ioctl) the same as
	  Original commit message from CVS:
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  Treat ENOTTY (driver does not implement ioctl) the same as
	  EINVAL since it implies there are no available standards.
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format),
	  (gst_v4l2src_get_nearest_size):
	  Replace gst_v4l2src_get_size_limits with 2 calls to new function
	  gst_v4l2src_get_nearest_size, and get it to use VIDIOC_S_FMT to
	  probe if the driver does not support VIDIOC_TRY_FMT for whatever
	  reason, and if we aren't yet actively capturing.
	  * sys/v4l2/v4l2src_calls.h:
	  Remove replaced function declaration.

2008-01-31 16:03:48 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Bump plugins-base requirement to 0.10.16 for the gst_video_format_*
	  Original commit message from CVS:
	  * configure.ac:
	  Bump plugins-base requirement to 0.10.16 for the gst_video_format_*
	  API.

2008-01-31 09:50:31 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/soup/gstsouphttpsrc.c: Add changes to gstsouphttpsrc.c that were missing from last commit.
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c: (_do_init),
	  (gst_soup_http_src_base_init), (gst_soup_http_src_class_init),
	  (gst_soup_http_src_init), (gst_soup_http_src_dispose),
	  (gst_soup_http_src_set_property), (gst_soup_http_src_get_property),
	  (gst_soup_http_src_unicodify), (gst_soup_http_src_cancel_message),
	  (gst_soup_http_src_queue_message),
	  (gst_soup_http_src_add_range_header),
	  (gst_soup_http_src_session_unpause_message),
	  (gst_soup_http_src_session_pause_message),
	  (gst_soup_http_src_session_close),
	  (gst_soup_http_src_got_headers_cb),
	  (gst_soup_http_src_got_body_cb), (gst_soup_http_src_finished_cb),
	  (gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_response_cb),
	  (gst_soup_http_src_parse_status), (gst_soup_http_src_create),
	  (gst_soup_http_src_start), (gst_soup_http_src_stop),
	  (gst_soup_http_src_unlock), (gst_soup_http_src_unlock_stop),
	  (gst_soup_http_src_get_size), (gst_soup_http_src_is_seekable),
	  (gst_soup_http_src_do_seek), (gst_soup_http_src_set_location),
	  (gst_soup_http_src_set_proxy), (gst_soup_http_src_uri_get_type),
	  (gst_soup_http_src_uri_get_protocols),
	  (gst_soup_http_src_uri_get_uri), (gst_soup_http_src_uri_set_uri),
	  (gst_soup_http_src_uri_handler_init), (plugin_init):
	  Add changes to gstsouphttpsrc.c that were missing from last commit.

2008-01-31 08:57:16 +0000  Wouter Cloetens <wouter@mind.be>

	  Make coding style more consistent, including class renaming.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/inspect/plugin-soup.xml:
	  (gst_soup_http_src_base_init), (gst_soup_http_src_class_init),
	  (gst_soup_http_src_init), (gst_soup_http_src_dispose),
	  (gst_soup_http_src_set_property), (gst_soup_http_src_get_property),
	  (gst_soup_http_src_unicodify), (gst_soup_http_src_cancel_message),
	  (gst_soup_http_src_queue_message),
	  (gst_soup_http_src_add_range_header),
	  (gst_soup_http_src_session_unpause_message),
	  (gst_soup_http_src_session_pause_message),
	  (gst_soup_http_src_session_close),
	  (gst_soup_http_src_got_headers_cb),
	  (gst_soup_http_src_got_body_cb), (gst_soup_http_src_finished_cb),
	  (gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_response_cb),
	  (gst_soup_http_src_parse_status), (gst_soup_http_src_create),
	  (gst_soup_http_src_start), (gst_soup_http_src_stop),
	  (gst_soup_http_src_unlock), (gst_soup_http_src_unlock_stop),
	  (gst_soup_http_src_get_size), (gst_soup_http_src_is_seekable),
	  (gst_soup_http_src_do_seek), (gst_soup_http_src_set_location),
	  (gst_soup_http_src_set_proxy), (gst_soup_http_src_uri_get_type),
	  (gst_soup_http_src_uri_get_protocols),
	  (gst_soup_http_src_uri_get_uri), (gst_soup_http_src_uri_set_uri),
	  (gst_soup_http_src_uri_handler_init), (plugin_init):
	  * ext/soup/gstsouphttpsrc.h:
	  Make coding style more consistent, including class renaming.

2008-01-31 00:03:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Fix typo.
	  Original commit message from CVS:
	  * configure.ac:
	  Fix typo.

2008-01-31 00:00:23 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/alpha/: Re-write the 'alpha' plugin to be BaseTransform based, simplifying some stuff, and making buffer-alloc an...
	  Original commit message from CVS:
	  * gst/alpha/Makefile.am:
	  * gst/alpha/gstalpha.c:
	  Re-write the 'alpha' plugin to be BaseTransform based, simplifying
	  some stuff, and making buffer-alloc and resizing work automatically.
	  No longer crashes on odd frame widths and heights, although there
	  seems to be a disagreement with ffmpegcolorspace about what size
	  an AYUV frame with odd height should be.

2008-01-30 15:40:36 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.c: Update documentation a bit.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c:
	  Update documentation a bit.
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-alsaspdif.xml:
	  * docs/plugins/inspect/plugin-dvb.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-glimagesink.xml:
	  * docs/plugins/inspect/plugin-mpegvideoparse.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-rawparse.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-sdl.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-soup.xml:
	  * docs/plugins/inspect/plugin-spcdec.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-speexresample.xml:
	  * docs/plugins/inspect/plugin-switch.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  Regenerate everything for the documentation changes we had.

2008-01-30 13:29:15 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.c: Let the proxy property default to the content of the $http_proxy environment variable.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_init):
	  Let the proxy property default to the content of the $http_proxy
	  environment variable.

2008-01-30 13:08:45 +0000  Wouter Cloetens <wouter@mind.be>

	  tests/check/: Add missing files for the unit test.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * tests/check/test-cert.pem:
	  * tests/check/test-key.pem:
	  Add missing files for the unit test.

2008-01-30 13:06:01 +0000  Wouter Cloetens <wouter@mind.be>

	  docs/plugins/: Add souphttpsrc to the docs.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  Add souphttpsrc to the docs.
	  * configure.ac:
	  * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init),
	  (gst_souphttp_src_init), (gst_souphttp_src_dispose),
	  (gst_souphttp_src_set_property), (gst_souphttp_src_get_property),
	  (gst_souphttp_src_cancel_message),
	  (gst_souphttp_src_queue_message),
	  (gst_souphttp_src_add_range_header),
	  (gst_souphttp_src_session_unpause_message),
	  (gst_souphttp_src_session_pause_message),
	  (gst_souphttp_src_session_close),
	  (gst_souphttp_src_got_headers_cb), (gst_souphttp_src_got_body_cb),
	  (gst_souphttp_src_finished_cb), (gst_souphttp_src_got_chunk_cb),
	  (gst_souphttp_src_response_cb), (gst_souphttp_src_parse_status),
	  (gst_souphttp_src_create), (gst_souphttp_src_start),
	  (gst_souphttp_src_stop), (gst_souphttp_src_unlock),
	  (gst_souphttp_src_unlock_stop), (gst_souphttp_src_get_size),
	  (gst_souphttp_src_is_seekable), (gst_souphttp_src_do_seek),
	  (gst_souphttp_src_set_location), (gst_souphttp_src_set_proxy),
	  (plugin_init):
	  * ext/soup/gstsouphttpsrc.h:
	  Add support for libsoup2.4 and require it. Also implement redirection
	  and manual proxy specification. Fixes bug #510708.
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/souphttpsrc.c:
	  Add unit test for souphttpsrc.

2008-01-29 18:43:32 +0000  Alessandro Decina <alessandro@nnva.org>

	  ext/libpng/gstpngenc.*: Preallocate the output buffer so that g_memdup() and gst_buffer_merge() aren't needed anymore...
	  Original commit message from CVS:
	  Patch by: Alessandro Decina <alessandro at nnva dot org>
	  * ext/libpng/gstpngenc.c: (user_write_data), (gst_pngenc_chain):
	  * ext/libpng/gstpngenc.h:
	  Preallocate the output buffer so that g_memdup() and
	  gst_buffer_merge() aren't needed anymore. This greatly improves
	  performances and fixes #512544.

2008-01-29 18:24:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: GStreamer timestamps are PTS values while AVI only knows about DTS timestamps. Make sure we on...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry),
	  (gst_avi_demux_stream_data):
	  GStreamer timestamps are PTS values while AVI only knows about DTS
	  timestamps. Make sure we only copy the DTS as the buffer timestamp when
	  we are dealing with a key frame.

2008-01-29 15:45:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/: Add add testsuite for the rtp-payloader that tries simulating dataflow. Needs more test data.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/rtp-payloading.c:
	  Add add testsuite for the rtp-payloader that tries simulating
	  dataflow. Needs more test data.

2008-01-29 15:27:02 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/alphacolor.c: Remove two unused variables.
	  Original commit message from CVS:
	  * tests/check/elements/alphacolor.c:
	  Remove two unused variables.

2008-01-28 12:17:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtsp/gstrtspsrc.c: Use g_ascii_strtoll() instead of atoll, which is only available in C99.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo):
	  Use g_ascii_strtoll() instead of atoll, which is only
	  available in C99.

2008-01-26 16:19:26 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: Don't implement get_unit_size() ourselves, the GstAudioFilter base class already does this for us.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
	  Don't implement get_unit_size() ourselves, the GstAudioFilter base
	  class already does this for us.

2008-01-25 10:53:17 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/rtp/: Add MPEG2 video payloader
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c:
	  * gst/rtp/gstrtpmpvpay.c:
	  * gst/rtp/gstrtpmpvpay.h:
	  Add MPEG2 video payloader

2008-01-23 17:05:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/level/gstlevel.c: Use #include <math.h> instead of #include "math.h".
	  Original commit message from CVS:
	  * gst/level/gstlevel.c:
	  Use #include <math.h> instead of #include "math.h".

2008-01-21 19:41:45 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Fix up some CFLAGS sets.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Fix up some CFLAGS sets.
	  Don't include gconfvideosrc in the states test.
	  * tests/check/elements/autodetect.c: (GST_START_TEST):
	  Add some error strings to fail_unless arguments to fix some weird
	  compiler errors on Solaris.

2008-01-21 19:35:58 +0000  Brian Cameron <brian.cameron@sun.com>

	  configure.ac: Detect video4linux headers on Solaris too.
	  Original commit message from CVS:
	  * configure.ac:
	  Detect video4linux headers on Solaris too.
	  * sys/v4l2/gstv4l2colorbalance.h:
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/v4l2_calls.c:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize),
	  (gst_v4l2_buffer_new):
	  Make v4l2 build on Solaris.
	  Patch by: Brian Cameron  <brian.cameron at sun dot com>
	  Fixes: #510505

2008-01-21 11:46:19 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/gst-plugins-good-plugins-docs.sgml: Update list from (still local) scanning script.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  Update list from (still local) scanning script.

2008-01-21 09:57:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: Add symbols from -unused.txt to the right place.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  Add symbols from -unused.txt to the right place.
	  * gst/dvdspu/gstdvdspu.c:
	  * gst/dvdspu/gstdvdspu.h:
	  Coherent namespace usage.
	  * gst/spectrum/gstspectrum.c:
	  Fix broken XML fragment in doc snippet even more.

2008-01-21 07:54:02 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/Makefile.am: Update include list.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  Update include list.
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  Update xml includes.
	  * docs/plugins/inspect/plugin-alsaspdif.xml:
	  * docs/plugins/inspect/plugin-amrwb.xml:
	  * docs/plugins/inspect/plugin-bayer.xml:
	  * docs/plugins/inspect/plugin-bz2.xml:
	  * docs/plugins/inspect/plugin-cdxaparse.xml:
	  * docs/plugins/inspect/plugin-dtsdec.xml:
	  * docs/plugins/inspect/plugin-dvbsrc.xml:
	  * docs/plugins/inspect/plugin-dvdspu.xml:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * docs/plugins/inspect/plugin-faac.xml:
	  * docs/plugins/inspect/plugin-faad.xml:
	  * docs/plugins/inspect/plugin-fbdevsink.xml:
	  * docs/plugins/inspect/plugin-festival.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-flvdemux.xml:
	  * docs/plugins/inspect/plugin-freeze.xml:
	  * docs/plugins/inspect/plugin-gsm.xml:
	  * docs/plugins/inspect/plugin-gstinterlace.xml:
	  * docs/plugins/inspect/plugin-gstrtpmanager.xml:
	  * docs/plugins/inspect/plugin-h264parse.xml:
	  * docs/plugins/inspect/plugin-interleave.xml:
	  * docs/plugins/inspect/plugin-ladspa.xml:
	  * docs/plugins/inspect/plugin-metadata.xml:
	  * docs/plugins/inspect/plugin-modplug.xml:
	  * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
	  * docs/plugins/inspect/plugin-mpegtsparse.xml:
	  * docs/plugins/inspect/plugin-mpegvideoparse.xml:
	  * docs/plugins/inspect/plugin-musicbrainz.xml:
	  * docs/plugins/inspect/plugin-mve.xml:
	  * docs/plugins/inspect/plugin-nsfdec.xml:
	  * docs/plugins/inspect/plugin-nuvdemux.xml:
	  * docs/plugins/inspect/plugin-qtdemux.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-real.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-sdl.xml:
	  * docs/plugins/inspect/plugin-sdp.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-speexresample.xml:
	  * docs/plugins/inspect/plugin-stereo.xml:
	  * docs/plugins/inspect/plugin-switch.xml:
	  * docs/plugins/inspect/plugin-timidity.xml:
	  * docs/plugins/inspect/plugin-tta.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-videoparse.xml:
	  * docs/plugins/inspect/plugin-videosignal.xml:
	  * docs/plugins/inspect/plugin-vmnc.xml:
	  * docs/plugins/inspect/plugin-wildmidi.xml:
	  * docs/plugins/inspect/plugin-x264.xml:
	  * docs/plugins/inspect/plugin-xingheader.xml:
	  * docs/plugins/inspect/plugin-xvid.xml:
	  * docs/plugins/inspect/plugin-y4menc.xml:
	  Regenerate files.
	  * gst/spectrum/gstspectrum.c:
	  Fix broken XML fragment in doc snippet.
	  * tests/check/elements/.cvsignore:
	  Add test binary to ignores.

2008-01-20 05:07:52 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.c: Report the size of the stream as the total size instead of the remaining Content-Length, w...
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c: (soup_got_headers):
	  Report the size of the stream as the total size instead of
	  the remaining Content-Length, which is wrong after a seek.

2008-01-19 14:59:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	* ChangeLog:
	  Add bug number to the latest entry
	  Original commit message from CVS:
	  Add bug number to the latest entry

2008-01-19 14:53:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavparse/gstwavparse.c: Set variable to NULL after freeing it to prevent double frees or make failures by another...
	  Original commit message from CVS:
	  Based on a patch by:
	  Victor STINNER <victor dot stinner at haypocalc dot com>
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Set variable to NULL after freeing it to prevent double frees
	  or make failures by another use of it afterwards more obvious
	  and fix use of it after the freeing.

2008-01-19 14:34:50 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.c: Correctly set duration on the GstBaseSrc segment when we know it to fix failing the durati...
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c: (soup_got_headers):
	  Correctly set duration on the GstBaseSrc segment when we know it
	  to fix failing the duration query.

2008-01-18 13:40:38 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/udp/gstmultiudpsink.c: use GST_WARNING for logging
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c:
	  use GST_WARNING for logging

2008-01-18 10:05:53 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/multifile/gstmultifilesrc.c: Fix memory leak spotted by the unit test.
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
	  Fix memory leak spotted by the unit test.

2008-01-18 10:04:25 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/udp/gstmultiudpsink.c: Don't try to leave a multicast group with an invalid socket
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c:
	  Don't try to leave a multicast group with an invalid socket

2008-01-18 08:49:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/: Add some minimal tests for the equalizer plugin.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/equalizer.c: (setup_equalizer),
	  (cleanup_equalizer), (GST_START_TEST), (equalizer_suite), (main):
	  Add some minimal tests for the equalizer plugin.

2008-01-18 07:03:23 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.c: Unparent all bands from the equalizer when finalizing to stop leaking	them.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_finalize):
	  Unparent all bands from the equalizer when finalizing to stop
	  leaking	them.

2008-01-18 05:32:26 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/soup/gstsouphttpsrc.c: Add support for WebDAV.
	  Original commit message from CVS:
	  * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_uri_get_protocols):
	  Add support for WebDAV.

2008-01-18 05:24:39 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.*: Add support for seeking to souphttpsrc. Fixes bug #502335.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init),
	  (gst_souphttp_src_init), (gst_souphttp_src_create),
	  (gst_souphttp_src_is_seekable), (gst_souphttp_src_do_seek),
	  (soup_add_range_header), (soup_got_headers), (soup_got_chunk):
	  * ext/soup/gstsouphttpsrc.h:
	  Add support for seeking to souphttpsrc. Fixes bug #502335.

2008-01-17 21:23:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.c: where the picture metadata defines and structs don't exist yet.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c:
	  Fix compilation against flac 1.1.2 (as on debian stable), where
	  the picture metadata defines and structs don't exist yet.
	  Fixes #509301.

2008-01-17 17:26:48 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.*: Fix the case where you initially have stereo input, and so lame's mode is not set to mono, and th...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c:
	  * ext/lame/gstlame.h:
	  Fix the case where you initially have stereo input, and so lame's
	  mode is not set to mono, and then you get input with mono audio and
	  soon after you get stereo input again. What happened before this
	  commit is that it would keep the encoding mode as mono. It should
	  change it back to the one requested by the app (or the default one)
	  if not requested.

2008-01-17 11:13:16 +0000  Olivier Crete <tester@tester.ca>

	  gst/udp/gstmultiudpsink.*: Add property to automatically join a multicast group or not. This can be useful when shari...
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	  (gst_multiudpsink_init), (gst_multiudpsink_set_property),
	  (gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
	  (gst_multiudpsink_add_internal), (gst_multiudpsink_remove):
	  * gst/udp/gstmultiudpsink.h:
	  Add property to automatically join a multicast group or not. This can be
	  useful when sharing a socket between multiple elements.
	  Fixes #509531.

2008-01-16 21:53:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/videomixer/Makefile.am: Add controller flags.
	  Original commit message from CVS:
	  * gst/videomixer/Makefile.am:
	  Add controller flags.

2008-01-16 20:17:08 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/videomixer/videomixer.c: Also commit the missing gst_object_sync_values().
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c:
	  Also commit the missing gst_object_sync_values().

2008-01-16 08:11:46 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/Makefile.am: Remove duplicate entry.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  Remove duplicate entry.

2008-01-15 16:52:10 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: Add 3 more plugins to docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-gamma.xml:
	  * docs/plugins/inspect/plugin-monoscope.xml:
	  * docs/plugins/inspect/plugin-video4linux2.xml:
	  Add 3 more plugins to docs.

2008-01-15 16:04:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Revert previous change caused by a file that got stuck on an old revision.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * sys/osxvideo/osxvideosink.h:
	  Revert previous change caused by a file that got stuck on an old
	  revision.

2008-01-15 15:40:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Re-add multipartdemux to the docs. Last round of section cleanup.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * gst/multipart/Makefile.am:
	  * gst/multipart/multipartdemux.c:
	  * gst/multipart/multipartdemux.h:
	  * gst/multipart/multipartmux.c:
	  * gst/multipart/multipartmux.h:
	  Re-add multipartdemux to the docs. Last round of section cleanup.

2008-01-15 15:22:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Managed to resolve most unused declarations. Filed a bug for one left.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * sys/osxaudio/gstosxaudiosink.h:
	  * sys/osxvideo/osxvideosink.h:
	  Managed to resolve most unused declarations. Filed a bug for one left.

2008-01-15 08:03:49 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/gst-plugins-good-plugins-sections.txt: Cleanup section file.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Cleanup section file.

2008-01-15 07:42:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: Update plugin docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.signals:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-ladspa.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-shout2send.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  Update plugin docs.
	  * gst/videomixer/Makefile.am:
	  * gst/videomixer/videomixer.c:
	  * gst/videomixer/videomixer.h:
	  * gst/videomixer/videomixerpad.h:
	  Split out header to fix warnings from the doc-build.

2008-01-14 12:35:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  As found by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
	  Original commit message from CVS:
	  As found by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo):
	  Use atoll to parse the rtptime with enough precision. Fixes #509329.

2008-01-14 12:11:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Initialise variables to work around (false) 'foo might be used uninitialized in this function' warnings by gcc-...
	  Original commit message from CVS:
	  * gst/avi/gstavisubtitle.c: (gst_avi_subtitle_extract_file):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
	  Initialise variables to work around (false) 'foo might be used
	  uninitialized in this function' warnings by gcc-3.3.3 (#509298).

2008-01-14 11:24:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/lame/gstlame.c: Use gst_util_uint64_scale instead of gst_util_uint64_scale_int as 8 * GST_SECOND is too large for...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_event):
	  Use gst_util_uint64_scale instead of gst_util_uint64_scale_int
	  as 8 * GST_SECOND is too large for int.

2008-01-14 09:17:47 +0000  Mark Nauwelaerts <manauw@syknet.be>

	  ext/lame/gstlame.c: Correctly set number of channels when using mono-encoding mode and fix the duration calculation o...
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw at syknet dot be>
	  * ext/lame/gstlame.c: (gst_lame_sink_setcaps),
	  (gst_lame_sink_event):
	  Correctly set number of channels when using mono-encoding mode
	  and fix the duration calculation of the EOS buffer.

2008-01-12 02:32:35 +0000  David Schleef <ds@schleef.org>

	  Ignore more files for the buildbot.
	  Original commit message from CVS:
	  * docs/plugins/.cvsignore:
	  * tests/check/pipelines/.cvsignore:
	  Ignore more files for the buildbot.

2008-01-11 21:08:59 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Generate the image-type values correctly. Leave them out of the caps when outputting a "preview image" tag, since it ...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer):
	  * gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Generate the image-type values correctly. Leave them out of the caps
	  when outputting a "preview image" tag, since it only makes sense
	  to have one of those - the type is irrelevant.
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_open):
	  If we can, mark the mixer multiple open when we use it, in case
	  (for some reason) the process wants to open it again elsewhere.

2008-01-11 19:16:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/: It's "endianness", not "endianess". Fixes unit tests.
	  Original commit message from CVS:
	  * tests/check/elements/rganalysis.c: (test_buffer_const_float_mono),
	  (test_buffer_const_float_stereo), (test_buffer_const_int16_mono),
	  (test_buffer_const_int16_stereo), (test_buffer_square_float_mono),
	  (test_buffer_square_float_stereo), (test_buffer_square_int16_mono),
	  (test_buffer_square_int16_stereo):
	  * tests/check/elements/rglimiter.c: (create_test_buffer):
	  * tests/check/elements/rgvolume.c: (test_buffer_new):
	  It's "endianness", not "endianess". Fixes unit tests.

2008-01-11 18:56:06 +0000  Edward Hervey <bilboed@bilboed.com>

	* tests/check/pipelines/.cvignore:
	  ignore some more
	  Original commit message from CVS:
	  ignore some more

2008-01-11 18:54:31 +0000  Edward Hervey <bilboed@bilboed.com>

	* tests/check/elements/.gitignore:
	  ignore some more
	  Original commit message from CVS:
	  ignore some more

2008-01-11 17:21:30 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/: Fix the clock rate to 90000 as required by the RFC.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps):
	  * gst/rtp/gstrtptheorapay.c:
	  Fix the clock rate to 90000 as required by the RFC.
	  Fixes #508644.

2008-01-11 17:12:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/icydemux.c: Don't use deprecated GST_PLUGIN_DEFINE_STATIC.
	  Original commit message from CVS:
	  * tests/check/elements/icydemux.c: (GST_START_TEST), (icydemux_suite):
	  Don't use deprecated GST_PLUGIN_DEFINE_STATIC.

2008-01-10 12:25:44 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
	  Original commit message from CVS:
	  * autogen.sh:
	  Add -Wno-portability to the automake parameters to stop warnings
	  about GNU make extensions being used. We require GNU make in almost
	  every Makefile anyway.
	  * configure.ac:
	  Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
	  at the same time is required for per target flags.

2008-01-09 15:28:29 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/videomixer/videomixer.c: Fix error from my last commit.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_init):
	  Fix error from my last commit.

2008-01-09 15:20:19 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/id3demux/id3v2frames.c: Make sure the ISO 639-X language code in ID3v2 COMM frames so we don't end up with non-UT...
	  Original commit message from CVS:
	  Based on patch by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
	  * gst/id3demux/id3v2frames.c: (parse_comment_frame):
	  Make sure the ISO 639-X language code in ID3v2 COMM frames
	  is actually valid UTF-8 (or rather: ASCII), so we don't end
	  up with non-UTF8 strings in tags if there's garbage in the
	  language field. Also make sure the language code is always
	  lower case. Fixes: #508291.

2008-01-09 13:55:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ChangeLog: Fix ChangeLog typo.
	  Original commit message from CVS:
	  * ChangeLog:
	  Fix ChangeLog typo.

2008-01-09 13:50:09 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Makefile.am: Include lcov.mak to allow builging coverage reports. Guard check-torture target like in the other packages.
	  Original commit message from CVS:
	  * Makefile.am:
	  Include lcov.mak to allow builging coverage reports. Guard
	  check-torture target like in the other packages.

2008-01-09 12:33:58 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/videomixer/videomixer.c: Implement GstChildProxy interface.
	  Original commit message from CVS:
	  reviewed by: Edward Hervey  <edward.hervey@collabora.co.uk>
	  * gst/videomixer/videomixer.c:
	  (gst_videomixer_set_master_geometry), (_do_init),
	  (gst_videomixer_child_proxy_get_child_by_index),
	  (gst_videomixer_child_proxy_get_children_count),
	  (gst_videomixer_child_proxy_init), (gst_videomixer_reset),
	  (gst_videomixer_init), (gst_videomixer_request_new_pad),
	  (gst_videomixer_release_pad), (gst_videomixer_fill_queues):
	  Implement GstChildProxy interface.
	  Send newsegment at the right moment
	  Fixes #488879

2008-01-09 12:01:14 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/alpha/: Make the various properties of 'alpha' controllable. This allows doing niceties like fade-in/fade-out.
	  Original commit message from CVS:
	  * gst/alpha/Makefile.am:
	  * gst/alpha/gstalpha.c: (gst_alpha_class_init), (gst_alpha_init),
	  (gst_alpha_sink_event), (gst_alpha_chain),
	  (gst_alpha_change_state), (plugin_init):
	  Make the various properties of 'alpha' controllable. This allows doing
	  niceties like fade-in/fade-out.

2008-01-09 11:11:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/rtp/: Remove copy/paste unused code (property setters and getter) found by the coverage suite (yay, saves ~20k on...
	  Original commit message from CVS:
	  * gst/rtp/gstasteriskh263.c:
	  * gst/rtp/gstrtpL16depay.c:
	  * gst/rtp/gstrtpac3depay.c:
	  * gst/rtp/gstrtpamrpay.c:
	  * gst/rtp/gstrtpdepay.c:
	  * gst/rtp/gstrtpgsmdepay.c:
	  * gst/rtp/gstrtph263depay.c:
	  * gst/rtp/gstrtph263pdepay.c:
	  * gst/rtp/gstrtph263ppay.c:
	  * gst/rtp/gstrtph264depay.c:
	  * gst/rtp/gstrtph264pay.c:
	  * gst/rtp/gstrtpmp2tdepay.c:
	  * gst/rtp/gstrtpmp4adepay.c:
	  * gst/rtp/gstrtpmp4gdepay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmp4vdepay.c:
	  * gst/rtp/gstrtpmpadepay.c:
	  * gst/rtp/gstrtpmpvdepay.c:
	  * gst/rtp/gstrtpsv3vdepay.c:
	  * gst/rtp/gstrtptheoradepay.c:
	  * gst/rtp/gstrtptheorapay.c:
	  * gst/rtp/gstrtpvorbisdepay.c:
	  * gst/rtp/gstrtpvorbispay.c:
	  Remove copy/paste unused code (property setters and getter) found by
	  the coverage suite (yay, saves ~20k on disk).

2008-01-08 20:03:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-mux.c: Also fix up pad templates to indicate that image/jpeg doesn't absolutely require the fra...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (COMMON_VIDEO_CAPS_NO_FRAMERATE),
	  (videosink_templ):
	  Also fix up pad templates to indicate that image/jpeg doesn't
	  absolutely require the framerate property to be set (#504081).

2008-01-08 19:57:23 +0000  Wouter Cloetens <wouter@mind.be>

	  gst/matroska/matroska-mux.*: Keep track of first and last timestamps for each incoming stream, so we can calculate th...
	  Original commit message from CVS:
	  Based on patch by: Wouter Cloetens  <wouter at mind be>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps),
	  (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
	  (gst_matroska_mux_finish), (gst_matroska_mux_collected):
	  * gst/matroska/matroska-mux.h:
	  Keep track of first and last timestamps for each incoming stream,
	  so we can calculate the total duration for live sources and other
	  input where we can't query the duration from the start or where
	  there's no constant framerate from which we can deduce the
	  duration; also use calculated/observed duration if it is bigger
	  than the previously queried duration. Furthermore, use
	  gst_pad_query_peer_duration() and take into account that it may
	  return TRUE but still a duration of CLOCK_TIME_NONE, which easily
	  screws up comparisons when using unsigned integers. Fixes #504081.

2008-01-08 14:58:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Make elements GST_BUFFER_FLAG_GAP aware and call gst_base_transform_set_gap_aware for this.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/audiofx/audioamplify.c:
	  (gst_audio_amplify_clipping_method_get_type),
	  (gst_audio_amplify_init), (gst_audio_amplify_transform_ip):
	  * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_init),
	  (gst_audio_dynamic_transform_ip):
	  * gst/audiofx/audioinvert.c: (gst_audio_invert_init),
	  (gst_audio_invert_transform_ip):
	  * gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
	  (gst_audio_panorama_transform):
	  * gst/level/gstlevel.c: (gst_level_init):
	  Make elements GST_BUFFER_FLAG_GAP aware and call
	  gst_base_transform_set_gap_aware for this.
	  Bump core requirement to CVS.
	  * gst/audiofx/audiochebyshevfreqband.c:
	  (gst_audio_chebyshev_freq_band_transform_ip):
	  * gst/audiofx/audiochebyshevfreqlimit.c:
	  (gst_audio_chebyshev_freq_limit_transform_ip):
	  Also sync GObject properties to the controller if operating
	  in passthrough mode.

2008-01-07 16:41:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/directdraw/gstdirectdrawsink.c: FALSE is not a gpointer.
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c:
	  (gst_directdraw_sink_window_thread):
	  FALSE is not a gpointer.

2008-01-05 21:20:08 +0000  Julien Moutte <julien@moutte.net>

	  sys/directdraw/gstdirectdrawsink.c: Make sure we create our internal window only when we need it. That will give a ch...
	  Original commit message from CVS:
	  2008-01-05  Julien Moutte  <julien@fluendo.com>
	  * sys/directdraw/gstdirectdrawsink.c:
	  (gst_directdraw_sink_set_window_id),
	  (gst_directdraw_sink_set_caps),
	  (gst_directdraw_sink_change_state),
	  (gst_directdraw_sink_buffer_alloc),
	  (gst_directdraw_sink_draw_borders),
	  (gst_directdraw_sink_show_frame),
	  (gst_directdraw_sink_setup_ddraw),
	  (gst_directdraw_sink_window_thread),
	  (gst_directdraw_sink_get_ddrawcaps),
	  (gst_directdraw_sink_surface_create): Make sure we create our
	  internal window only when we need it. That will give a chance to
	  the application to get the prepare-xwindow-id bus message. Draw
	  black borders when keeping aspect ratio. Handle the case where
	  our
	  rendering window disappears (closed or errors) like other sinks
	  do. Various 80 columns fixes, improve state change order. That
	  element could need some more love.

2008-01-04 18:30:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/taglib/: Remove useless typedefs without new type name. Fixes a warning with gcc 4.3.
	  Original commit message from CVS:
	  * ext/taglib/gstapev2mux.h:
	  * ext/taglib/gstid3v2mux.h:
	  Remove useless typedefs without new type name. Fixes a warning with
	  gcc 4.3.

2008-01-03 12:26:03 +0000  John Millikin <jmillikin@gmail.com>

	  ext/flac/gstflacdec.c: Emit metadata messages when a PICTURE block is encountered.
	  Original commit message from CVS:
	  Patch by: John Millikin <jmillikin at gmail dot com>
	  * ext/flac/gstflacdec.c: (gst_flac_dec_setup_seekable_decoder),
	  (gst_flac_dec_setup_stream_decoder),
	  (gst_flac_normalize_picture_mime_type),
	  (gst_flac_extract_picture_buffer),
	  (gst_flac_dec_metadata_callback):
	  Emit metadata messages when a PICTURE block is encountered.
	  Fixes #506715.

2008-01-02 13:54:10 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/avi/gstavi.c: increase rank because no known issues anymore ...
	  Original commit message from CVS:
	  * gst/avi/gstavi.c:
	  increase rank because no known issues anymore ...
	  * gst/avi/gstavisubtitle.c:
	  send subtitle name to the srcpad

2007-12-31 13:27:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Implement redirect for the DESCRIBE reply. Fixes #506025.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open):
	  * gst/rtsp/gstrtspsrc.h:
	  Implement redirect for the DESCRIBE reply. Fixes #506025.

2007-12-29 16:48:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacdec.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached() ...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_write):
	  Fix 'xyz may be used uninitialized' compiler warnings caused
	  by broken g_assert_not_reached() macro in GLib-2.15.x and don't
	  abort() in any case but properly report the error.

2007-12-28 11:44:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/soup/: Use gst_tag_freeform_string_to_utf8() and post radio station info as tags on the bus.
	  Original commit message from CVS:
	  * ext/soup/Makefile.am:
	  * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_get_property),
	  (gst_souphttp_src_unicodify), (soup_got_headers):
	  Use gst_tag_freeform_string_to_utf8() and post radio station
	  info as tags on the bus.

2007-12-26 16:03:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached() macro in GLib-2.15.x (i...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_loop):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_chain):
	  * sys/ximage/gstximagesrc.c: (composite_pixel):
	  Fix 'xyz may be used uninitialized' compiler warnings caused
	  by broken g_assert_not_reached() macro in GLib-2.15.x (it's
	  not really nice to abort in any case). Fixes #505745.

2007-12-20 17:07:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Ignore more.
	  Original commit message from CVS:
	  * gst/equalizer/.cvsignore:
	  * gst/switch/.cvsignore:
	  Ignore more.

2007-12-18 23:17:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/avisubtitle.c: Small unit test fix (has no practical impact at the moment, since we're only feed...
	  Original commit message from CVS:
	  * tests/check/elements/avisubtitle.c: (check_correct_buffer):
	  Small unit test fix (has no practical impact at the moment,
	  since we're only feeding utf8 and hence just create a sub-
	  buffer for the output).

2007-12-18 21:13:05 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  Add seeking support for avi subtitle
	  Original commit message from CVS:
	  * gst/avi/gstavisubtitle.c:
	  * tests/check/elements/avisubtitle.c:
	  Add seeking support for avi subtitle

2007-12-18 17:40:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/flac/gstflacdec.*: Remove some unused vars.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
	  (gst_flac_dec_update_metadata), (gst_flac_dec_metadata_callback),
	  (gst_flac_dec_write):
	  * ext/flac/gstflacdec.h:
	  Remove some unused vars.
	  Do more cleanup of leftover events and tags.
	  Output tags after the segment event. Fixes #504018.

2007-12-18 14:31:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavisubtitle.c: Detect other UTF byte order markers and convert to UTF-8 as appropriate.
	  Original commit message from CVS:
	  * gst/avi/gstavisubtitle.c: (IS_BOM_UTF8), (IS_BOM_UTF16_BE),
	  (IS_BOM_UTF16_LE), (IS_BOM_UTF32_BE), (IS_BOM_UTF32_LE),
	  (gst_avi_subtitle_extract_file), (gst_avi_subtitle_parse_gab2_chunk):
	  Detect other UTF byte order markers and convert to UTF-8 as
	  appropriate.

2007-12-18 13:30:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavisubtitle.*: Refactor a bit; fix name extraction; don't assume all the data in the chunk is actually sub...
	  Original commit message from CVS:
	  * gst/avi/gstavisubtitle.c: (src_template),
	  (gst_avi_subtitle_extract_utf8_file),
	  (gst_avi_subtitle_parse_gab2_chunk), (gst_avi_subtitle_chain),
	  (gst_avi_subtitle_base_init), (gst_avi_subtitle_class_init),
	  (gst_avi_subtitle_init), (gst_avi_subtitle_change_state):
	  * gst/avi/gstavisubtitle.h:
	  Refactor a bit; fix name extraction; don't assume all the data
	  in the chunk is actually subtitle data, there may be padding at
	  the end; fix GST_ELEMENT_ERROR usage; store extracted subtitle
	  file so it's there to send again after a seek (for future use).

2007-12-18 09:13:12 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  Add avi subtitle element for bug #442034. Need seeking support and more support for character conversion.
	  Original commit message from CVS:
	  * gst/avi/Makefile.am:
	  * gst/avi/gstavi.c:
	  * gst/avi/gstavisubtitle.c:
	  * gst/avi/gstavisubtitle.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/avisubtitle.c:
	  * win32/common/config.h:
	  Add avi subtitle element for bug #442034. Need seeking support
	  and more support for character conversion.

2007-12-18 09:07:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Makefile.am: Include common/win32.mak for CRLF check of win32 project files (see #393626).
	  Original commit message from CVS:
	  * Makefile.am:
	  Include common/win32.mak for CRLF check of win32 project
	  files (see #393626).
	  * win32/vs6/libgstpng.dsp:
	  Fix line endings and do cvs admin -kb.

2007-12-17 21:12:28 +0000  David Schleef <ds@schleef.org>

	  gst/multifile/gstmultifilesrc.*: When subsequent files are read, if the file doesn't exist, send an EOS instead of ca...
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesrc.c:
	  * gst/multifile/gstmultifilesrc.h:
	  When subsequent files are read, if the file doesn't exist, send
	  an EOS instead of causing an error.

2007-12-16 23:43:46 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/jpeg/gstjpegdec.c: Actually drop the buffers which are outside the currently configured segment instead of just e...
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
	  Actually drop the buffers which are outside the currently configured
	  segment instead of just emitting a WARNING.

2007-12-14 18:49:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/flac/gstflacdec.*: Send segments from the streaming thread. Fixes #502187.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_metadata_callback),
	  (gst_flac_dec_write):
	  * ext/flac/gstflacdec.h:
	  Send segments from the streaming thread. Fixes #502187.
	  Fix segment seeking and a bunch of other seeking cases.

2007-12-14 10:17:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: Parse WOAF frames and put the result into GST_TAG_CONTACT, which is where it would end up...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (parse_url_link_frame):
	  Parse WOAF frames and put the result into GST_TAG_CONTACT,
	  which is where it would end up if the same information was
	  put in a vorbis comment (don't think it's worth adding a
	  new URI tag for this). Fixes #488112.

2007-12-11 22:29:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: We need core/base 0.10.15 or later.
	  Original commit message from CVS:
	  * configure.ac:
	  We need core/base 0.10.15 or later.

2007-12-11 16:47:12 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/avi/gstavimux.c: Fix regression in stream numbering. Fixes #502655.
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw skynet be>
	  * gst/avi/gstavimux.c: (gst_avi_mux_start_file):
	  Fix regression in stream numbering. Fixes #502655.

2007-12-11 16:39:39 +0000  Wouter Cloetens <wouter@mind.be>

	  ext/soup/gstsouphttpsrc.*: Do not try to unpause I/O in the "queued" state.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * ext/soup/gstsouphttpsrc.c: (_do_init),
	  (gst_souphttp_src_class_init), (gst_souphttp_src_init),
	  (gst_souphttp_src_dispose), (gst_souphttp_src_set_property),
	  (gst_souphttp_src_get_property), (unicodify),
	  (gst_souphttp_src_unicodify), (gst_souphttp_src_create),
	  (gst_souphttp_src_start), (gst_souphttp_src_stop),
	  (gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop),
	  (gst_souphttp_src_get_size), (gst_souphttp_src_is_seekable),
	  (soup_got_headers), (soup_got_body), (soup_finished),
	  (soup_got_chunk), (soup_response), (soup_parse_status),
	  (gst_souphttp_src_uri_get_type),
	  (gst_souphttp_src_uri_get_protocols),
	  (gst_souphttp_src_uri_get_uri), (gst_souphttp_src_uri_set_uri),
	  (gst_souphttp_src_uri_handler_init):
	  * ext/soup/gstsouphttpsrc.h:
	  Do not try to unpause I/O in the "queued" state.
	  Reorganise a bunch of things and cleanups.
	  Uses G_GUINT64_FORMAT instead of hard-coding %llu.
	  See #502335.

2007-12-11 16:31:49 +0000  Wai-Ming Ho <webregbox@yahoo.co.uk>

	  gst/rtp/gstrtph264pay.*: Use higher performance start-code searching.
	  Original commit message from CVS:
	  Patch by: Wai-Ming Ho <webregbox at yahoo dot co dot uk>
	  * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_init),
	  (gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
	  (next_start_code), (is_nal_equal), (gst_rtp_h264_pay_decode_nal),
	  (encode_base64), (gst_rtp_h264_pay_parse_sps_pps),
	  (gst_rtp_h264_pay_handle_buffer):
	  * gst/rtp/gstrtph264pay.h:
	  Use higher performance start-code searching.
	  Parse NALs and store SPS, PPS and profile in the caps so that they can
	  be used in the SDP. Fixes #502814.

2007-12-11 11:50:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/: Init some structs to zero before we pass them to ioctl, which avoids valgrind warnings.  Also fix a small ...
	  Original commit message from CVS:
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list):
	  Init some structs to zero before we pass them to ioctl, which
	  avoids valgrind warnings.  Also fix a small memory leak.

2007-12-11 11:05:57 +0000  Wouter Cloetens <wouter@mind.be>

	  gst/multipart/multipartdemux.c: Copy timestamp from input to output. Not very perfect yet but better than nothing. Fi...
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
	  Copy timestamp from input to output. Not very perfect yet but better
	  than nothing. Fixes #503023.

2007-12-09 16:49:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackdec.c: Also print a useful error message with the old Wavpack API if possible.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  Also print a useful error message with the old Wavpack API
	  if possible.

2007-12-09 16:34:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/gstwavpackdec.c: More build fixes for old libwavpack versions: include config.h so that WAVPACK_OLD_API i...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c:
	  More build fixes for old libwavpack versions: include config.h so
	  that WAVPACK_OLD_API is actually defined as detected; only use
	  WavpackGetErrorMessage if it is available. This fixes the build
	  on debian stable for me.

2007-12-09 16:21:02 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: Workaround the non-existance of WavpackGetChannelMask in Wavpack versions below 4.40.0.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_create_src_pad):
	  Workaround the non-existance of WavpackGetChannelMask in Wavpack
	  versions below 4.40.0.

2007-12-09 05:13:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: And now do it right for real...
	  Original commit message from CVS:
	  * configure.ac:
	  And now do it right for real...

2007-12-09 05:09:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Correctly reset $LIBS to not contain -lm.
	  Original commit message from CVS:
	  * configure.ac:
	  Correctly reset $LIBS to not contain -lm.

2007-12-09 05:02:17 +0000  Kwang Yul Seo <kwangyul.seo@gmail.com>

	  Fix compilation with MSVC by using gst_util_guint64_to_gdouble() and checking for rint() and implementing it ourself ...
	  Original commit message from CVS:
	  Based on a patch by: Kwang Yul Seo <kwangyul dot seo at gmail dot com>
	  * configure.ac:
	  * ext/cairo/gsttimeoverlay.c:
	  (gst_cairo_time_overlay_print_smpte_time):
	  Fix compilation with MSVC by using gst_util_guint64_to_gdouble()
	  and checking for rint() and implementing it ourself if it doesn't
	  exist.

2007-12-09 04:29:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
	  Original commit message from CVS:
	  * configure.ac:
	  Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.

2007-12-08 16:47:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/oss/gstosshelper.c: Verify that the format returned after the ioctl is the one we requested. It is valid for the ...
	  Original commit message from CVS:
	  * sys/oss/gstosshelper.c:
	  Verify that the format returned after the ioctl is the one
	  we requested. It is valid for the ioctl to succeed while
	  substituting an alternate 'supported' sample format.

2007-12-07 20:07:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/: Post decent (and translated) error message when we can't open the audio device for some reason.
	  Original commit message from CVS:
	  * sys/oss/gstossaudio.c: (plugin_init):
	  * sys/oss/gstosssink.c: (gst_oss_sink_open):
	  * sys/oss/gstosssrc.c: (gst_oss_src_open):
	  Post decent (and translated) error message when we can't
	  open the audio device for some reason.

2007-12-07 19:29:39 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/oss/: Allow the AUDIODEV environment variable to redirect us to a different default OSS device, like sunaudiosink...
	  Original commit message from CVS:
	  * sys/oss/gstosssink.c:
	  * sys/oss/gstosssrc.c:
	  Allow the AUDIODEV environment variable to redirect us
	  to a different default OSS device, like sunaudiosink does
	  on Solaris (makes audio play automatically on SunRays).

2007-12-06 12:45:50 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.c: Fix compilation.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_transform_ip):
	  Fix compilation.

2007-12-06 12:42:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.c: Don't process buffers in passthrough mode.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_transform_ip):
	  Don't process buffers in passthrough mode.

2007-12-06 12:37:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: The transform() methods are not called in passthrough mode so there's no need for checking if the elemen...
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (bpwsinc_transform):
	  * gst/filter/gstlpwsinc.c: (lpwsinc_transform):
	  The transform() methods are not called in passthrough mode so
	  there's no need for checking if the element is in passthrough mode.

2007-12-06 12:29:26 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: Sync the GObject properties with the controller even in passthrough mode to get consistent property values.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (bpwsinc_transform):
	  * gst/filter/gstlpwsinc.c: (lpwsinc_transform):
	  Sync the GObject properties with the controller even in passthrough
	  mode to get consistent property values.

2007-12-06 12:11:29 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: The transform_ip() methods should do nothing if in passthrough mode.
	  Original commit message from CVS:
	  * gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
	  * gst/audiofx/audiochebyshevfreqband.c:
	  (gst_audio_chebyshev_freq_band_transform_ip):
	  * gst/audiofx/audiochebyshevfreqlimit.c:
	  (gst_audio_chebyshev_freq_limit_transform_ip):
	  * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
	  * gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
	  The transform_ip() methods should do nothing if in passthrough mode.
	  It might get non-writable buffers in that case but the buffer might
	  as well be writable.
	  * gst/audiofx/audiopanorama.c: (gst_audio_panorama_transform):
	  The transform() methods won't be called in passthrough mode and
	  otherwise the buffer is always writable so don't check here.

2007-12-06 11:46:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: Fix seeking in .wav files again (#501775).  Some people seem to think they don't need to ...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_srcpad_event):
	  Fix seeking in .wav files again (#501775).  Some people seem to think
	  they don't need to test their changes when they're just 'reflowing'
	  some code.

2007-12-05 16:04:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/autodetect/gstautovideosink.*: Fix docs.
	  Original commit message from CVS:
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
	  (gst_auto_video_sink_init),
	  (gst_auto_video_sink_create_element_with_pretty_name),
	  (gst_auto_video_sink_find_best),
	  (gst_auto_video_sink_set_property),
	  (gst_auto_video_sink_get_property):
	  * gst/autodetect/gstautovideosink.h:
	  Fix docs.
	  Use same error reporting code as autoaudiosink.
	  Add property to filter sinks based on caps. Only select raw video sinks
	  by default for backwards compat.
	  API: GstAutoVideoSink::filter-caps

2007-12-05 16:02:15 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/autodetect/gstautoaudiosink.*: Add property to filter sinks based on caps. Only select raw audio sinks by default...
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
	  (gst_auto_audio_sink_init), (gst_auto_audio_sink_find_best),
	  (gst_auto_audio_sink_set_property),
	  (gst_auto_audio_sink_get_property):
	  * gst/autodetect/gstautoaudiosink.h:
	  Add property to filter sinks based on caps. Only select raw audio sinks
	  by default for backwards compat.  Fixes #417420.
	  API: GstAutoAudioSink::filter-caps

2007-11-29 11:40:15 +0000  Arek Korbik <arkadini@gmail.com>

	  gst/videobox/gstvideobox.c: Initialise liboil in plugin_init()
	  Original commit message from CVS:
	  Patch by: Arek Korbik <arkadini@gmail.com>
	  * gst/videobox/gstvideobox.c: (plugin_init):
	  Initialise liboil in plugin_init()

2007-11-29 10:49:18 +0000  Wouter Cloetens <wouter@mind.be>

	  configure.ac: Bump libsoup requirement as libsoup does not support async client operation prior to version 2.2.104 an...
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * configure.ac:
	  Bump libsoup requirement as libsoup does not support async client
	  operation prior to version 2.2.104 and it has some leaks.
	  * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init),
	  (gst_souphttp_src_init), (gst_souphttp_src_dispose),
	  (gst_souphttp_src_set_property), (gst_souphttp_src_create),
	  (gst_souphttp_src_start), (gst_souphttp_src_stop),
	  (gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop),
	  (gst_souphttp_src_get_size), (soup_got_headers), (soup_got_body),
	  (soup_finished), (soup_got_chunk), (soup_response),
	  (soup_session_close):
	  * ext/soup/gstsouphttpsrc.h:
	  Implement unlock().
	  Picks up the size from the Content-Length header and emit a duration
	  message.
	  Don't leak the GMainContext object.
	  Fixes #500099.

2007-11-29 10:34:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libpng/gstpngdec.c: Post error before sending EOS. Fixes #499178.
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  Post error before sending EOS. Fixes #499178.

2007-11-28 21:54:50 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/vs6/: Add a project file for libgstpng
	  Original commit message from CVS:
	  * win32/vs6/gst_plugins_good.dsw:
	  * win32/vs6/libgstpng.dsp:
	  Add a project file for libgstpng

2007-11-28 17:48:45 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/rtp/gstrtph263depay.c: Code beautification.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_class_init),
	  (gst_rtp_h263_depay_process):
	  Code beautification.
	  Added debug statements.
	  Don't bit-shift everything, just do operations on last/first byte
	  instead.

2007-11-27 11:11:08 +0000  Jayarama S. Santana <sundarsantana@gmail.com>

	  gst/rtp/gstrtpmp4adepay.c: Fix wrong comparison in overrun check. Fixes #499239 some more.
	  Original commit message from CVS:
	  Patch by: Jayarama S. Santana <sundarsantana at gmail dot com>
	  * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_process):
	  Fix wrong comparison in overrun check. Fixes #499239 some more.

2007-11-27 00:01:41 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/rtp/gstrtph263depay.*: Fix h263 depayloader so that ANY h263 decoder can handle the outgoing stream.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_init),
	  (gst_rtp_h263_depay_process):
	  * gst/rtp/gstrtph263depay.h:
	  Fix h263 depayloader so that ANY h263 decoder can handle the outgoing
	  stream.

2007-11-26 19:17:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4adepay.*: Fix depayloading when multiple frames are inside one RTP packet.
	  Original commit message from CVS:
	  Based on Path by: Jayarama S. Santana <sundarsantana at gmail dot com>
	  * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
	  (gst_rtp_mp4a_depay_process):
	  * gst/rtp/gstrtpmp4adepay.h:
	  Fix depayloading when multiple frames are inside one RTP packet.
	  Fixes #499239.

2007-11-26 12:26:20 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/level/gstlevel.c: Add GAP-flag support.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c:
	  Add GAP-flag support.

2007-11-26 12:01:11 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/rtp/gstrtph263depay.c: Read the I flag for Mode A h263 rtp stream and set the
	  Original commit message from CVS:
	  * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_process):
	  Read the I flag for Mode A h263 rtp stream and set the
	  GST_BUFFER_FLAG_DELTA_UNIT accordingly.
	  Fixes #499383

2007-11-26 10:08:20 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.c: Use dispose and finalize. Dispose can be called multiple times.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c:
	  Use dispose and finalize. Dispose can be called multiple times.

2007-11-26 10:04:49 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/level/gstlevel.c: Remove some dead code and do cleanups.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c:
	  Remove some dead code and do cleanups.

2007-11-26 09:13:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/pipelines/simple-launch-lines.c: Improve the tests by allowing to set a target state.
	  Original commit message from CVS:
	  * tests/check/pipelines/simple-launch-lines.c:
	  Improve the tests by allowing to set a target state.

2007-11-26 09:04:17 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/wavpackenc.c: Don't check the caps of the output buffer if they're equal some other caps. The ca...
	  Original commit message from CVS:
	  * tests/check/elements/wavpackenc.c: (GST_START_TEST):
	  Don't check the caps of the output buffer if they're equal some
	  other caps. The caps can change in a backward compatible way
	  and did at this point.

2007-11-24 14:55:04 +0000  Julien Moutte <julien@moutte.net>

	  gst/qtdemux/qtdemux.c: Implement reverse playback support.
	  Original commit message from CVS:
	  2007-11-24  Julien MOUTTE  <julien@moutte.net>
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_find_segment),
	  (gst_qtdemux_move_stream), (gst_qtdemux_do_seek),
	  (gst_qtdemux_seek_to_previous_keyframe),
	  (gst_qtdemux_activate_segment), (gst_qtdemux_advance_sample),
	  (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop): Implement
	  reverse playback support.

2007-11-21 09:56:54 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: Post a GST_MESSAGE_LATENCY if the latency changes.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (bpwsinc_set_property):
	  * gst/filter/gstlpwsinc.c: (lpwsinc_set_property):
	  Post a GST_MESSAGE_LATENCY if the latency changes.

2007-11-21 08:21:10 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/: Remove preset iface again. We'll re-add this after its been released in -good.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer10bands.c:
	  * gst/equalizer/gstiirequalizer3bands.c:
	  Remove preset iface again. We'll re-add this after its been released
	  in -good.

2007-11-20 13:14:40 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackcommon.c: Also set the channel layout on the Wavpack caps if we're having a mono layout. Of cou...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackcommon.c: (gst_wavpack_set_channel_layout):
	  Also set the channel layout on the Wavpack caps if we're having
	  a mono layout. Of course only do it for "audio/x-wavpack".

2007-11-20 13:08:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: Add support for encoding, parsing and decoding multichannel files with up to 8 channels. This also impr...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackcommon.c:
	  (gst_wavpack_get_default_channel_mask),
	  (gst_wavpack_set_channel_layout),
	  (gst_wavpack_get_default_channel_positions),
	  (gst_wavpack_get_channel_mask_from_positions),
	  (gst_wavpack_set_channel_mapping):
	  * ext/wavpack/gstwavpackcommon.h:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	  (gst_wavpack_dec_sink_set_caps), (gst_wavpack_dec_chain):
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
	  (gst_wavpack_enc_init), (gst_wavpack_enc_sink_set_caps),
	  (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_push_block),
	  (gst_wavpack_enc_fix_channel_order), (gst_wavpack_enc_chain),
	  (gst_wavpack_enc_rewrite_first_block),
	  (gst_wavpack_enc_sink_event):
	  * ext/wavpack/gstwavpackenc.h:
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
	  (gst_wavpack_parse_scan_to_find_sample),
	  (gst_wavpack_parse_sink_event), (gst_wavpack_parse_create_src_pad),
	  (gst_wavpack_parse_push_buffer), (gst_wavpack_parse_loop):
	  * ext/wavpack/gstwavpackparse.h:
	  Add support for encoding, parsing and decoding multichannel
	  files with up to 8 channels. This also improves the robustness
	  of parsing quite a bit.
	  * ext/wavpack/gstwavpackstreamreader.c:
	  (gst_wavpack_stream_reader_read_bytes),
	  (gst_wavpack_stream_reader_get_pos),
	  (gst_wavpack_stream_reader_set_pos_abs),
	  (gst_wavpack_stream_reader_set_pos_rel),
	  (gst_wavpack_stream_reader_push_back_byte),
	  (gst_wavpack_stream_reader_get_length),
	  (gst_wavpack_stream_reader_can_seek),
	  (gst_wavpack_stream_reader_write_bytes):
	  Improve debugging.

2007-11-20 12:20:38 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/libpng/gstpngdec.*: Don't release the png-memory from within the callback.
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c:
	  * ext/libpng/gstpngdec.h:
	  Don't release the png-memory from within the callback.

2007-11-20 12:11:13 +0000  René Stadler <mail@renestadler.de>

	  ext/libpng/gstpngenc.c: Don't leak buffer data memory. Fixes #498395.
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler dot de>
	  * ext/libpng/gstpngenc.c:
	  Don't leak buffer data memory. Fixes #498395.

2007-11-20 11:46:28 +0000  René Stadler <mail@renestadler.de>

	  tests/check/pipelines/simple-launch-lines.c: Tests for #498395.
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler dot de>
	  * tests/check/pipelines/simple-launch-lines.c:
	  Tests for #498395.

2007-11-20 11:41:13 +0000  Julien Moutte <julien@moutte.net>

	  Fix build on Mac OS X 10.5
	  Original commit message from CVS:
	  2007-11-20  Julien MOUTTE  <julien@moutte.net>
	  * ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag),
	  (gst_tag_lib_mux_adjust_event_offsets):
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_theora_extension):
	  * sys/osxaudio/Makefile.am:
	  * sys/osxvideo/cocoawindow.h:
	  * sys/osxvideo/cocoawindow.m: Fix build on Mac OS X 10.5

2007-11-19 20:30:19 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/: Activate preset iface and upload two presets here.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer10bands.c:
	  * gst/equalizer/gstiirequalizer3bands.c:
	  Activate preset iface and upload two presets here.

2007-11-16 05:52:55 +0000  David Schleef <ds@schleef.org>

	  ext/cairo/gsttextoverlay.c: Change strcasecmp() to g_strcasecmp().  Fixes #497292.
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c:
	  Change strcasecmp() to g_strcasecmp().  Fixes #497292.

2007-11-15 18:19:19 +0000  Jordi Jaen Pallares <jordijp@gmail.com>

	  gst/rtp/gstrtpmp2tpay.*: Fill the MTU with as many packets as possible. Fixes #491323.
	  Original commit message from CVS:
	  Patch by: Jordi Jaen Pallares <jordijp at gmail dot com>
	  * gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_class_init),
	  (gst_rtp_mp2t_pay_init), (gst_rtp_mp2t_pay_finalize),
	  (gst_rtp_mp2t_pay_flush), (gst_rtp_mp2t_pay_handle_buffer):
	  * gst/rtp/gstrtpmp2tpay.h:
	  Fill the MTU with as many packets as possible. Fixes #491323.

2007-11-15 17:47:43 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/rtsp/gstrtspsrc.c: Fix some more leaks. Fixes #497007.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	  Fix some more leaks. Fixes #497007.

2007-11-15 17:35:18 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/rtsp/gstrtspsrc.c: Fix 3 pad leaks. Fixes #496983.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_free),
	  (gst_rtspsrc_stream_configure_tcp):
	  Fix 3 pad leaks. Fixes #496983.

2007-11-15 17:26:25 +0000  Wouter Cloetens <wouter@mind.be>

	  Added HTTP source based on libsoup. Fixes #497020.
	  Original commit message from CVS:
	  Patch by: Wouter Cloetens <wouter at mind dot be>
	  * configure.ac:
	  * ext/Makefile.am:
	  * ext/soup/Makefile.am:
	  * ext/soup/gstsouphttpsrc.c: (_do_init),
	  (gst_souphttp_src_base_init), (gst_souphttp_src_class_init),
	  (gst_souphttp_src_init), (gst_souphttp_src_dispose),
	  (gst_souphttp_src_set_property), (gst_souphttp_src_get_property),
	  (gst_souphttp_src_create), (gst_souphttp_src_start),
	  (gst_souphttp_src_stop), (gst_souphttp_src_unlock),
	  (gst_souphttp_src_set_location), (soup_got_chunk), (soup_response),
	  (soup_session_close), (plugin_init):
	  * ext/soup/gstsouphttpsrc.h:
	  Added HTTP source based on libsoup. Fixes #497020.

2007-11-15 17:01:32 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/rtp/gstrtph264depay.c: Fix small leak. Fixes #497017.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	  Fix small leak. Fixes #497017.

2007-11-15 16:31:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/: Add suppport for theora in quicktime according to XiphQT.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
	  (gst_qtdemux_prepare_current_sample),
	  (gst_qtdemux_loop_state_movie), (qtdemux_parse_theora_extension),
	  (qtdemux_parse_node), (qtdemux_parse_trak), (qtdemux_video_caps):
	  * gst/qtdemux/qtdemux_fourcc.h:
	  * gst/qtdemux/qtdemux_types.c:
	  Add suppport for theora in quicktime according to XiphQT.

2007-11-15 12:22:10 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2src_calls.c:
	  Always copy buffers by default (handle safer with bugged drivers) and added a property to make it possible to use mma...
	  Original commit message from CVS:
	  Always copy buffers by default (handle safer with bugged drivers) and added a property to make it possible to use mmap effectively (no copy if possible) when application wants to. Fixes: #480557.

2007-11-14 21:39:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/: We don't want the same string multiple times in a tag list for the same tag ever, for any tag, not jus...
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c:
	  * gst/id3demux/id3tags.h:
	  * gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
	  We don't want the same string multiple times in a tag list for the
	  same tag ever, for any tag, not just for GST_TAG_GENRE, so make sure
	  this doesn't happen and remove special-case code for GST_TAG_GENRE.

2007-11-14 21:04:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gstid3v2mux.cc: Write GST_TAG_MUSICBRAINZ_DISCID and GST_TAG_CDDA_CDDB_DISCID into ID3v2 TXXX frames (fixe...
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc: (add_musicbrainz_tag), (add_funcs):
	  Write GST_TAG_MUSICBRAINZ_DISCID and GST_TAG_CDDA_CDDB_DISCID
	  into ID3v2 TXXX frames (fixes #347848).

2007-11-14 20:34:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtsp/gstrtspsrc.c: Don't leak sdp message contents (fixes #496773).
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	  Don't leak sdp message contents (fixes #496773).
	  * gst/udp/gstudpsink.c: (gst_udpsink_finalize):
	  Don't leak URI string.

2007-11-14 19:10:37 +0000  Julien Puydt <julien.puydt@laposte.net>

	  ext/raw1394/: Implement GstPropertyProbe interface and add "device-name" property, so applications can use this to pr...
	  Original commit message from CVS:
	  Patch by: Julien Puydt <julien dot puydt at laposte net>
	  * ext/raw1394/Makefile.am:
	  * ext/raw1394/gst1394probe.c: (gst_1394_get_guid_array),
	  (gst_1394_property_probe_get_properties),
	  (gst_1394_property_probe_probe_property),
	  (gst_1394_property_probe_needs_probe),
	  (gst_1394_property_probe_get_values),
	  (gst_1394_property_probe_interface_init),
	  (gst_1394_type_add_property_probe_interface):
	  * ext/raw1394/gst1394probe.h: (GST_1394_PROBE_H):
	  * ext/raw1394/gstdv1394src.c: (_do_init), (gst_dv1394src_class_init),
	  (gst_dv1394src_init), (gst_dv1394src_dispose),
	  (gst_dv1394src_set_property), (gst_dv1394src_get_property),
	  (gst_dv1394src_discover_avc_node), (gst_dv1394src_query),
	  (gst_dv1394src_update_device_name):
	  * ext/raw1394/gstdv1394src.h:
	  Implement GstPropertyProbe interface and add "device-name" property,
	  so applications can use this to probe for available devices in the
	  same way they can already with v4lsrc and v4l2src (however horrible
	  this property probe interface may be). Fixes #358841.

2007-11-14 17:03:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/spectrum.c: Fix spectrum unit test for the latest spectrum changes.
	  Original commit message from CVS:
	  * tests/check/elements/spectrum.c: (GST_START_TEST):
	  Fix spectrum unit test for the latest spectrum changes.

2007-11-14 15:29:05 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/rtsp/gstrtspsrc.c: Don't leak event, don't leak range (fixes #496752).
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
	  (gst_rtspsrc_parse_range):
	  Don't leak event, don't leak range (fixes #496752).

2007-11-14 10:22:41 +0000  Arek Korbik <arkadini@gmail.com>

	  gst/alpha/gstalphacolor.c: Detect RGBA/BGRA correctly on little endian systems.
	  Original commit message from CVS:
	  Patch by: Arek Korbik <arkadini@gmail.com>
	  * gst/alpha/gstalphacolor.c: (gst_alpha_color_set_caps):
	  Detect RGBA/BGRA correctly on little endian systems.

2007-11-13 17:19:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/v4l2src_calls.c: but the corresponding ioctl() call fails even though the driver claims to support this form...
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format):
	  If VIDIOC_ENUM_FRAMESIZES is defined (= recent kernel), but the
	  corresponding ioctl() call fails even though the driver claims to
	  support this format, just fall back to the pre-2.6.19 kernel
	  routine that creates caps with suitable height and width ranges
	  (see #448278).

2007-11-13 17:01:07 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/matroska/: Extract palette data for dvd subpicture streams and send it downstream as custom gstreamer dvd event (...
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw skynet be>
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_push_dvd_clut_change_event),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_subtitle_caps):
	  * gst/matroska/matroska-ids.h:
	  Extract palette data for dvd subpicture streams and send it
	  downstream as custom gstreamer dvd event (fixes #453417).

2007-11-13 14:51:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cairo/gsttextoverlay.c: Implement minimal parsing of the passed pango font description string, so passing a font ...
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_font_init):
	  Implement minimal parsing of the passed pango font description
	  string, so passing a font size works the same as with the
	  pango textoverlay plugin; fixes #455086.
	  (Maybe we could just use pangocairo here at some point).

2007-11-13 06:55:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: Return the result in _activate_pull(). Don't ref element there.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  * gst/wavparse/gstwavparse.c:
	  Return the result in _activate_pull(). Don't ref element there.

2007-11-13 06:23:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: Ref the element when we should, but not when we its not needed. Reflow the event_handling...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
	  (gst_wavparse_pad_convert), (gst_wavparse_pad_query),
	  (gst_wavparse_srcpad_event):
	  Ref the element when we should, but not when we its not needed. Reflow
	  the event_handling to not leak the event.

2007-11-12 21:07:31 +0000  René Stadler <mail@renestadler.de>

	  gst/replaygain/rganalysis.c: Avoid slowdown from denormals when processing near-silence input data.
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler dot de>
	  * gst/replaygain/rganalysis.c: (yule_filter):
	  Avoid slowdown from denormals when processing near-silence input data.
	  Spotted by Gabriel Bouvigne. Fixes #494499.

2007-11-12 17:59:40 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Properly free QTDemuxSamples array.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
	  (qtdemux_parse_samples):
	  Properly free QTDemuxSamples array.
	  Protect table write with a sensible check, some files apparently DO contain
	  stts values starting with 0 :(

2007-11-12 17:21:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: Drop EOS in _handle_src_event(). Fix the refcount in qtdemux that previous commit messed up.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  * gst/qtdemux/qtdemux.c:
	  Drop EOS in _handle_src_event(). Fix the refcount in qtdemux that
	  previous commit messed up.

2007-11-12 17:06:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: Sync _handle_src_event() with oggdemux. In avidemux also ref the element when we should, but not when we its no...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  * gst/qtdemux/qtdemux.c:
	  Sync _handle_src_event() with oggdemux. In avidemux also ref the
	  element when we should, but not when we its not needed.

2007-11-11 21:12:10 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/: Change the meaning of the magnitude values given in the
	  Original commit message from CVS:
	  * gst/equalizer/demo.c: (draw_spectrum):
	  * gst/spectrum/demo-audiotest.c: (draw_spectrum):
	  * gst/spectrum/demo-osssrc.c: (draw_spectrum):
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
	  Change the meaning of the magnitude values given in the
	  GstMessages by spectrum to decibel instead of
	  decibel+threshold.

2007-11-11 13:55:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/: And continue to update docs. Also include some sample code for the n-band equalizer in the docs.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer10bands.c:
	  * gst/equalizer/gstiirequalizer3bands.c:
	  * gst/equalizer/gstiirequalizernbands.c:
	  And continue to update docs. Also include some sample code
	  for the n-band equalizer in the docs.

2007-11-11 12:54:31 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/: Update docs and property ranges to the real values.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer10bands.c:
	  (gst_iir_equalizer_10bands_class_init):
	  * gst/equalizer/gstiirequalizer3bands.c:
	  (gst_iir_equalizer_3bands_class_init):
	  * gst/equalizer/gstiirequalizernbands.c:
	  Update docs and property ranges to the real values.

2007-11-09 17:27:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/spectrum/gstspectrum.c: Now do the scaling right for real. Also initialize a previously uninitialized variable.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c:
	  Now do the scaling right for real. Also initialize a previously
	  uninitialized variable.

2007-11-08 15:56:46 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/demo.c: Make default volume a bit less. Improve layout by giving more space to the slider with big-numb...
	  Original commit message from CVS:
	  * gst/equalizer/demo.c:
	  Make default volume a bit less. Improve layout by giving more space to
	  the slider with big-numbers and enable fill.

2007-11-08 15:00:40 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: Return FALSE if we can't handle a query instead of changing the format. Ignore fact when ...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Return FALSE if we can't handle a query instead of changing the
	  format. Ignore fact when dealing with mpeg audio.

2007-11-06 12:23:35 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/spectrum/demo-audiotest.c: Use autoaudiosink instead of alsasink and use a sine wave.
	  Original commit message from CVS:
	  * gst/spectrum/demo-audiotest.c: (main):
	  Use autoaudiosink instead of alsasink and use a sine wave.
	  * gst/spectrum/gstspectrum.c:
	  Fix the magnitude calculation.

2007-11-03 19:50:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/: Allow setting 0 as bandwidth and handle this correctly.
	  Original commit message from CVS:
	  * gst/equalizer/demo.c: (main):
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_band_class_init), (setup_filter):
	  Allow setting 0 as bandwidth and handle this correctly.
	  Also handle a bandwidth of rate/2 properly.
	  * gst/equalizer/gstiirequalizernbands.c:
	  (gst_iir_equalizer_nbands_class_init):
	  Make it possible to generate a N-band equalizer with 1 bands. The
	  previous limit of 2 was caused by a nowadays replaced calculation
	  doing a division by zero if number of bands was 1.

2007-11-02 21:16:09 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  Fix includes for MSVC and GLib-2.14.0 (#492388).
	  Original commit message from CVS:
	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
	  * configure.ac:
	  * gst/udp/gstdynudpsink.c:
	  * gst/udp/gstdynudpsink.h:
	  * gst/udp/gstmultiudpsink.c:
	  * gst/udp/gstmultiudpsink.h:
	  * gst/udp/gstudpsink.c:
	  * gst/udp/gstudpsink.h:
	  Fix includes for MSVC and GLib-2.14.0 (#492388).
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  No more pipe define since GLib-2.14.0, need to use _pipe() directly.

2007-11-02 17:23:43 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/law/mulaw-decode.*: Calculate outgoing buffer duration if incoming buffer didn't have a valid duration.
	  Original commit message from CVS:
	  * gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
	  (gst_mulawdec_chain):
	  * gst/law/mulaw-decode.h:
	  Calculate outgoing buffer duration if incoming buffer didn't have a
	  valid duration.

2007-10-30 21:37:49 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/: Add small demo application based on the spectrum demo applications that gets white noise as input, pu...
	  Original commit message from CVS:
	  * gst/equalizer/Makefile.am:
	  * gst/equalizer/demo.c: (on_window_destroy), (on_configure_event),
	  (on_gain_changed), (on_bandwidth_changed), (on_freq_changed),
	  (draw_spectrum), (message_handler), (main):
	  Add small demo application based on the spectrum demo applications
	  that gets white noise as input, pushes it through an equalizer and
	  paints the spectrum. For every equalizer band it's possible to set
	  gain, bandwidth and frequency.
	  * gst/equalizer/gstiirequalizer.c: (setup_filter):
	  Add some guarding against too large or too small frequencies and
	  bandwidths. Also improve debugging a bit.

2007-10-30 21:18:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.c: Replace filters with a bit better filters for which we can actually find documentati...
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_band_set_property),
	  (gst_iir_equalizer_band_get_property),
	  (gst_iir_equalizer_band_class_init), (arg_to_scale),
	  (setup_filter), (gst_iir_equalizer_compute_frequencies):
	  Replace filters with a bit better filters for which we can actually
	  find documentation, which don't change anything on zero gain, etc.
	  Make the frequency property of the bands writable, rename the
	  band-width property to bandwidth and change the	meaning to the
	  frequency difference between bandedges, change the meaning of the
	  gain property to dB instead of a weird scale between -1	and 1 that
	  has no real meaning.

2007-10-30 12:29:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Smarter combine_flow code that also deals with downstream elements returning UNEXPECTED when t...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	  (gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie):
	  Smarter combine_flow code that also deals with downstream elements
	  returning UNEXPECTED when they receive data out of the segment
	  boundaries. Fixes #491305.

2007-10-27 16:04:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/interleave/interleave.c: Let's not call every request pad we create "sink%d", that'll create problems if there's ...
	  Original commit message from CVS:
	  * gst/interleave/interleave.c: (gst_interleave_request_new_pad):
	  Let's not call every request pad we create "sink%d", that'll
	  create problems if there's to be more than one pad. Fixes #490682.
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/interleave.c:
	  Add unit test for the above.

2007-10-26 15:03:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/v4l2src_calls.c: Fix 'unused variable' compiler warning when compiling against older kernel headers.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c:
	  Fix 'unused variable' compiler warning when compiling against
	  older kernel headers.

2007-10-26 12:10:43 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  update spec file
	  Original commit message from CVS:
	  update spec file

2007-10-25 23:42:52 +0000  David Schleef <ds@schleef.org>

	  Improve documentation, write some tests for multifilesrc/sink for upcoming ->good review.
	  Original commit message from CVS:
	  * gst/multifile/Makefile.am:
	  * gst/multifile/gstmultifilesink.c:
	  * gst/multifile/gstmultifilesrc.c:
	  * tests/check/Makefile.am:
	  * tests/check/elements/multifile.c:
	  Improve documentation, write some tests for multifilesrc/sink
	  for upcoming ->good review.

2007-10-25 15:00:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gstid3v2mux.cc (add_funcs): Map new SORTNAME tags to ID3v2 TSOP, TSOA and TSOT frames (#414539).
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc (add_funcs):
	  Map new SORTNAME tags to ID3v2 TSOP, TSOA and TSOT frames (#414539).

2007-10-24 07:01:47 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/pipelines/simple-launch-lines.c: Improve the tests a little more.
	  Original commit message from CVS:
	  * tests/check/pipelines/simple-launch-lines.c:
	  Improve the tests a little more.

2007-10-23 08:38:50 +0000  Yun Zheng Hu <yunzheng.hu@gmail.com>

	  sys/osxaudio/gstosxaudiosrc.c: Use default input device instead of default output device and only memcpy actual avail...
	  Original commit message from CVS:
	  patch by: Yun Zheng Hu
	  * sys/osxaudio/gstosxaudiosrc.c:
	  Use default input device instead of default output device and
	  only memcpy actual available bytes.

2007-10-22 19:14:08 +0000  Edgard Lima <edgard.lima@indt.org.br>

	  sys/v4l2/v4l2src_calls.c: Fixes "v4l2src ! queue ! xvimagesink". The queue ask for buffer too early. It is temporary ...
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  Fixes "v4l2src ! queue ! xvimagesink". The queue ask for buffer too
	  early. It is temporary until we find something better.

2007-10-22 16:44:48 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/rtsp/gstrtspsrc.c: Fix race when pausing a RTSP stream in interleaved.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved):
	  Fix race when pausing a RTSP stream in interleaved.
	  Fixes #475784.

2007-10-22 09:53:16 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtp/gstrtpmp4vpay.c: Use correct unref function for buffers. #488844.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt <pkj at axis com>
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_finalize):
	  Use correct unref function for buffers. #488844.

2007-10-19 19:33:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add some debug and sync tests with the fix.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  * tests/check/elements/avimux.c:
	  Add some debug and sync tests with the fix.

2007-10-18 17:04:14 +0000  Laurent Glayal <spglegle@yahoo.fr>

	  gst/udp/gstudpsrc.c: When the socket is used by the app for other purposes, don't generate an error if there is activ...
	  Original commit message from CVS:
	  Based on patch by: Laurent Glayal  <spglegle yahoo fr>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  When the socket is used by the app for other purposes, don't generate an
	  error if there is activaty on the socket that is not data related.
	  Fixes #487488.

2007-10-18 14:55:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/v4l2src_calls.c: Add some more debug info. Generate an error when we run out of buffers for some reason. See...
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize),
	  (gst_v4l2src_grab_frame):
	  Add some more debug info. Generate an error when we run out of buffers
	  for some reason. See #480557.

2007-10-18 08:27:56 +0000  Anders Skargren <anders.skargren@axis.com>

	  gst/rtp/gstrtph264pay.c: Set marker bit correctly.
	  Original commit message from CVS:
	  Patch by: Anders Skargren <anders dot skargren at axis dot com>
	  * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
	  Set marker bit correctly.

2007-10-18 06:20:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.c: Add a missing break.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_band_set_property):
	  Add a missing break.

2007-10-18 06:14:42 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/equalizer/gstiirequalizer.*: Move bandwidth property to the separate bands and add float64 support.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_band_set_property),
	  (gst_iir_equalizer_band_get_property),
	  (gst_iir_equalizer_band_class_init), (gst_iir_equalizer_band_init),
	  (gst_iir_equalizer_band_get_type), (gst_iir_equalizer_class_init),
	  (setup_filter), (gst_iir_equalizer_setup):
	  * gst/equalizer/gstiirequalizer.h:
	  Move bandwidth property to the separate bands and add float64 support.

2007-10-17 15:08:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Use allowed name for the GstStructure.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	  Use allowed name for the GstStructure.

2007-10-17 11:47:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use new gst_bus_pop_filtered().
	  Original commit message from CVS:
	  * ext/gconf/gstswitchsink.c:
	  * gst/autodetect/gstautoaudiosink.c:
	  Use new gst_bus_pop_filtered().

2007-10-13 12:03:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/: When probing the formats and sizes a camera supports, make sure the best ones (highest resolution, prefere...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c:
	  * sys/v4l2/v4l2src_calls.c:
	  When probing the formats and sizes a camera supports, make
	  sure the best ones (highest resolution, prefered format)
	  end up at the beginning of the probed caps and the less
	  desirable ones at the end.  This is important because the
	  order within the caps matters for things like fixation and
	  negotiation, ie. what format is chosen in the end.
	  With recent kernels, the current probing code will end up
	  querying the supported sizes from lowest resolution to
	  highest resolution, adding them to the probed caps in that
	  order, resulting to v4l2src fixating to the lowest possible
	  resolution if downstream does not express a size preference.
	  Also make up a somewhat random ranking of prefered output
	  formats for the same reason. Fixes #485828.

2007-10-11 17:55:29 +0000  Jason Kivlighn <jkivlighn@gmail.com>

	  gst/id3demux/id3v2frames.c: Extract license/copyright URIs from ID3v2 WCOP frames (Fixes #447000).
	  Original commit message from CVS:
	  Based on patch by: Jason Kivlighn  <jkivlighn gmail com>
	  * gst/id3demux/id3v2frames.c:
	  Extract license/copyright URIs from ID3v2 WCOP frames
	  (Fixes #447000).
	  * tests/check/elements/id3demux.c:
	  * tests/files/Makefile.am:
	  * tests/files/id3-447000-wcop.tag:
	  Add simple unit test.

2007-10-11 16:41:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gstid3v2mux.cc: Add support for license/copyright URI tags (ID3v2 WCOP frame).
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc:
	  Add support for license/copyright URI tags (ID3v2 WCOP frame).
	  Prerequisite for #447000.

2007-10-08 17:44:42 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/rtsp/gstrtspsrc.c: Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise a GstClockTime.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush):
	  Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise
	  a GstClockTime.

2007-10-08 11:58:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: More seeking fixes, mostly passing around the new playback segment in order to configure it pr...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	  (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
	  (gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
	  (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
	  (gst_rtspsrc_change_state):
	  More seeking fixes, mostly passing around the new playback segment in
	  order to configure it properly.
	  Also reset base_time of udp sources when setting them back to PLAYING as
	  a temporary hack until core supports seek in live sources properly.

2007-10-08 10:34:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4adepay.c: Fix caps as to not confuse autopluggers.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4adepay.c:
	  Fix caps as to not confuse autopluggers.

2007-10-06 16:13:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/: Port ID3 tag demuxer over to the new GstTagDemux in -base (now would be a good time to test re-importi...
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c:
	  * gst/id3demux/gstid3demux.h:
	  * gst/id3demux/id3tags.c:
	  * gst/id3demux/id3tags.h:
	  * gst/id3demux/id3v2frames.c:
	  Port ID3 tag demuxer over to the new GstTagDemux in -base
	  (now would be a good time to test re-importing your music
	  collection).

2007-10-06 15:13:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/apetag/: Port APE tag demuxer over to the new GstTagDemux in -base.
	  Original commit message from CVS:
	  * gst/apetag/Makefile.am:
	  * gst/apetag/gstapedemux.c:
	  * gst/apetag/gstapedemux.h:
	  * gst/apetag/gsttagdemux.c:
	  * gst/apetag/gsttagdemux.h:
	  Port APE tag demuxer over to the new GstTagDemux in -base.

2007-10-05 13:18:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Improve flushing behaviour.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	  (gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event),
	  (gst_rtspsrc_handle_internal_src_query),
	  (gst_rtspsrc_handle_src_query), (new_session_pad),
	  (gst_rtspsrc_stream_configure_tcp),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_loop_send_cmd):
	  Improve flushing behaviour.
	  Set state of the udp sources to PAUSE/PLAYING correctly.
	  Handle events and queries for UDP and TCP transport now.

2007-10-04 07:29:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/rtp/: Add log category.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpgsmdepay.c:
	  * gst/rtp/gstrtpgsmpay.c:
	  Add log category.

2007-10-04 07:24:02 +0000  Timo Hotti <Timo.Hotti@sysopendigia.com>

	  tests/check/: Add unit tests for payloaders/depayloaders.
	  Original commit message from CVS:
	  Patch by: Timo Hotti <Timo.Hotti@sysopendigia.com>
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/simple-launch-lines.c:
	  Add unit tests for payloaders/depayloaders.

2007-10-02 10:49:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.*: Also save codec data for audio streams. Fixes #482495.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  * gst/avi/gstavimux.h:
	  Also save codec data for audio streams. Fixes #482495.

2007-10-02 10:23:04 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: Fix "Index entry has invalid stream nr 1".
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Fix "Index entry has invalid stream nr 1".
	  Add support for muxing aac - work in progress (see #482495).

2007-10-01 16:34:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
	  (gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
	  (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
	  * gst/rtsp/gstrtspsrc.h:
	  Parse bandwidth modifiers, they are not yet configured in the session
	  manager because we don't have an API for that yet.

2007-10-01 13:57:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Use shiny new function in -base to get the default clock-rate.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
	  (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
	  Use shiny new function in -base to get the default clock-rate.
	  Update some docs.

2007-09-29 12:50:36 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/MANIFEST: Add files to win32 manifest.
	  Original commit message from CVS:
	  * win32/MANIFEST:
	  Add files to win32 manifest.
	  * win32/vs6/libgstaudiofx.dsp:
	  * win32/vs6/libgstqtdemux.dsp:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgstrtsp.dsp:
	  Update project files.

2007-09-28 14:56:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense ...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_play):
	  * gst/rtsp/gstrtspsrc.h:
	  In TCP mode, only timestamp the first buffer. TCP is not real time and
	  it does not make sense to try to skew compensate, also some servers send
	  the first batch of data in a burst.

2007-09-27 15:00:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: Fix setting the discont flag on the first buffer pushed downstream for formats with pr...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  Fix setting the discont flag on the first buffer
	  pushed downstream for formats with private codec
	  data that needs to be deserialised into buffers
	  (such as vorbis and FLAC when in a matroska container).

2007-09-27 11:10:12 +0000  Antoine Tremblay <hexa00@gmail.com>

	  gst/rtp/gstrtpmp4vpay.*: Free the config string. Fixes #480707.
	  Original commit message from CVS:
	  Patch by: Antoine Tremblay <hexa00 at gmail dot com>
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
	  (gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush),
	  (gst_rtp_mp4v_pay_handle_buffer):
	  * gst/rtp/gstrtpmp4vpay.h:
	  Free the config string. Fixes #480707.
	  Clean up the timestamp code a little.

2007-09-26 20:12:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Set timestamps on RTP buffers in interleaved mode.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	  (gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
	  * gst/rtsp/gstrtspsrc.h:
	  Set timestamps on RTP buffers in interleaved mode.
	  Mark first buffers with a DISCONT.
	  Remove flush hack now that sync for live sources has been figured out.

2007-09-26 14:28:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Update documentation.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Update documentation.

2007-09-26 14:26:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/gstrtpxqtdepay.*: Fail if we don't know the quicktime format.
	  Original commit message from CVS:
	  * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
	  (gst_rtp_xqt_depay_change_state):
	  * gst/qtdemux/gstrtpxqtdepay.h:
	  Fail if we don't know the quicktime format.

2007-09-26 13:40:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.c: Fix up case where there is no peer, in which case _get_allowed_caps() will return NULL.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c:
	  Fix up case where there is no peer, in which case
	  _get_allowed_caps() will return NULL.

2007-09-26 13:19:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacenc.*: Save the flow return from the last gst_pad_push() and make sure we pass the right flow return ...
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c:
	  * ext/flac/gstflacenc.h:
	  Save the flow return from the last gst_pad_push() and
	  make sure we pass the right flow return value upstream
	  in the case of failure; minor clean-ups.

2007-09-25 19:09:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add support for the new GST_TAG_COMPOSER (#459809).
	  Original commit message from CVS:
	  * ext/taglib/gstapev2mux.cc:
	  * ext/taglib/gstid3v2mux.cc:
	  * gst/apetag/gstapedemux.c:
	  Add support for the new GST_TAG_COMPOSER (#459809).

2007-09-25 17:18:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/law/: Compulsive clean-ups: use boilerplate macros, add debug categories, fix up things to conform to symbol nome...
	  Original commit message from CVS:
	  * gst/law/alaw-decode.c:
	  * gst/law/alaw-decode.h:
	  * gst/law/alaw-encode.c:
	  * gst/law/alaw-encode.h:
	  * gst/law/alaw.c:
	  * gst/law/mulaw-conversion.h:
	  Compulsive clean-ups: use boilerplate macros, add debug
	  categories, fix up things to conform to symbol nomenklatura,
	  etc.

2007-09-25 16:05:29 +0000  Laurent Glayal <spglegle@yahoo.fr>

	  gst/law/: Use static tables for A-Law decoding and encoding; this makes
	  Original commit message from CVS:
	  Based on patch by: Laurent Glayal  <spglegle yahoo fr>
	  * gst/law/alaw-decode.c:
	  * gst/law/alaw-encode.c:
	  Use static tables for A-Law decoding and encoding; this makes
	  A-Law decoding and encoding less CPU-intensive, but increases
	  the binary size a bit. Leaving old code around for now,
	  selectable by a define in the code. Fixes #435435.

2007-09-25 13:20:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.c: Use GST_PTR_FORMAT to print caps in debug statement.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c:
	  Use GST_PTR_FORMAT to print caps in debug statement.

2007-09-25 08:51:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and
	  Original commit message from CVS:
	  * configure.ac:
	  Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and
	  AG_GST_ARG_ENABLE_EXPERIMENTAL instead of duplicating those macros
	  in configure.ac.

2007-09-25 05:03:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/qtdemux/qtdemux.c: Add fourccs for MPEG2 HDV streams. Fixes #479960.
	  Original commit message from CVS:
	  Patch by: <j at bootlab dot org>
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add fourccs for MPEG2 HDV streams. Fixes #479960.

2007-09-24 10:53:36 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Massive leak fixing, plus code cleanups.
	  Original commit message from CVS:
	  * ext/audioresample/gstaudioresample.c:
	  * ext/x264/gstx264enc.c:
	  * gst/dvdspu/gstdvdspu.c:
	  * gst/dvdspu/gstdvdspu.h:
	  * gst/festival/gstfestival.c:
	  * gst/h264parse/gsth264parse.c:
	  * gst/mpegtsparse/mpegtspacketizer.c:
	  * gst/mpegtsparse/mpegtsparse.c:
	  * gst/multifile/gstmultifilesink.c:
	  * gst/multifile/gstmultifilesrc.c:
	  * gst/nuvdemux/gstnuvdemux.c:
	  * sys/dshowsrcwrapper/gstdshowaudiosrc.c:
	  * sys/dshowsrcwrapper/gstdshowvideosrc.c:
	  * sys/vcd/vcdsrc.c:
	  Massive leak fixing, plus code cleanups.

2007-09-24 10:26:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/lame/gstlame.c: Allow fixing the sample rate lame converts to by negotiating fixed sample rate on the src pad caps.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c:
	  Allow fixing the sample rate lame converts to by negotiating fixed
	  sample rate on the src pad caps.
	  Add docs for it.
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/lame.c:
	  Add a check for it.

2007-09-23 18:57:14 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/oss/gstosshelper.c: Use GST_WARNING instead of a g_critical. This situation is not caused by the application.
	  Original commit message from CVS:
	  * sys/oss/gstosshelper.c:
	  Use GST_WARNING instead of a g_critical. This situation is not caused
	  by the application.

2007-09-22 18:15:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/: Updated translations.
	  Original commit message from CVS:
	  * po/LINGUAS:
	  * po/nl.po:
	  Updated translations.

2007-09-22 18:13:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/eu.po: Added Basque translation.
	  Original commit message from CVS:
	  translated by: Mikel Olasagasti <hey_neken@mundurat.net>
	  * po/eu.po:
	  Added Basque translation.

2007-09-22 18:13:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/: Added Chinese (traditional and Hong Kong) translation.
	  Original commit message from CVS:
	  translated by: Abel Cheung <abelcheung@gmail.com>
	  * po/zh_HK.po:
	  * po/zh_TW.po:
	  Added Chinese (traditional and Hong Kong) translation.

2007-09-22 18:10:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/pl.po: Added Polish translation.
	  Original commit message from CVS:
	  translated by: Jakub Bogusz <qboosh@pld-linux.org>
	  * po/pl.po:
	  Added Polish translation.

2007-09-22 18:09:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/fi.po: Added Finnish translation.
	  Original commit message from CVS:
	  translated by: Ilkka Tuohela <hile@iki.fi>
	  * po/fi.po:
	  Added Finnish translation.

2007-09-22 18:09:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/es.po: Added Spanish translation.
	  Original commit message from CVS:
	  translated by: Jorge González González <aloriel@gmail.com>
	  * po/es.po:
	  Added Spanish translation.

2007-09-22 18:08:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/da.po: Added Danish translation.
	  Original commit message from CVS:
	  translated by: Mogens Jaeger <mogens@jaeger.tf>
	  * po/da.po:
	  Added Danish translation.

2007-09-22 18:06:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/zh_CN.po: Added Chinese (simplified) translation.
	  Original commit message from CVS:
	  translated by: Funda Wang <fundawang@linux.net.cn>
	  * po/zh_CN.po:
	  Added Chinese (simplified) translation.

2007-09-22 18:05:37 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/bg.po: Added Bulgarian translation.
	  Original commit message from CVS:
	  translated by: Alexander Shopov <ash@contact.bg>
	  * po/bg.po:
	  Added Bulgarian translation.

2007-09-22 08:12:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* sys/directdraw/gstdirectdrawsink.c:
	* sys/directdraw/gstdirectdrawsink.h:
	  fix header and comments
	  Original commit message from CVS:
	  fix header and comments

2007-09-21 11:34:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpamrdepay.c: Set outgoing packet duration because we can. Fixes #478244 some more.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process):
	  Set outgoing packet duration because we can. Fixes #478244 some more.

2007-09-20 13:35:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/cairo/gsttextoverlay.c: Add info about static leak.
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c:
	  Add info about static leak.
	  * tests/check/Makefile.am:
	  * tests/check/generic/states.c:
	  Improved state change unit test.

2007-09-19 18:19:49 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Ignore registries in any format.
	  Original commit message from CVS:
	  * docs/plugins/.cvsignore:
	  * tests/check/.cvsignore:
	  Ignore registries in any format.

2007-09-19 16:24:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpL16pay.c: Removed some unused code.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer):
	  Removed some unused code.
	  * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
	  * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer):
	  * gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer):
	  * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer):
	  * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet),
	  (gst_rtp_theora_pay_flush_packet):
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet):
	  Try to preserve the incomming buffer duration on the outgoing
	  packets. Fixes #478244.

2007-09-19 10:22:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/: Work around compiler warnings with g++-4.2 when assigning a string constant to a gchar * (partially fixe...
	  Original commit message from CVS:
	  * ext/taglib/gstapev2mux.cc:
	  * ext/taglib/gstid3v2mux.cc:
	  Work around compiler warnings with g++-4.2 when assigning a
	  string constant to a gchar * (partially fixes #478092).

2007-09-18 16:44:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: We require core CVS now for gst_base_src_set_do_timestamp().
	  Original commit message from CVS:
	  * configure.ac:
	  We require core CVS now for gst_base_src_set_do_timestamp().

2007-09-18 13:55:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/: Handling window resize.
	  Original commit message from CVS:
	  * gst/spectrum/demo-audiotest.c:
	  * gst/spectrum/demo-osssrc.c:
	  Handling window resize.

2007-09-18 11:45:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ChangeLog: Add missing newline.
	  Original commit message from CVS:
	  * ChangeLog:
	  Add missing newline.
	  * gst/librfb/rfbdecoder.c:
	  Fix the build (missing stdlib.h).
	  * gst/spectrum/gstspectrum.c:
	  * gst/spectrum/gstspectrum.h:
	  Use basetransform segment so that it is correctly managed on flushes
	  and start/stop. Report message timestamp as stream time, which is what
	  an application can understand. (Yes these are adapted from wim recent
	  level element changes)

2007-09-17 17:35:13 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/: Fix compiler warnings shown with Forte.
	  Original commit message from CVS:
	  * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	  (new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos),
	  (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
	  (gst_rtspsrc_handle_message):
	  Fix compiler warnings shown with Forte.

2007-09-17 02:05:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Give meaningfull error when all streams failed to configure for some reason.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
	  (gst_rtspsrc_dup_printf):
	  Give meaningfull error when all streams failed to configure for some
	  reason.

2007-09-16 19:13:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/README: Update README with the design for synchronisation rules of RTP on sender and receiver.
	  Original commit message from CVS:
	  * gst/rtp/README:
	  Update README with the design for synchronisation rules of RTP on
	  sender and receiver.

2007-09-14 09:40:49 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavparse/gstwavparse.c: Don't push EOS from the chain function, the element driving the pipeline is responsible f...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
	  (gst_wavparse_chain):
	  Don't push EOS from the chain function, the element
	  driving the pipeline is responsible for this. The bug
	  this was meant to fix seems to be queue not forwarding
	  EOS in all cases (see #476514).

2007-09-13 17:31:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/level/gstlevel.*: Use basetransform segment so that it is correctly managed on flushes and start/stop.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
	  (gst_level_transform_ip):
	  * gst/level/gstlevel.h:
	  Use basetransform segment so that it is correctly managed on flushes and
	  start/stop.
	  Report message timestamp as stream time, which is what an application
	  can understand.

2007-09-13 15:04:15 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Update my mail address.
	  Original commit message from CVS:
	  * ext/taglib/gstapev2mux.cc:
	  * ext/taglib/gstapev2mux.h:
	  * ext/taglib/gsttaglibmux.c:
	  * tests/check/elements/apev2mux.c:
	  Update my mail address.

2007-09-13 12:37:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavparse/gstwavparse.c: Add EOS logic for the push-based mode too. Fixes #476514.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
	  (gst_wavparse_loop), (gst_wavparse_chain):
	  Add EOS logic for the push-based mode too. Fixes #476514.

2007-09-12 22:01:59 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/law/: Fix law encoder timestamps.
	  Original commit message from CVS:
	  * gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain):
	  * gst/law/alaw-encode.h:
	  * gst/law/mulaw-encode.c: (gst_mulawenc_init),
	  (gst_mulawenc_chain):
	  * gst/law/mulaw-encode.h:
	  Fix law encoder timestamps.

2007-09-12 09:13:39 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/gconf/gstgconfaudiosink.c: Fix warning when building without debug.
	  Original commit message from CVS:
	  * ext/gconf/gstgconfaudiosink.c:
	  Fix warning when building without debug.
	  * sys/oss/gstossmixertrack.c:
	  Use const like in alsamixertrack.c (fixes warnings).

2007-09-12 08:38:21 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/: Printf format fixes (#476128).
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst-libs/gst/app/gstappsink.c:
	  * gst/flv/gstflvdemux.c:
	  * gst/flv/gstflvparse.c:
	  * gst/interleave/deinterleave.c:
	  * gst/switch/gstswitch.c:
	  Printf format fixes (#476128).

2007-09-11 15:37:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/v4l2src_calls.c: Fix framerate detection code some more.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c:
	  (gst_v4l2src_probe_caps_for_format_and_size):
	  Fix framerate detection code some more.
	  Handle the case where there is a weird step in the stepwise framerates.
	  Don't overwrite the min interval with the framerate, use a temp variable
	  instead.
	  Use max in the Continuous framerate intervals instead of step, which is
	  1 according to the docs. Fixes #475424.

2007-09-10 19:53:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Make udpsrc timestamp outgoing buffers based on when they were received.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
	  Make udpsrc timestamp outgoing buffers based on when they were received.
	  Also make it output a segment in time.

2007-09-10 06:49:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Plug a little leak. Little code cleanups.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Plug a little leak. Little code cleanups.

2007-09-09 18:08:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old flac versions, 's good for cross-compilation ...
	  Original commit message from CVS:
	  * configure.ac:
	  Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old
	  flac versions, 's good for cross-compilation karma.

2007-09-07 18:04:41 +0000  Haakon Sporsheim <haakon.sporsheim@tandberg.com>

	  gst/rtp/gstrtph263pay.c: Fix up header structure so that compilers don't add padding between the structure fields, si...
	  Original commit message from CVS:
	  Patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>
	  * gst/rtp/gstrtph263pay.c:
	  Fix up header structure so that compilers don't add padding
	  between the structure fields, since that would lead to us
	  sending RTP packets with broken headers (as is currently the
	  case when compiling with MSVC). Also see similar fixes in
	  libgstrtp in gst-plugins-base. (#474616; #471194)

2007-09-07 16:04:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/v4l2src_calls.c: Don't overwrite our GValue with 0 but instead use the previously computed value. Fixes #471...
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c:
	  (gst_v4l2src_probe_caps_for_format_and_size):
	  Don't overwrite our GValue with 0 but instead use the previously
	  computed value. Fixes #471823 some more.

2007-09-07 15:54:38 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/spectrum/gstspectrum.c: Use the correct parameter order for the memset calls.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_start),
	  (gst_spectrum_transform_ip):
	  Use the correct parameter order for the memset calls.
	  Thanks to Christian Schaller for noticing.

2007-09-06 12:00:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/gst-plugins-good-plugins.hierarchy: No tabs in this file please, or gtk-doc will end up documenting rath...
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  No tabs in this file please, or gtk-doc will end up documenting
	  rather absurd class hierarchies.

2007-09-06 10:48:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gconf/gstswitchsink.c: If the new kid element fails to change state for some reason forward the error message it ...
	  Original commit message from CVS:
	  * ext/gconf/gstswitchsink.c:
	  If the new kid element fails to change state for some reason
	  (e.g. esdsink not being able to connect to the sound server),
	  forward the error message it posted on the bus instead of just
	  posting a generic 'Internal state change error: please file a
	  bug' error message. Fixes #471364.

2007-09-06 07:21:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, float and double, use floats for the message...
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/spectrum/Makefile.am:
	  * gst/spectrum/demo-audiotest.c: (draw_spectrum),
	  (message_handler), (main):
	  * gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
	  (gst_spectrum_class_init), (gst_spectrum_init),
	  (gst_spectrum_dispose), (gst_spectrum_set_property),
	  (gst_spectrum_get_property), (gst_spectrum_start),
	  (gst_spectrum_setup), (gst_spectrum_message_new),
	  (gst_spectrum_transform_ip):
	  * gst/spectrum/gstspectrum.h:
	  Port GstSpectrum to GstAudioFilter and libgstfft, add support
	  for int32, float and double, use floats for the message contents,
	  average all FFTs done in one interval for better results, use
	  a better windowing function, allow posting the phase in the message
	  and actually do an FFT with the requested number of bands instead
	  of interpolating.
	  * tests/check/elements/spectrum.c: (GST_START_TEST),
	  (spectrum_suite):
	  Improve the units tests by checking for a 11025Hz sine wave
	  and add unit tests for all 4 supported sample types.

2007-09-05 16:23:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/: Don't assume tags are encoded as UTF-8 (#473670).
	  Original commit message from CVS:
	  * gst/qtdemux/Makefile.am:
	  * gst/qtdemux/qtdemux.c:
	  Don't assume tags are encoded as UTF-8 (#473670).

2007-09-05 14:43:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/: Implement LATENCY queries in the crudest way possible so I don't have to use sync=false any longer when te...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c:
	  * sys/v4l2/gstv4l2src.h:
	  * sys/v4l2/v4l2src_calls.c:
	  Implement LATENCY queries in the crudest way possible so I don't
	  have to use sync=false any longer when testing with videosinks.

2007-09-05 09:25:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Fix build.
	  Original commit message from CVS:
	  * configure.ac:
	  Fix build.

2007-09-05 00:12:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/v4l2src_calls.c: Add some more debugging in the framerate function.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c:
	  (gst_v4l2src_probe_caps_for_format_and_size):
	  Add some more debugging in the framerate function.
	  Iterate stepwise framerate up to and _including_ the max and if nothing
	  was added to the list, add a dummy 0/1 to 100/1 framerate so that we
	  don't end up with an empty list.

2007-09-04 22:42:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.c: Add property do configure destination address/port pairs
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	  (gst_multiudpsink_set_clients_string),
	  (gst_multiudpsink_get_clients_string),
	  (gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
	  (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
	  (gst_multiudpsink_add), (gst_multiudpsink_clear_internal),
	  (gst_multiudpsink_clear):
	  Add property do configure destination address/port pairs
	  API:GstMultiUDPSink::clients

2007-09-04 18:30:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/: Added some RTP example scripts for sending and receiving RTP streams.
	  Original commit message from CVS:
	  * tests/examples/Makefile.am:
	  * tests/examples/rtp/Makefile.am:
	  * tests/examples/rtp/client-H263p-AMR.sh:
	  * tests/examples/rtp/client-H263p-PCMA.sdp:
	  * tests/examples/rtp/client-H263p-PCMA.sh:
	  * tests/examples/rtp/client-H264-PCMA.sdp:
	  * tests/examples/rtp/client-H264-PCMA.sh:
	  * tests/examples/rtp/client-PCMA.sh:
	  * tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
	  * tests/examples/rtp/server-alsasrc-PCMA.sh:
	  * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	  * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	  Added some RTP example scripts for sending and receiving RTP streams.

2007-09-04 16:40:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/gstv4l2src.c: Restructure the setcaps function so that we can also compute the expected GStreamer output siz...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2_get_caps_info),
	  (gst_v4l2src_set_caps), (gst_v4l2src_get_mmap):
	  Restructure the setcaps function so that we can also compute the
	  expected GStreamer output size of the video frames.
	  Set frame_byte_size correctly so that read-based devices have a chance
	  of working correctly.
	  When grabbing a frame, discard frames that are not of the expected size.
	  Some cameras don't output the right framesize for the first buffer.
	  Try only a couple of times to get a valid frame, else error out.
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	  (gst_v4l2_fill_lists), (gst_v4l2_get_input):
	  Add some more debug info when scanning the device.
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_new),
	  (gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate),
	  (gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame),
	  (gst_v4l2src_set_capture), (gst_v4l2src_capture_init):
	  Add some more debug info when dequeing a frame.

2007-09-04 14:37:22 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: More code cleanups. Add some more comment and improve debugs logs.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  More code cleanups. Add some more comment and improve debugs logs.

2007-09-04 07:58:36 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.*: Implement seek-query. Refactor duration calculations. Appropriate use of uint64_scale_int...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  * gst/wavparse/gstwavparse.h:
	  Implement seek-query. Refactor duration calculations. Appropriate use
	  of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
	  out of loops.

2007-09-03 07:44:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Implement seek-query.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Implement seek-query.

2007-08-29 21:43:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP packet not wait for preroll.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
	  (gst_rtspsrc_dup_printf):
	  Use new basesink async property to make sparse RTCP packet not wait for
	  preroll.

2007-08-27 14:44:19 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/audiofx/Makefile.am: Dist the right file.
	  Original commit message from CVS:
	  * gst/audiofx/Makefile.am:
	  Dist the right file.

2007-08-23 16:27:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Make sure we generate and parse floating point values in the POSIX locale instead of the curre...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
	  (gst_rtspsrc_get_float), (gst_rtspsrc_play):
	  Make sure we generate and parse floating point values in the POSIX
	  locale instead of the current locale.

2007-08-22 15:01:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Fix method detection again.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
	  (gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	  (gst_rtspsrc_play):
	  * gst/rtsp/gstrtspsrc.h:
	  Fix method detection again.
	  Keep track of when we must send a Range header.
	  Use segment values for Range, Speed and Scale headers.
	  Parse Speed and Scale headers to update the segment values.

2007-08-22 08:22:50 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  sys/v4l2/v4l2src_calls.c: Handle optional v4l2 ioctls gracefully.
	  Original commit message from CVS:
	  patch by: Mark Nauwelaerts <manauw@skynet.be>
	  * sys/v4l2/v4l2src_calls.c:
	  Handle optional v4l2 ioctls gracefully.

2007-08-20 16:52:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added an H263 depayloader. Fixes #369392.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init),
	  (gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init),
	  (gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps),
	  (gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property),
	  (gst_rtp_h263_depay_get_property),
	  (gst_rtp_h263_depay_change_state),
	  (gst_rtp_h263_depay_plugin_init):
	  * gst/rtp/gstrtph263depay.h:
	  Added an H263 depayloader. Fixes #369392.
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	  (gst_rtp_h263p_depay_process):
	  * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	  (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush):
	  Make the H263+ pay/depayloader support H263-1998 and H263-2000
	  payloads.
	  Also alow plain H263 on the h263p payloaders. Fixes #465040.

2007-08-19 19:16:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: Add small comparision with the chebyshev filters in the docs.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c:
	  * gst/filter/gstlpwsinc.c:
	  Add small comparision with the chebyshev filters in the docs.

2007-08-19 19:11:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Add small comparision with the windowed sinc filters in the docs.
	  Original commit message from CVS:
	  * gst/audiofx/audiochebyshevfreqband.c:
	  * gst/audiofx/audiochebyshevfreqlimit.c:
	  Add small comparision with the windowed sinc filters in the docs.

2007-08-19 19:01:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/: Also test everything in 32 bit float mode.
	  Original commit message from CVS:
	  * tests/check/elements/bpwsinc.c: (GST_START_TEST),
	  (bpwsinc_suite):
	  * tests/check/elements/lpwsinc.c: (GST_START_TEST),
	  (lpwsinc_suite):
	  Also test everything in 32 bit float mode.

2007-08-19 18:47:19 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/: Also test 32 bit float mode and the type 2 variants of the filters.
	  Original commit message from CVS:
	  * tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST),
	  (audiochebyshevfreqband_suite):
	  * tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST),
	  (audiochebyshevfreqlimit_suite):
	  Also test 32 bit float mode and the type 2 variants of the filters.

2007-08-18 19:44:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Refactor the udp and interleaved loop function a bit.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	  (gst_rtspsrc_loop):
	  Refactor the udp and interleaved loop function a bit.

2007-08-17 17:08:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
	  (gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
	  (gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
	  (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	  (gst_rtspsrc_try_send), (gst_rtspsrc_pause):
	  * gst/rtsp/gstrtspsrc.h:
	  Protect connection activity with a new lock, avoids deadlocks when going
	  to PAUSED. Fixes #455808.

2007-08-17 15:30:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/debug/rndbuffersize.c: Fix debug statement.
	  Original commit message from CVS:
	  * gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
	  Fix debug statement.

2007-08-17 15:28:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Fix stray %u in debug line as spotted by Saur on IRC.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
	  Fix stray %u in debug line as spotted by Saur on IRC.

2007-08-17 15:05:17 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GOb...
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
	  (bpwsinc_set_property), (bpwsinc_get_property):
	  * gst/filter/gstbpwsinc.h:
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
	  (gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
	  (lpwsinc_get_property):
	  * gst/filter/gstlpwsinc.h:
	  * tests/check/elements/lpwsinc.c: (GST_START_TEST):
	  Use generator macros for the process functions for the different
	  sample types, add lower upper boundaries for the GObject properties
	  so automatically generated UIs can use sliders and change frequency
	  properties to floats to save a bit of memory, even ints would in
	  theory be enough. Also rename frequency to cutoff for consistency
	  reasons.
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-gstrtpmanager.xml:
	  Regenerated for the above changes.

2007-08-17 14:43:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Use generator macros for the process functions for the different sample types, add lower upper boundari...
	  Original commit message from CVS:
	  * gst/audiofx/audiochebyshevfreqband.c:
	  (gst_audio_chebyshev_freq_band_class_init):
	  * gst/audiofx/audiochebyshevfreqlimit.c:
	  (gst_audio_chebyshev_freq_limit_class_init):
	  Use generator macros for the process functions for the different
	  sample types, add lower upper boundaries for the GObject properties
	  so automatically generated UIs can use sliders and add a note about
	  the number of poles as a too high number of poles combined with
	  very low or very high frequencies will produce only noise.
	  * docs/plugins/gst-plugins-good-plugins.args:
	  Regenerated for the property changes.

2007-08-17 14:15:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Improve timeout handling.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
	  (gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
	  (gst_rtspsrc_stream_configure_udp_sink),
	  (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	  (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	  (gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
	  (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
	  (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Improve timeout handling.
	  Use the same socket for sending and receiving RTCP packets so that some
	  servers can track clients better.
	  Improve connection closed handling. Try to reconnect.
	  Don't overwrite our content base with NULL.
	  Improve debugging.
	  Improve range parsing and handling.
	  Remove flushing hack now that core does the right thing.

2007-08-17 13:59:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.*: Add support for getting and setting the socket to use.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	  (gst_multiudpsink_init), (gst_multiudpsink_set_property),
	  (gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
	  (gst_multiudpsink_close), (gst_multiudpsink_add):
	  * gst/udp/gstmultiudpsink.h:
	  Add support for getting and setting the socket to use.
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	  (gst_udpsrc_create), (gst_udpsrc_get_property):
	  Add support for getting the currently used socket.

2007-08-16 19:22:48 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstbpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
	  (gst_bpwsinc_init), (process_32), (process_64),
	  (bpwsinc_build_kernel), (bpwsinc_push_residue),
	  (bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
	  (bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
	  * gst/filter/gstbpwsinc.h:
	  Implement latency query and only forward those samples downstream
	  that actually contain the data we want, i.e. drop kernel_length/2
	  in the beginning and append kernel_length/2 (created by convolving
	  the filter kernel with zeroes) to the end.
	  * tests/check/elements/bpwsinc.c: (GST_START_TEST):
	  Adjust the unit test for this slightly changed behaviour.
	  * gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
	  Reset residue length only when actually creating a residue.

2007-08-16 17:02:07 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
	  Original commit message from CVS:
	  reviewed by: Stefan Kost  <ensonic@users.sf.net>
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiochebyshevfreqband.c:
	  (gst_audio_chebyshev_freq_band_mode_get_type),
	  (gst_audio_chebyshev_freq_band_base_init),
	  (gst_audio_chebyshev_freq_band_dispose),
	  (gst_audio_chebyshev_freq_band_class_init),
	  (gst_audio_chebyshev_freq_band_init),
	  (generate_biquad_coefficients), (calculate_gain),
	  (generate_coefficients),
	  (gst_audio_chebyshev_freq_band_set_property),
	  (gst_audio_chebyshev_freq_band_get_property),
	  (gst_audio_chebyshev_freq_band_setup), (process), (process_64),
	  (process_32), (gst_audio_chebyshev_freq_band_transform_ip),
	  (gst_audio_chebyshev_freq_band_start):
	  * gst/audiofx/audiochebyshevfreqband.h:
	  * gst/audiofx/audiochebyshevfreqlimit.c:
	  (gst_audio_chebyshev_freq_limit_mode_get_type),
	  (gst_audio_chebyshev_freq_limit_base_init),
	  (gst_audio_chebyshev_freq_limit_dispose),
	  (gst_audio_chebyshev_freq_limit_class_init),
	  (gst_audio_chebyshev_freq_limit_init),
	  (generate_biquad_coefficients), (calculate_gain),
	  (generate_coefficients),
	  (gst_audio_chebyshev_freq_limit_set_property),
	  (gst_audio_chebyshev_freq_limit_get_property),
	  (gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
	  (process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
	  (gst_audio_chebyshev_freq_limit_start):
	  * gst/audiofx/audiochebyshevfreqlimit.h:
	  * gst/audiofx/audiofx.c: (plugin_init):
	  Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
	  Fixes #464800.
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/audiochebyshevfreqband.c:
	  (setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
	  (GST_START_TEST), (audiochebyshevfreqband_suite), (main):
	  * tests/check/elements/audiochebyshevfreqlimit.c:
	  (setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
	  (GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
	  Add unit tests for the chebyshev filters.
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-1394.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-shout2send.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  And add docs for the chebyshev filters. While doing
	  that also run make update in docs/plugins.

2007-08-16 12:15:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Make ro memory to share.
	  Original commit message from CVS:
	  * ext/annodex/gstcmmltag.c:
	  * gst/rtp/gstrtpvorbispay.c:
	  Make ro memory to share.

2007-08-16 11:49:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Improve UDP performance by avoiding a select() when we have data available immediatly.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Improve UDP performance by avoiding a select() when we have data
	  available immediatly.

2007-08-16 11:47:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtpdec.*: Add (dummy) SSRC management signals.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
	  (gst_rtp_dec_class_init):
	  * gst/rtsp/gstrtpdec.h:
	  Add (dummy) SSRC management signals.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	  (gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	  (find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
	  (request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
	  (on_timeout), (gst_rtspsrc_stream_configure_manager),
	  (gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
	  (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
	  (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Add connection-speed property.
	  Add find_stream helper functions.
	  Handle stream EOS based on BYE messages or SSRC timeout.
	  Returns SUCCESS from the state change function as we hide our async
	  elements from the parent.

2007-08-16 09:48:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstlpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...
	  Original commit message from CVS:
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
	  (gst_lpwsinc_init), (process_32), (process_64),
	  (lpwsinc_build_kernel), (lpwsinc_push_residue),
	  (lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
	  (lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
	  * gst/filter/gstlpwsinc.h:
	  Implement latency query and only forward those samples downstream
	  that actually contain the data we want, i.e. drop kernel_length/2
	  in the beginning and append kernel_length/2 (created by convolving
	  the filter kernel with zeroes) to the end.
	  * tests/check/elements/lpwsinc.c: (GST_START_TEST):
	  Adjust the unit test for this slightly changed behaviour.

2007-08-16 07:40:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/debug/rndbuffersize.c: Fix da leak.
	  Original commit message from CVS:
	  * gst/debug/rndbuffersize.c:
	  Fix da leak.

2007-08-14 13:50:43 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/debug/: Add new test element and clean-up the others a little.
	  Original commit message from CVS:
	  * gst/debug/Makefile.am:
	  * gst/debug/breakmydata.c:
	  * gst/debug/gstdebug.c:
	  * gst/debug/negotiation.c:
	  * gst/debug/progressreport.c:
	  * gst/debug/rndbuffersize.c:
	  * gst/debug/testplugin.c:
	  Add new test element and clean-up the others a little.

2007-08-13 13:50:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other doc...
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-bz2.xml:
	  * docs/plugins/inspect/plugin-cdxaparse.xml:
	  * docs/plugins/inspect/plugin-dtsdec.xml:
	  * docs/plugins/inspect/plugin-faac.xml:
	  * docs/plugins/inspect/plugin-faad.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-freeze.xml:
	  * docs/plugins/inspect/plugin-gsm.xml:
	  * docs/plugins/inspect/plugin-gstrtpmanager.xml:
	  * docs/plugins/inspect/plugin-h264parse.xml:
	  * docs/plugins/inspect/plugin-modplug.xml:
	  * docs/plugins/inspect/plugin-mpeg2enc.xml:
	  * docs/plugins/inspect/plugin-musepack.xml:
	  * docs/plugins/inspect/plugin-musicbrainz.xml:
	  * docs/plugins/inspect/plugin-nsfdec.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-spcdec.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-tta.xml:
	  * docs/plugins/inspect/plugin-videosignal.xml:
	  * docs/plugins/inspect/plugin-xingheader.xml:
	  * docs/plugins/inspect/plugin-xvid.xml:
	  * gst/filter/gstbpwsinc.c:
	  * gst/filter/gstbpwsinc.h:
	  * gst/filter/gstlpwsinc.c:
	  * gst/filter/gstlpwsinc.h:
	  Add docs for lpwsinc and bpwsinc and integrate them
	  into the build system. While doing that also update
	  all other docs via make update in docs/plugins.

2007-08-12 20:55:01 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/bpwsinc.c: Make one test constraint a bit stricter.
	  Original commit message from CVS:
	  * tests/check/elements/bpwsinc.c: (GST_START_TEST):
	  Make one test constraint a bit stricter.

2007-08-12 20:53:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/: Add unit tests for bpwsinc, testing fundamental functionality again.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/bpwsinc.c: (setup_bpwsinc),
	  (cleanup_bpwsinc), (GST_START_TEST), (bpwsinc_suite), (main):
	  Add unit tests for bpwsinc, testing fundamental functionality again.

2007-08-12 20:19:37 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/: Add unit tests for lpwsinc, testing fundamental functionality.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/lpwsinc.c: (setup_lpwsinc),
	  (cleanup_lpwsinc), (GST_START_TEST), (lpwsinc_suite), (main):
	  Add unit tests for lpwsinc, testing fundamental functionality.

2007-08-12 15:41:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: Improve debugging a bit.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
	  * gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
	  Improve debugging a bit.

2007-08-12 14:35:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Fix parsing of mp4a version 0 atoms. Fixes #465774.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	  Fix parsing of mp4a version 0 atoms. Fixes #465774.

2007-08-12 12:46:20 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: Reset the residue in BaseTransform::start to get a clean residue on stream changes.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
	  (bpwsinc_start):
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
	  (lpwsinc_start):
	  Reset the residue in BaseTransform::start to get a clean residue
	  on stream changes.

2007-08-11 15:58:30 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: Fix processing with buffer sizes that are larger than the filter kernel size.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (process_32), (process_64):
	  * gst/filter/gstlpwsinc.c: (process_32), (process_64):
	  Fix processing with buffer sizes that are larger than the filter
	  kernel size.

2007-08-10 17:08:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/rtp/gstrtpilbcdepay.c: Include stdlib.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpilbcdepay.c:
	  Include stdlib.

2007-08-10 16:10:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmpvdepay.c: Set the mpegversion in the caps so that autoplugging does not get confused.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmpvdepay.c:
	  Set the mpegversion in the caps so that autoplugging does not get
	  confused.

2007-08-10 05:51:40 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstbpwsinc.c: Fix a segfault with more than one channel and don't rebuild the kernel & residue with every ...
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
	  Fix a segfault with more than one channel and don't rebuild
	  the kernel & residue with every buffer.

2007-08-10 05:35:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstbpwsinc.*: Add support for a bandreject mode and allow specifying the window function that should be used.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_mode_get_type),
	  (gst_bpwsinc_window_get_type), (gst_bpwsinc_class_init),
	  (gst_bpwsinc_init), (bpwsinc_build_kernel), (bpwsinc_set_property),
	  (bpwsinc_get_property):
	  * gst/filter/gstbpwsinc.h:
	  Add support for a bandreject mode and allow specifying the window
	  function that should be used.
	  * gst/filter/gstlpwsinc.c:
	  And another small formatting fix.

2007-08-10 05:20:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstbpwsinc.*: Apply the same changes to the bandpass filter:
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
	  (gst_bpwsinc_init), (process_32), (process_64),
	  (bpwsinc_build_kernel), (bpwsinc_setup), (bpwsinc_get_unit_size),
	  (bpwsinc_transform), (bpwsinc_set_property),
	  (bpwsinc_get_property):
	  * gst/filter/gstbpwsinc.h:
	  Apply the same changes to the bandpass filter:
	  - Support double input
	  - Fix processing for input with >1 channels
	  - Specify frequency in Hz
	  - Specify actual filter kernel length
	  - Use transform instead of transform_ip as we're working
	  out of place anyway
	  - Factor out filter kernel generation and update the filter
	  kernel when the properties are set
	  Fix bandpass filter kernel generation to actually generate
	  a bandpass filter by creating a highpass instead of a second
	  lowpass.
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
	  Small formatting fix.

2007-08-10 04:44:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstlpwsinc.*: Specify the actual filter length instead of a weird 2N+1. Setting the property will round to...
	  Original commit message from CVS:
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
	  (gst_lpwsinc_init), (process_32), (process_64),
	  (lpwsinc_build_kernel), (lpwsinc_set_property),
	  (lpwsinc_get_property):
	  * gst/filter/gstlpwsinc.h:
	  Specify the actual filter length instead of a weird
	  2N+1. Setting the property will round to the next odd number.
	  Also remove now obsolete FIXMEs.

2007-08-10 04:32:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstlpwsinc.*: Allow choosing between hamming and blackman window. The blackman window provides a better st...
	  Original commit message from CVS:
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_window_get_type),
	  (gst_lpwsinc_class_init), (gst_lpwsinc_init),
	  (lpwsinc_build_kernel), (lpwsinc_set_property),
	  (lpwsinc_get_property):
	  * gst/filter/gstlpwsinc.h:
	  Allow choosing between hamming and blackman window. The blackman
	  window provides a better stopband attenuation but a bit slower
	  rolloff.

2007-08-10 04:21:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstlpwsinc.*: Add a highpass mode.
	  Original commit message from CVS:
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_mode_get_type),
	  (gst_lpwsinc_class_init), (process_32), (process_64),
	  (lpwsinc_build_kernel), (lpwsinc_set_property),
	  (lpwsinc_get_property):
	  * gst/filter/gstlpwsinc.h:
	  Add a highpass mode.

2007-08-10 04:06:53 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstlpwsinc.c: Fix processing if the input has more than one channel.
	  Original commit message from CVS:
	  * gst/filter/gstlpwsinc.c: (process_32), (process_64),
	  (lpwsinc_build_kernel):
	  Fix processing if the input has more than one channel.

2007-08-09 19:23:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstbpwsinc.c: "this" is a C++ keyword, use "self" instead.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
	  (gst_bpwsinc_init), (bpwsinc_setup), (bpwsinc_transform_ip),
	  (bpwsinc_set_property), (bpwsinc_get_property):
	  "this" is a C++ keyword, use "self" instead.
	  Add TODOs and FIXMEs and remove two wrong FIXMEs.
	  * gst/filter/gstlpwsinc.c:
	  Add FIXMEs and a new TODO.

2007-08-09 18:08:05 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/gstlpwsinc.*: Add double support, replace "this" with "self" as the former is a C++ keyword.
	  Original commit message from CVS:
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
	  (gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32),
	  (process_64), (lpwsinc_build_kernel), (lpwsinc_setup),
	  (lpwsinc_get_unit_size), (lpwsinc_transform),
	  (lpwsinc_set_property), (lpwsinc_get_property):
	  * gst/filter/gstlpwsinc.h:
	  Add double support, replace "this" with "self" as the former
	  is a C++ keyword.
	  Implement the frequency property in Hz instead of fraction
	  of sampling frequency.
	  Remove some unecessary FIXMEs and add some TODOs, add some
	  required locking and refactor the kernel generation into a
	  separate function that is also called when the properties
	  change now.
	  And use BaseTransform::transform instead of transform_ip
	  as the convolution is done out of place anyway. Should
	  be done in place later.

2007-08-09 17:39:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Port the stereo element to GStreamer 0.10.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/stereo/Makefile.am:
	  * gst/stereo/gststereo.c: (gst_stereo_base_init),
	  (gst_stereo_class_init), (gst_stereo_init),
	  (gst_stereo_transform_ip), (gst_stereo_set_property),
	  (gst_stereo_get_property):
	  * gst/stereo/gststereo.h:
	  Port the stereo element to GStreamer 0.10.

2007-08-09 10:54:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/: Updated translations.
	  Original commit message from CVS:
	  * po/hu.po:
	  * po/uk.po:
	  * po/vi.po:
	  Updated translations.

2007-08-08 20:47:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/filter/: Use GstAudioFilter as base class and don't leak the memory of the filter kernel and residue.
	  Original commit message from CVS:
	  * gst/filter/Makefile.am:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
	  (gst_bpwsinc_base_init), (gst_bpwsinc_class_init),
	  (gst_bpwsinc_init), (bpwsinc_setup):
	  * gst/filter/gstbpwsinc.h:
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
	  (gst_lpwsinc_base_init), (gst_lpwsinc_class_init),
	  (gst_lpwsinc_init), (lpwsinc_setup):
	  * gst/filter/gstlpwsinc.h:
	  Use GstAudioFilter as base class and don't leak the memory
	  of the filter kernel and residue.

2007-08-08 17:47:05 +0000  Michael Smith <msmith@xiph.org>

	  gst/videobox/gstvideobox.c: Render right border in the correct location.
	  Original commit message from CVS:
	  * gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
	  Render right border in the correct location.

2007-08-08 10:54:50 +0000  Olivier Crete <tester@tester.ca>

	  gst/rtp/: Make mode property a string. Fixes #464475.
	  Original commit message from CVS:
	  Patch by: Olivier Crete <tester at tester dot ca>
	  * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
	  * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	  Make mode property a string. Fixes #464475.

2007-08-05 14:58:20 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/flac/gstflacenc.c: Widen caps to match decoder a bit and add more FIXMEs.
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c:
	  Widen caps to match decoder a bit and add more FIXMEs.

2007-08-05 14:53:36 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/avi/gstavimux.c: Fix ODML index tag numbering. Fixes #463624.
	  Original commit message from CVS:
	  patch by: Mark Nauwelaerts <manauw@skynet.be>
	  * gst/avi/gstavimux.c:
	  Fix ODML index tag numbering. Fixes #463624.

2007-08-03 16:08:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Fix default clock-rate for realmedia.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
	  (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_stream_configure_tcp),
	  (gst_rtspsrc_stream_configure_udp_sink):
	  Fix default clock-rate for realmedia.
	  Fix parsing of transport.
	  Don't try to link NULL pads.

2007-07-30 17:17:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/POTFILES.skip: Add POTFILES.skip with list of source files that aren't disted at the moment but contain translatab...
	  Original commit message from CVS:
	  * po/POTFILES.skip:
	  Add POTFILES.skip with list of source files that aren't disted at the
	  moment but contain translatable strings. Should hopefully pacify
	  broken tools and make it clearer that these files are left out
	  intentionally (#461600).

2007-07-30 12:41:58 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: If the buffer was entirely clipped ... don't try sending it :)
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
	  If the buffer was entirely clipped ... don't try sending it :)

2007-07-27 16:56:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: If we don't hav a session manager, set the caps on outgoing buffers ourselves.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
	  (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
	  (gst_rtspsrc_create_transports_string),
	  (gst_rtspsrc_prepare_transports):
	  If we don't hav a session manager, set the caps on outgoing buffers
	  ourselves.
	  Force PAUSE/PLAY methods for now until the extensions can overwrite.
	  Append final bit of the transport string even when it does not contain a
	  placeholder.

2007-07-27 11:21:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Clean up the interface list.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
	  (gst_rtsp_ext_list_connect):
	  * gst/rtsp/gstrtspext.h:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
	  Clean up the interface list.
	  Allow connecting to interface signals for the extensions.
	  Remove old extension code.
	  Free list on cleanup.
	  Allow extensions to send additional RTSP messages.

2007-07-27 10:38:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/gconf/gconf.c: Handle a NULL gconf key gracefully by rendering the default element.
	  Original commit message from CVS:
	  * ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	  Handle a NULL gconf key gracefully by rendering the default element.

2007-07-27 10:11:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspext.h: Fix include path for extension interface.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspext.h:
	  Fix include path for extension interface.

2007-07-26 19:45:30 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/audioamplify.h: Also remove a now unecessary variable here.
	  Original commit message from CVS:
	  * gst/audiofx/audioamplify.h:
	  Also remove a now unecessary variable here.

2007-07-26 19:41:07 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Don't save format information ourselves, this is already saved in
	  Original commit message from CVS:
	  * gst/audiofx/audioamplify.c: (gst_audio_amplify_init),
	  (gst_audio_amplify_setup), (gst_audio_amplify_transform_ip):
	  * gst/audiofx/audiodynamic.c:
	  (gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init),
	  (gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip):
	  * gst/audiofx/audiodynamic.h:
	  * gst/audiofx/audioinvert.c: (gst_audio_invert_init),
	  (gst_audio_invert_setup), (gst_audio_invert_transform_ip):
	  * gst/audiofx/audioinvert.h:
	  Don't save format information ourselves, this is already saved in
	  GstAudioFilter.

2007-07-26 15:48:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Use rank to filter out extensions.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
	  (gst_rtsp_ext_list_stream_select):
	  * gst/rtsp/gstrtspext.h:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	  Use rank to filter out extensions.
	  Add url to stream_select interface call.

2007-07-25 18:50:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Use shiny new RTSP and SDP library.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/base64.c:
	  * gst/rtsp/base64.h:
	  * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
	  (gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
	  (gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
	  (gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
	  (gst_rtsp_ext_list_setup_media),
	  (gst_rtsp_ext_list_configure_stream),
	  (gst_rtsp_ext_list_get_transports),
	  (gst_rtsp_ext_list_stream_select):
	  * gst/rtsp/gstrtspext.h:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
	  (gst_rtspsrc_class_init), (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
	  (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
	  (gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
	  (gst_rtspsrc_stream_configure_tcp),
	  (gst_rtspsrc_stream_configure_mcast),
	  (gst_rtspsrc_stream_configure_udp),
	  (gst_rtspsrc_stream_configure_udp_sink),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
	  (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	  (gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
	  (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
	  (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	  (gst_rtspsrc_parse_methods),
	  (gst_rtspsrc_create_transports_string),
	  (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
	  (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
	  (gst_rtspsrc_play), (gst_rtspsrc_pause),
	  (gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
	  * gst/rtsp/gstrtspsrc.h:
	  * gst/rtsp/rtsp.h:
	  * gst/rtsp/rtspconnection.c:
	  * gst/rtsp/rtspconnection.h:
	  * gst/rtsp/rtspdefs.c:
	  * gst/rtsp/rtspdefs.h:
	  * gst/rtsp/rtspext.h:
	  * gst/rtsp/rtspextwms.c:
	  * gst/rtsp/rtspextwms.h:
	  * gst/rtsp/rtspmessage.c:
	  * gst/rtsp/rtspmessage.h:
	  * gst/rtsp/rtsprange.c:
	  * gst/rtsp/rtsprange.h:
	  * gst/rtsp/rtsptransport.c:
	  * gst/rtsp/rtsptransport.h:
	  * gst/rtsp/rtspurl.c:
	  * gst/rtsp/rtspurl.h:
	  * gst/rtsp/sdp.h:
	  * gst/rtsp/sdpmessage.c:
	  * gst/rtsp/sdpmessage.h:
	  * gst/rtsp/test.c:
	  Use shiny new RTSP and SDP library.
	  Implement RTSP extensions using the new interface.
	  Remove a lot of old code.

2007-07-24 14:31:56 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Add codec mapping for '2vuy' (Raw YUV produced by FCP) and 'divx'.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add codec mapping for '2vuy' (Raw YUV produced by FCP) and 'divx'.

2007-07-24 05:07:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackdec.c: Don't unref the outgoing buffer twice when dropping it because it's outside of the segment.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  Don't unref the outgoing buffer twice when dropping it because it's
	  outside of the segment.

2007-07-24 04:57:20 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Use the new buffer clipping function from gstaudio here and require gst-plugins-base CVS.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	  (gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event):
	  Use the new buffer clipping function from gstaudio here and
	  require gst-plugins-base CVS.
	  * tests/check/elements/wavpackdec.c: (GST_START_TEST):
	  For framed Wavpack buffers we require a valid timestamp.

2007-07-23 18:03:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Clip raw audio and video when we can, keep track of current output segment.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	  (gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie),
	  (qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
	  Clip raw audio and video when we can, keep track of current output
	  segment.
	  Don't leak buffers and events when there is no output pad.
	  Improve debugging here and there.

2007-07-23 09:02:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Sync liboil check with plugins-base.
	  Original commit message from CVS:
	  * configure.ac:
	  Sync liboil check with plugins-base.

2007-07-20 11:37:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/: Better algorith for the center frequencies. Subtract band filters from input for negative gains. Rewo...
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  (gst_iir_equalizer_band_set_property),
	  (gst_iir_equalizer_child_proxy_get_child_by_index),
	  (gst_iir_equalizer_child_proxy_get_children_count),
	  (gst_iir_equalizer_child_proxy_interface_init),
	  (gst_iir_equalizer_class_init), (arg_to_scale), (setup_filter),
	  (gst_iir_equalizer_compute_frequencies):
	  * gst/equalizer/gstiirequalizer10bands.c:
	  (gst_iir_equalizer_10bands_class_init):
	  * gst/equalizer/gstiirequalizer3bands.c:
	  (gst_iir_equalizer_3bands_class_init):
	  * gst/equalizer/gstiirequalizernbands.c:
	  Better algorith for the center frequencies. Subtract band filters from
	  input for negative gains. Rework the gain mapping.

2007-07-20 07:41:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/annodex/Makefile.am: Fix CFLAGS/LIBS.
	  Original commit message from CVS:
	  * ext/annodex/Makefile.am:
	  Fix CFLAGS/LIBS.
	  * ext/cdio/gstcdiocddasrc.c:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  Include stdlib
	  * ext/cairo/Makefile.am:
	  * gst/videofilter/Makefile.am:
	  * tests/examples/level/Makefile.am:
	  Use $(LIBM) instead of -lm

2007-07-18 11:55:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/gstv4l2src.c: Add another example pipeline.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c:
	  Add another example pipeline.

2007-07-18 11:42:33 +0000  Alexander Eichner <alexeichi@yahoo.de>

	  sys/v4l2/gstv4l2src.c: Use define here.
	  Original commit message from CVS:
	  Patch by: Alexander Eichner <alexeichi@yahoo.de>
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  Use define here.
	  * sys/v4l2/gstv4l2tuner.c:
	  (gst_v4l2_tuner_set_frequency_and_notify):
	  Don't touch the property - its still disabled.
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format),
	  (gst_v4l2src_grab_frame), (gst_v4l2src_get_size_limits):
	  * sys/v4l2/v4l2src_calls.h:
	  Improve fallback format negotionation. Fixes #451388

2007-07-18 10:33:39 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/videocrop.c: Fix the test.
	  Original commit message from CVS:
	  * tests/check/elements/videocrop.c: (GST_START_TEST):
	  Fix the test.

2007-07-18 09:21:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  More docs. More logs in pngdec.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * ext/jpeg/gstjpegdec.c:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_task),
	  (gst_pngdec_sink_setcaps):
	  More docs. More logs in pngdec.

2007-07-18 07:51:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/multifile/gstmultifilesrc.c: Add example to the docs. Fix buffer-offset-end and add some debug.
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
	  Add example to the docs. Fix buffer-offset-end and add some debug.

2007-07-18 07:35:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add stdlib include (free, atoi, exit).
	  Original commit message from CVS:
	  * examples/app/appsrc_ex.c:
	  * examples/switch/switcher.c:
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/timidity/gstwildmidi.c:
	  * ext/x264/gstx264enc.c:
	  * gst/mve/mveaudioenc.c: (mve_compress_audio):
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/spectrum/demo-audiotest.c:
	  * gst/spectrum/demo-osssrc.c:
	  * sys/dvb/gstdvbsrc.c:
	  Add stdlib include (free, atoi, exit).

2007-07-17 11:35:29 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/gstv4l2src.c: Initialize num_buffers with minimum value.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  Initialize num_buffers with minimum value.
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame):
	  Handle frame-size query failure gracefully.

2007-07-16 12:11:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Fix parsing of esds atoms inside mp4a atoms so that we can set correct codec_info for AAC audi...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	  Fix parsing of esds atoms inside mp4a atoms so that we can set correct
	  codec_info for AAC audio. Fixes #457097 along with a whole other bunch
	  of qt/aac files.

2007-07-16 09:16:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackdec.c: Fix buffer clipping to correctly clip to the segment stop.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c:
	  (gst_wavpack_dec_clip_outgoing_buffer):
	  Fix buffer clipping to correctly clip to the segment stop.

2007-07-13 16:31:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
	  Original commit message from CVS:
	  * configure.ac:
	  * tests/Makefile.am:
	  Remove bogus check for libcheck, since we check for
	  gstreamer-check and it pulls in the required info from there,
	  and we weren't actually _using_ the information for libcheck
	  ourselves anyway.

2007-07-12 11:21:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Use pkg-config to locate check.
	  Original commit message from CVS:
	  * configure.ac:
	  Use pkg-config to locate check.

2007-07-11 23:43:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Fix build against core CVS.
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c: (gst_deinterleave_process):
	  * gst/vmnc/vmncdec.c: (vmnc_make_buffer):
	  Fix build against core CVS.

2007-07-11 22:31:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Fix build against core CVS.
	  Original commit message from CVS:
	  * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	  * ext/libpng/gstpngenc.c: (gst_pngenc_chain):
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  * gst/debug/gstnavigationtest.c: (gst_navigationtest_transform):
	  * gst/effectv/gstaging.c: (gst_agingtv_transform):
	  * gst/effectv/gstdice.c: (gst_dicetv_transform):
	  * gst/effectv/gstedge.c: (gst_edgetv_transform):
	  * gst/effectv/gstquark.c: (gst_quarktv_transform):
	  * gst/effectv/gstrev.c: (gst_revtv_transform):
	  * gst/effectv/gstshagadelic.c: (gst_shagadelictv_transform):
	  * gst/effectv/gstvertigo.c: (gst_vertigotv_transform):
	  * gst/effectv/gstwarp.c: (gst_warptv_transform):
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_add_wvpk_header),
	  (gst_matroska_demux_check_subtitle_buffer),
	  (gst_matroska_decode_buffer):
	  * gst/videofilter/gstvideoflip.c: (gst_video_flip_transform):
	  Fix build against core CVS.

2007-07-10 10:16:38 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/id3demux/gstid3demux.c: Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We don't have enough gra...
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We
	  don't have enough granularity to convert that boolean into a
	  GstFlowReturn.

2007-07-06 15:00:47 +0000  Michael Smith <msmith@xiph.org>

	  gst/law/: Fix capsnego bogosity in *law decoders.
	  Original commit message from CVS:
	  * gst/law/alaw-decode.c: (alawdec_sink_setcaps),
	  (gst_alawdec_class_init), (gst_alawdec_init), (gst_alawdec_chain),
	  (gst_alawdec_change_state):
	  * gst/law/alaw-decode.h:
	  * gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
	  (gst_mulawdec_class_init), (gst_mulawdec_init),
	  (gst_mulawdec_chain), (gst_mulawdec_change_state):
	  * gst/law/mulaw-decode.h:
	  Fix capsnego bogosity in *law decoders.

2007-07-06 14:35:59 +0000  Michael Smith <msmith@xiph.org>

	  ext/jpeg/gstsmokeenc.*: Remove stupidity in get/set caps functions.
	  Original commit message from CVS:
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init),
	  (gst_smokeenc_setcaps), (gst_smokeenc_chain),
	  (gst_smokeenc_change_state):
	  * ext/jpeg/gstsmokeenc.h:
	  Remove stupidity in get/set caps functions.
	  Fix some refcounting problems.

2007-07-06 11:42:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/libpng/gstpngdec.c: Remove endianness-flipping hack that seems to have been required only because of a bug in ffm...
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
	  Remove endianness-flipping hack that seems to have been required
	  only because of a bug in ffmpegcolorspace.
	  Partially Fixes: #451908

2007-07-05 08:44:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/Makefile.am: Simplify --extra-dir as gtkdoc scans recursively.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  Simplify --extra-dir as gtkdoc scans recursively.

2007-07-03 09:59:46 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/rtp/gstrtpilbcpay.c: Set the encoding-name in the rtp caps to all uppercase, as required by the caps spec.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	  Set the encoding-name in the rtp caps to all uppercase, as required by
	  the caps spec.
	  Some small cleanups in the error paths. Fixes #453037.

2007-07-03 08:01:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/multifile/: Add .h files to be able to add it to the docs.
	  Original commit message from CVS:
	  * gst/multifile/Makefile.am:
	  * gst/multifile/gstmultifile.c:
	  * gst/multifile/gstmultifilesink.c:
	  * gst/multifile/gstmultifilesink.h:
	  * gst/multifile/gstmultifilesrc.c:
	  * gst/multifile/gstmultifilesrc.h:
	  Add .h files to be able to add it to the docs.

2007-07-03 07:16:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/replaygain/gstrgvolume.h: Fix GObject macros.
	  Original commit message from CVS:
	  * gst/replaygain/gstrgvolume.h:
	  Fix GObject macros.

2007-06-28 19:00:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.*: Use a GSList for the GArray that is used like a list anyway.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_index_get_last_entry),
	  (gst_wavpack_parse_index_get_entry_from_sample),
	  (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
	  (gst_wavpack_parse_scan_to_find_sample):
	  * ext/wavpack/gstwavpackparse.h:
	  Use a GSList for the GArray that is used like a list anyway.

2007-06-28 13:25:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gdk_pixbuf/gstgdkpixbuf.c: Add state change function where we set 0/1 as default framerate in case our setcaps fu...
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
	  (gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_flush),
	  (gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_change_state):
	  Add state change function where we set 0/1 as default framerate in
	  case our setcaps function isn't called, like it might not in a
	  filesrc ! gdkpixbufdec scenario. Fixes assertion triggered by
	  gdkpixbufdec trying to create caps with a 0/0 framerate.
	  Also post an error message on the bus if gst_pad_push() fails when
	  called from our sink event handler (+1 for flow returns for event
	  functions in 0.11) instead of failing silently.

2007-06-27 11:36:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Cast stack args to the proper types. Fixes #451249.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps):
	  Cast stack args to the proper types. Fixes #451249.

2007-06-27 11:04:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: For container formats we only need to activate one of the streams so that we correctly signal ...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	  (new_session_pad), (gst_rtspsrc_setup_streams):
	  * gst/rtsp/gstrtspsrc.h:
	  For container formats we only need to activate one of the streams so
	  that we correctly signal no-more-pads. Fixes #451015.

2007-06-25 12:46:08 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: Update docs with caps info.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-ladspa.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update docs with caps info.

2007-06-25 12:13:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/POTFILES.in: Add more files with translatable strings (#450878).
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  Add more files with translatable strings (#450878).

2007-06-22 20:23:18 +0000  Jens Granseuer <jensgr@gmx.net>

	  gst/: Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
	  Original commit message from CVS:
	  Patch by: Jens Granseuer  <jensgr at gmx net>
	  * gst/equalizer/gstiirequalizer.c:
	  * gst/equalizer/gstiirequalizer10bands.c:
	  * gst/equalizer/gstiirequalizer3bands.c:
	  * gst/equalizer/gstiirequalizernbands.c:
	  * gst/rtpmanager/async_jitter_queue.c:
	  (async_jitter_queue_push_sorted):
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  (gst_rtp_jitter_buffer_chain):
	  * gst/switch/gstswitch.c: (gst_switch_chain):
	  Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
	  Fixes #450185.

2007-06-22 14:26:36 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  MAINTAINERS: Updating all the maintainers files
	  Original commit message from CVS:
	  * MAINTAINERS:
	  Updating all the maintainers files

2007-06-22 10:12:15 +0000  Edward Hervey <bilboed@bilboed.com>

	  Fix memory leaks.
	  Original commit message from CVS:
	  * ext/flac/gstflactag.c: (gst_flac_tag_init):
	  * gst/interleave/deinterleave.c: (deinterleave_init),
	  (deinterleave_sink_link):
	  * gst/interleave/interleave.c: (interleave_init):
	  * gst/median/gstmedian.c: (gst_median_init):
	  * gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init):
	  Fix memory leaks.
	  * tests/check/elements/id3demux.c: (pad_added_cb):
	  Remove unused variable.

2007-06-21 10:48:10 +0000  Damien Carbery <damien.carbery@sun.com>

	  ext/gconf/gconf.h: Make the prototype of gst_gconf_get_key_for_sink_profile match the implementation.
	  Original commit message from CVS:
	  * ext/gconf/gconf.h:
	  Make the prototype of gst_gconf_get_key_for_sink_profile
	  match the implementation.
	  Patch by: Damien Carbery <damien dot carbery at sun dot com>
	  Fixes: #449747

2007-06-20 12:56:12 +0000  Michael Smith <msmith@xiph.org>

	  gst/rtp/gstrtpdepay.c: Fix description - rtpdepay is not a payloader.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpdepay.c:
	  Fix description - rtpdepay is not a payloader.

2007-06-20 10:15:00 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/gstiirequalizer.c: Document parameter mapping.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c:
	  Document parameter mapping.

2007-06-20 08:56:17 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.c: Fix leaking buffers.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_event),
	  (gst_spectrum_transform_ip):
	  Fix leaking buffers.
	  * tests/check/Makefile.am:
	  * tests/check/elements/spectrum.c: (setup_spectrum),
	  (cleanup_spectrum), (GST_START_TEST), (spectrum_suite), (main):
	  Add simple test for spectrum element.

2007-06-20 08:26:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/: Add MJPG to the variants of motion jpeg.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_samples),
	  (qtdemux_video_caps):
	  * gst/qtdemux/qtdemux_fourcc.h:
	  Add MJPG to the variants of motion jpeg.

2007-06-19 16:40:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the error flags are included and it errors...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/audiopanorama.c: (GST_START_TEST):
	  * tests/check/elements/videocrop.c: (GST_START_TEST):
	  * tests/check/elements/videofilter.c:
	  * tests/check/elements/wavpackdec.c: (GST_START_TEST):
	  * tests/check/elements/wavpackparse.c: (GST_START_TEST):
	  Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the
	  error flags are included and it errors out on compiler warnings
	  for CVS builds; remove unused variables in various unit tests.

2007-06-19 14:48:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspconnection.c: Use threadsafe inet_ntop to convert an ip number to a string.
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	  (rtsp_connection_close), (rtsp_connection_free):
	  Use threadsafe inet_ntop to convert an ip number to a string.
	  Fixes #447961.
	  Don't leak fd (and ip) when freeing a connection without first closing
	  it.

2007-06-19 14:11:49 +0000  Christian Schaller <uraeus@gnome.org>

	* gst/qtdemux/LEGAL:
	  add 'LEGAL' file describing why this is in -good and under what circumstances it might need to move.
	  Original commit message from CVS:
	  add 'LEGAL' file describing why this is in -good and under what
	  circumstances it might need to move.

2007-06-19 10:41:49 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS
	  * gst-plugins-good.doap:
	  Add 0.10.6 to the doap file.

=== release 0.10.6 ===

2007-06-19 10:24:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* win32/common/config.h:
	  Release 0.10.6
	  Original commit message from CVS:
	  Release 0.10.6

2007-06-18 17:53:20 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/ja.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2007-06-17 12:35:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtsp/rtspconnection.c: Revert previous commit again, since we are frozen (sorry).
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	  (rtsp_connection_free):
	  Revert previous commit again, since we are frozen (sorry).

2007-06-17 12:24:58 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtsp/rtspconnection.c: inet_ntoa() uses a static buffer internally, so we need to copy the returned string if we ...
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt <pkj at axis com>
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	  (rtsp_connection_free):
	  inet_ntoa() uses a static buffer internally, so we need to copy the
	  returned string if we want to store it for later (#447961).

2007-06-15 09:13:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  win32/vs6/: Mark *.dsp & *.dsw as binary files and convert to DOS line endings, as they don't load into VS6 correctly...
	  Original commit message from CVS:
	  * win32/vs6/autogen.dsp:
	  * win32/vs6/gst_plugins_good.dsw:
	  * win32/vs6/libgstalaw.dsp:
	  * win32/vs6/libgstalpha.dsp:
	  * win32/vs6/libgstalphacolor.dsp:
	  * win32/vs6/libgstapetag.dsp:
	  * win32/vs6/libgstaudiofx.dsp:
	  * win32/vs6/libgstauparse.dsp:
	  * win32/vs6/libgstautodetect.dsp:
	  * win32/vs6/libgstavi.dsp:
	  * win32/vs6/libgstcutter.dsp:
	  * win32/vs6/libgstdirectdraw.dsp:
	  * win32/vs6/libgstdirectsound.dsp:
	  * win32/vs6/libgsteffectv.dsp:
	  * win32/vs6/libgstflx.dsp:
	  * win32/vs6/libgstgoom.dsp:
	  * win32/vs6/libgsticydemux.dsp:
	  * win32/vs6/libgstid3demux.dsp:
	  * win32/vs6/libgstinterleave.dsp:
	  * win32/vs6/libgstjpeg.dsp:
	  * win32/vs6/libgstlevel.dsp:
	  * win32/vs6/libgstmatroska.dsp:
	  * win32/vs6/libgstmedian.dsp:
	  * win32/vs6/libgstmonoscope.dsp:
	  * win32/vs6/libgstmulaw.dsp:
	  * win32/vs6/libgstmultipart.dsp:
	  * win32/vs6/libgstqtdemux.dsp:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgstrtsp.dsp:
	  * win32/vs6/libgstsmpte.dsp:
	  * win32/vs6/libgstspeex.dsp:
	  * win32/vs6/libgstudp.dsp:
	  * win32/vs6/libgstvideobalance.dsp:
	  * win32/vs6/libgstvideobox.dsp:
	  * win32/vs6/libgstvideocrop.dsp:
	  * win32/vs6/libgstvideoflip.dsp:
	  * win32/vs6/libgstvideomixer.dsp:
	  * win32/vs6/libgstwaveform.dsp:
	  * win32/vs6/libgstwavenc.dsp:
	  * win32/vs6/libgstwavparse.dsp:
	  Mark *.dsp & *.dsw as binary files and convert to DOS line
	  endings, as they don't load into VS6 correctly otherwise.

2007-06-15 08:32:52 +0000  Vincent Torri <vtorri@univ-evry.fr>

	  gst/rtsp/rtspconnection.c: Fix the MingW build.
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	  (rtsp_connection_connect):
	  Fix the MingW build.
	  Patch By: Vincent Torri <vtorri at univ-evry dot fr>
	  Fixes: #446981

2007-06-14 14:03:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/: Hush the buildbots up
	  Original commit message from CVS:
	  * tests/check/elements/.cvsignore:
	  * tests/icles/.cvsignore:
	  Hush the buildbots up

2007-06-14 12:14:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Make sure to dist everything needed for win32 builds.
	  Original commit message from CVS:
	  * configure.ac:
	  * sys/Makefile.am:
	  * sys/directdraw/Makefile.am:
	  * sys/directsound/Makefile.am:
	  * sys/waveform/Makefile.am:
	  Make sure to dist everything needed for win32 builds.

2007-06-14 10:23:20 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: For AMR-NB streams, export the AMRSpecificBox as codec_data on the caps.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  For AMR-NB streams, export the AMRSpecificBox as codec_data on the
	  caps.
	  Fixes #447458

2007-06-13 17:11:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph264depay.c: Make sure we allocate enough memory for the codec_data.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	  Make sure we allocate enough memory for the codec_data.
	  Fixes #447210.

2007-06-12 21:05:22 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/MANIFEST: Add videocrop project file to the win32 manifest.
	  Original commit message from CVS:
	  * win32/MANIFEST:
	  Add videocrop project file to the win32 manifest.
	  * win32/vs6/gst_plugins_good.dsw:
	  Add qtdemux,videocrop and waveform projects to the workspace.
	  * win32/vs6/libgstqtdemux.dsp:
	  Add zlib to the link list of qtdemux.
	  * win32/vs6/libgstvideocrop.dsp:
	  Add a project file for videocrop.

2007-06-12 20:22:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  po/POTFILES.in: Add qtdemux for translation
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  Add qtdemux for translation

2007-06-12 20:15:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move videocrop and osxvideo from -bad.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * gst-plugins-good.spec.in:
	  * sys/Makefile.am:
	  * tests/check/Makefile.am:
	  * tests/icles/Makefile.am:
	  * tests/icles/videocrop-test.c:
	  Move videocrop and osxvideo from -bad.

2007-06-12 19:35:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move qtdemux from -bad.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-qtdemux.xml:
	  * docs/plugins/inspect/plugin-quicktime.xml:
	  * win32/MANIFEST:
	  Move qtdemux from -bad.
	  * gst-plugins-good.spec.in:
	  Update spec file to reflect moving of qtdemux and wavpack

2007-06-12 19:01:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* win32/MANIFEST:
	  Fix typo in the changelog and commit the manifest too
	  Original commit message from CVS:
	  Fix typo in the changelog and commit the manifest too

2007-06-12 18:52:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  win32/MANIFEST
	  Original commit message from CVS:
	  * win32/MANIFEST
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-directdraw.xml:
	  * docs/plugins/inspect/plugin-directsound.xml:
	  * docs/plugins/inspect/plugin-waveform.xml:
	  Move the waveform plugin from -bad too. Update the inspect xml
	  files to mention Plugins Good instead of Plugins Bad.

2007-06-12 13:33:56 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* sys/v4l2/v4l2src_calls.c:
	  Return a copy of the pool buffer if all mmap buffers have been dequeued.
	  Original commit message from CVS:
	  (gst_v4l2src_grab_frame): Return a copy of the pool buffer if all
	  mmap buffers have been dequeued.

2007-06-12 11:23:01 +0000  Andy Wingo <wingo@pobox.com>

	  sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize) (gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
	  Original commit message from CVS:
	  2007-06-12  Andy Wingo  <wingo@pobox.com>
	  * sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize)
	  (gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
	  (gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with
	  finalization and resuscitation. No longer public.
	  (gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init)
	  (gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type)
	  (gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate)
	  (gst_v4l2_buffer_pool_destroy): Make the pool follow common
	  miniobject semantics, and be threadsafe.
	  (gst_v4l2src_queue_frame): Remove this function, as we just call
	  the ioctls directly in the two places where we queue buffers.
	  (gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer
	  directly.
	  (gst_v4l2src_capture_init): Use the new buffer_pool_new function
	  to allocate the pool, which also preallocates the GstBuffers.
	  (gst_v4l2src_capture_start): Call buffer_pool_activate instead of
	  queueing the frames directly.
	  * sys/v4l2/gstv4l2src.h (struct _GstV4l2BufferPool): Make this a
	  real MiniObject instead of rolling our own refcounting and
	  finalizing. Give it a lock.
	  (struct _GstV4l2Buffer): Remove one intermediary object, having
	  the buffers hold the struct v4l2_buffer directly.
	  * sys/v4l2/gstv4l2src.c (gst_v4l2src_set_caps): Pass the caps to
	  capture_init so that it can set them on the buffers that it will
	  create.
	  (gst_v4l2src_get_read): For better or for worse, include the
	  timestamping and offsetting code here; really we should be using
	  bufferalloc though.
	  (gst_v4l2src_get_mmap): Just make grab_frame return one of our
	  preallocated, mmap'd buffers.

2007-06-11 11:41:56 +0000  daniel fischer <dan@f3c.com>

	  sys/ximage/gstximagesrc.c: Actually use the display_name property so that we can dump any available X display. Fixes ...
	  Original commit message from CVS:
	  Patch by: daniel fischer <dan at f3c dot com>
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_start),
	  (gst_ximage_src_get_caps):
	  Actually use the display_name property so that we can dump any
	  available X display. Fixes #445905.

2007-06-11 10:21:13 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/rtp/: Add missing rate fields to caps. Fixes #441118.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps):
	  * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps):
	  Add missing rate fields to caps. Fixes #441118.

2007-06-10 21:14:11 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/: Add DirectSound and DirectDraw sinks project files to workspace and solution files.
	  Original commit message from CVS:
	  * win32/vs6/gst_plugins_good.dsw:
	  * win32/vs8/gst-plugins-good.sln:
	  Add DirectSound and DirectDraw sinks project files to
	  workspace and solution files.

2007-06-10 10:53:26 +0000  Josh Coalson <xflac@yahoo.com>

	  Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.
	  Original commit message from CVS:
	  Patch by: Josh Coalson <xflac at yahoo dot com>,
	  updated by Alexis Ballier <aballier at gentoo dot org>:
	  * configure.ac:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
	  (gst_flac_dec_setup_seekable_decoder),
	  (gst_flac_dec_setup_stream_decoder), (gst_flac_dec_seek),
	  (gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof),
	  (gst_flac_dec_read_seekable), (gst_flac_dec_read_stream):
	  * ext/flac/gstflacdec.h:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_init),
	  (gst_flac_enc_finalize), (gst_flac_enc_set_metadata),
	  (gst_flac_enc_sink_setcaps), (gst_flac_enc_update_quality),
	  (gst_flac_enc_seek_callback), (gst_flac_enc_write_callback),
	  (gst_flac_enc_tell_callback), (gst_flac_enc_sink_event),
	  (gst_flac_enc_chain), (gst_flac_enc_set_property),
	  (gst_flac_enc_get_property), (gst_flac_enc_change_state):
	  * ext/flac/gstflacenc.h:
	  Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.

2007-06-09 15:41:52 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.c: Remove workaround for bug #421543. This is fixed in core 0.10.13 and not necessary anymo...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps):
	  Remove workaround for bug #421543. This is fixed in core 0.10.13 and
	  not necessary anymore as we need at least that core version.

2007-06-09 15:33:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: Improve discont handling by checking if the next Wavpack block has the expected, following block index.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	  (gst_wavpack_dec_chain):
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
	  (gst_wavpack_parse_push_buffer):
	  * ext/wavpack/gstwavpackparse.h:
	  Improve discont handling by checking if the next Wavpack block has
	  the expected, following block index.

2007-06-08 20:23:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/.gitignore:
	  moap ignore
	  Original commit message from CVS:
	  moap ignore

2007-06-08 20:20:56 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details): Fix element description.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details):
	  Fix element description.

2007-06-08 20:19:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  move wavpack plugin.  See #352605.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/gst-plugins-good-plugins.signals:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-ladspa.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * ext/Makefile.am:
	  * tests/check/Makefile.am:
	  move wavpack plugin.  See #352605.

2007-06-08 19:45:43 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/Makefile.am:
	  the alphabet tripping up people since 10929BC
	  Original commit message from CVS:
	  the alphabet
	  tripping up people since 10929BC

2007-06-08 17:37:02 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Add DirectDraw & DirectSound plugins to the build and docs.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * sys/Makefile.am:
	  * win32/MANIFEST:
	  Add DirectDraw & DirectSound plugins to the build and docs.

2007-06-08 16:31:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Rename the keep-aspect-ratio property to force-aspect-ratio to make it consistent with xvimagesink and ximagesink.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * sys/directdraw/gstdirectdrawsink.c:
	  (gst_directdraw_sink_class_init):
	  Rename the keep-aspect-ratio property to force-aspect-ratio to make
	  it consistent with xvimagesink and ximagesink.

2007-06-08 10:43:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/: When operating in pull mode, error out correct on not-linked.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
	  * ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task):
	  When operating in pull mode, error out correct on not-linked.

2007-06-08 08:12:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/videocrop-test.c: Default to xvimagesink instead of autovideosink while autovideosink/ghostpads/whatever ...
	  Original commit message from CVS:
	  * tests/icles/videocrop-test.c: (main):
	  Default to xvimagesink instead of autovideosink while
	  autovideosink/ghostpads/whatever don't handle the way we use it in
	  the way we expect it to.

2007-06-06 10:19:17 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* sys/v4l2/v4l2src_calls.c:
	  sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
	  Original commit message from CVS:
	  2007-06-06  Andy Wingo  <wingo@pobox.com>
	  * sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
	  (gst_v4l2src_probe_caps_for_format_and_size): Only probe for
	  format and size if the ioctls are defined; should fix compilation
	  on Linux < 2.16.19.

2007-06-06 08:53:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videobox/gstvideobox.c: Printf fixes in debug statements; use LOG level for debug statements that are printed for...
	  Original commit message from CVS:
	  * gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
	  Printf fixes in debug statements; use LOG level for debug statements
	  that are printed for each and every frame; convert c++ comments to
	  C-style comments; not much point using g_try_malloc() if we then not
	  even check the return value.

2007-06-05 16:32:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump requirements to released versions (core and base 0.10.13).
	  Original commit message from CVS:
	  * configure.ac:
	  Bump requirements to released versions (core and base 0.10.13).
	  * gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify):
	  Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
	  own implementation.

2007-06-05 14:17:25 +0000  Andy Wingo <wingo@pobox.com>

	  sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add some useless comments.
	  Original commit message from CVS:
	  2007-06-05  Andy Wingo  <wingo@pobox.com>
	  * sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add
	  some useless comments.
	  * sys/v4l2/v4l2src_calls.c (gst_v4l2src_capture_init): Don't queue
	  frames before calling STREAMON, that might leave them in a state
	  where they can't be dequeued if we go back to NULL without calling
	  STREAMON, according to the docs.
	  (gst_v4l2src_capture_start): Enqueue buffers here instead, right
	  before we call STREAMON.
	  (gst_v4l2src_capture_deinit): Remove crack to work around dequeue
	  failures. (For me this code hung.) The pool refcounting is still
	  crack; added a note to that effect.

2007-06-05 09:11:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/multipart/multipartmux.c: Add support for mapping gst structure names to the MIME type equivalent.
	  Original commit message from CVS:
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
	  (gst_multipart_mux_get_mime), (gst_multipart_mux_collected):
	  Add support for mapping gst structure names to the MIME type equivalent.
	  Implemented for audio/x-mulaw->audio/basic. Fixes #442874.

2007-06-03 11:21:44 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavenc/gstwavenc.*: Properly write wav files with width!=depth by having the depth most significant bytes set and...
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	  (gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),
	  (gst_wavenc_chain), (gst_wavenc_change_state):
	  * gst/wavenc/gstwavenc.h:
	  Properly write wav files with width!=depth by having the depth most
	  significant bytes set and all others zero. Fixes #442535.

2007-06-01 13:52:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspconnection.c: Add include to make buildbot happy.
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c:
	  Add include to make buildbot happy.

2007-06-01 13:07:11 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtsp/: Improves version checking, allowing an RTSP server to reply with "505
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	  (rtsp_connection_connect), (add_date_header),
	  (rtsp_connection_send), (parse_response_status),
	  (parse_request_line), (parse_line), (rtsp_connection_receive):
	  * gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
	  * gst/rtsp/rtspdefs.h:
	  * gst/rtsp/rtspmessage.c: (key_value_foreach),
	  (rtsp_message_init_request), (rtsp_message_init_response),
	  (rtsp_message_remove_header), (rtsp_message_append_headers),
	  (rtsp_message_dump):
	  * gst/rtsp/rtspmessage.h:
	  Improves version checking, allowing an RTSP server to reply with "505
	  RTSP Version not supported.
	  Adds a Date header to all messages.
	  Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
	  want to be able to send a response even if something in the request was
	  invalid. EINVAL is only used when passing wrong arguments to functions.
	  Do not handle an invalid method in parse_request_line(). Defer this to
	  the caller so it can respond with "405 Method Not Allowed".
	  Improves parsing of the timeout parameter to the Session header,
	  allowing whitespace after the semicolon.
	  Avoids a compiler warning due to variables shadowing a function argument.

2007-06-01 11:16:17 +0000  Daniel Charles <dcharles@ti.com>

	  gst/rtp/: Add support for AMR-WB.
	  Original commit message from CVS:
	  Based on Patch by: Daniel Charles <dcharles at ti dot com>
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	  (gst_rtp_amr_depay_process):
	  * gst/rtp/gstrtpamrdepay.h:
	  * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
	  (gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
	  (gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
	  * gst/rtp/gstrtpamrpay.h:
	  Add support for AMR-WB.
	  Small cleanups such as using BOILERPLATE.

2007-05-31 15:57:07 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspextwms.c: Fix compile warning when debug is disabled as spotted bu Saur on IRC.
	  Original commit message from CVS:
	  * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
	  Fix compile warning when debug is disabled as spotted bu Saur on IRC.

2007-05-30 14:57:44 +0000  Andy Wingo <wingo@pobox.com>

	  sys/v4l2/gstv4l2object.*: Revert some unintended changes.
	  Original commit message from CVS:
	  2007-05-30  Andy Wingo  <wingo@pobox.com>
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/gstv4l2object.c (gst_v4l2_object_new): Revert some
	  unintended changes.

2007-05-30 14:40:53 +0000  Andy Wingo <wingo@pobox.com>

	  sys/v4l2/v4l2src_calls.*: Store the format list in the order that the driver gives it to us.
	  Original commit message from CVS:
	  2007-05-30  Andy Wingo  <wingo@pobox.com>
	  * sys/v4l2/v4l2src_calls.h:
	  * sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
	  the format list in the order that the driver gives it to us.
	  (gst_v4l2src_probe_caps_for_format_and_size)
	  (gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
	  based on the capabilities of the device.
	  (gst_v4l2src_grab_frame): Update for object variable renaming.
	  (gst_v4l2src_set_capture): Update to be strict in its parameters,
	  as in the set_caps below.
	  (gst_v4l2src_capture_init): Update for object variable renaming,
	  and reflow.
	  (gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
	  (gst_v4l2src_capture_deinit): Update for object variable renaming.
	  (gst_v4l2src_update_fps, gst_v4l2src_set_fps)
	  (gst_v4l2src_get_fps): Remove; these functions don't have much
	  meaning outside of an atomic set_caps method.
	  (gst_v4l2src_buffer_new): Don't set buffer duration, it is not
	  known.
	  * sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
	  call to update_fps; not sure about this change.
	  (gst_v4l2_tuner_set_norm): Work around the fact that for the
	  moment we don't have an update_fps_func.
	  * sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
	  structures in the object, just store what we need. Do store the
	  probed caps of the device. Don't store the current frame rate.
	  * sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
	  update_fps_function, for now. Update for new object variable
	  naming.
	  (gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
	  new object variable naming.
	  (gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
	  (gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
	  (gst_v4l2src_get_caps): Rework to probe the device for supported
	  frame sizes and frame rates.
	  (gst_v4l2src_set_caps): Rework to be strict in the given
	  parameters: if someone asks us to have a certain size and rate,
	  that is what we configure.
	  (gst_v4l2src_get_read): Update for object variable naming. Don't
	  leak buffers on short reads.
	  (gst_v4l2src_get_mmap): Update for object variable naming, and add
	  comments.
	  (gst_v4l2src_create): Update for object variable naming.

2007-05-30 14:38:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.*: Parse subtitle text streams instead of erroring out (#442034). Still needs a parser for the su...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
	  (gst_avi_demux_reset), (gst_avi_demux_parse_stream):
	  * gst/avi/gstavidemux.h:
	  Parse subtitle text streams instead of erroring out (#442034). Still
	  needs a parser for the subtitles to actually show up.

2007-05-30 12:46:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Make _push_event() return TRUE if the event could be pushed on at least one pad and not only i...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_push_event),
	  (gst_avi_demux_loop):
	  Make _push_event() return TRUE if the event could be pushed on at
	  least one pad and not only if it could be pushed on all pads,
	  otherwise we'll end up posting an error message on EOS if one or
	  more source pads are not connected.

2007-05-28 16:39:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtsptransport.c: Use renamed RTP bin.
	  Original commit message from CVS:
	  * gst/rtsp/rtsptransport.c:
	  Use renamed RTP bin.

2007-05-28 15:01:33 +0000  Dejan Sakelšak <sakdean@gmail.com>

	  gst/videobox/gstvideobox.c: Add AYUV->AYUV and AYUV->I420 formats.
	  Original commit message from CVS:
	  Based on patch by: Dejan Sakelšak <sakdean at gmail dot com>
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	  (gst_video_box_set_property), (gst_video_box_transform_caps),
	  (video_box_recalc_transform), (gst_video_box_set_caps),
	  (gst_video_box_get_unit_size), (gst_video_box_apply_alpha),
	  (gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor),
	  (UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv),
	  (gst_video_box_i420_i420), (gst_video_box_transform),
	  (plugin_init):
	  Add AYUV->AYUV and AYUV->I420 formats.
	  Fix negotiation and I420->AYUV conversion.
	  Fixes #429329.

2007-05-26 15:25:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/speex/gstspeexdec.c: Use different variables for nested for loops so that the outer loop functions properly and s...
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
	  Use different variables for nested for loops so that the outer loop
	  functions properly and speex files with multiple frames per buffer work
	  properly.
	  Fixes #441408.

2007-05-25 20:51:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/gstid3demux.c: Don't leak newsegment events.
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_sink_event):
	  Don't leak newsegment events.

2007-05-25 20:33:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/Makefile.am: Add '-lm' to LIBS for ceil(), don't assume one of our dependencies drags it in.
	  Original commit message from CVS:
	  * gst/wavparse/Makefile.am:
	  Add '-lm' to LIBS for ceil(), don't assume one of our dependencies
	  drags it in.

2007-05-25 16:02:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacenc.*: Collect headers, add "streamheader" field to output caps and set
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_init),
	  (notgst_value_array_append_buffer),
	  (gst_flac_enc_process_stream_headers),
	  (gst_flac_enc_write_callback), (gst_flac_enc_chain),
	  (gst_flac_enc_change_state):
	  * ext/flac/gstflacenc.h:
	  Collect headers, add "streamheader" field to output caps and set
	  BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux
	  produces output according to the official FLAC-to-Ogg mapping
	  instead of completely broken files. Fixes #426044.

2007-05-25 10:44:12 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/: Handle and adjust new-segment events so that downstream really sees a stream with the tag pieces stripped off t...
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_reset),
	  (gst_id3demux_send_new_segment), (gst_id3demux_chain),
	  (gst_id3demux_sink_event):
	  * gst/id3demux/gstid3demux.h:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_reset),
	  (gst_tag_demux_chain), (gst_tag_demux_sink_event),
	  (gst_tag_demux_send_new_segment):
	  Handle and adjust new-segment events so that downstream really
	  sees a stream with the tag pieces stripped off the front and back.
	  Fixes strangeness in seeking when mp3 decoders use the new-segment
	  byte position to estimate their current playback position timestamp
	  and then the arriving buffers don't match up.

2007-05-25 10:23:49 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/autodetect/gstautoaudiosink.c: Don't unnecessarily perform a READY->NULL->READY transition on the detected audio ...
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
	  Don't unnecessarily perform a READY->NULL->READY transition on the
	  detected audio sink when starting up. Fixes: #440127

2007-05-24 17:00:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacenc.c: Don't crash in chain function if setcaps hasn't been called.
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps),
	  (gst_flac_enc_chain):
	  Don't crash in chain function if setcaps hasn't been called.

2007-05-24 08:35:23 +0000  Vincent Torri <vtorri@univ-evry.fr>

	  sys/directdraw/gstdirectdrawsink.*: Fix more warnings when compiling with MingW (#439914).
	  Original commit message from CVS:
	  Patch by: Vincent Torri  <vtorri at univ-evry fr>
	  * sys/directdraw/gstdirectdrawsink.c:
	  (gst_directdraw_sink_buffer_alloc),
	  (gst_directdraw_sink_show_frame),
	  (gst_directdraw_sink_check_primary_surface),
	  (gst_directdraw_sink_check_offscreen_surface),
	  (EnumModesCallback2), (gst_directdraw_sink_get_ddrawcaps),
	  (gst_directdraw_sink_surface_create):
	  * sys/directdraw/gstdirectdrawsink.h:
	  Fix more warnings when compiling with MingW (#439914).

2007-05-24 08:14:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Init value to avoid infinte loops.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
	  Init value to avoid infinte loops.

2007-05-24 08:10:42 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtsp/: Fix for new API.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
	  (gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
	  (gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	  (gst_rtspsrc_play):
	  (rtsp_connection_send), (rtsp_connection_receive):
	  * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
	  Fix for new API.
	  * gst/rtsp/rtspconnection.c: (add_auth_header),
	  Only add authorisation and session headers when sending messages.
	  * gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
	  (rtsp_message_init_request), (rtsp_message_init_response),
	  (rtsp_message_unset), (rtsp_message_add_header),
	  (rtsp_message_remove_header), (rtsp_message_get_header),
	  (rtsp_message_append_headers), (dump_key_value),
	  (rtsp_message_dump):
	  * gst/rtsp/rtspmessage.h:
	  Add support for multiple headers of the same type by storing the parsed
	  headers in a GArray instaed of a hashtable.

2007-05-23 22:44:12 +0000  Sébastien Moutte <sebastien@moutte.net>

	  docs/plugins/gst-plugins-bad-plugins.args: Remove directsoundsink property doc as this sink use the mixer interface now.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  Remove directsoundsink property doc as this sink use the mixer
	  interface now.
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  Add interfaces implemented by Windows sinks.
	  * sys/directsound/gstdirectsoundsink.c:
	  * sys/directsound/gstdirectsoundsink.h:
	  Remove directsoundsink property  and implement the mixer interface.
	  * win32/vs6/gst_plugins_bad.dsw:
	  * win32/vs6/libgstdirectsound.dsp:
	  Update project files.
	  * gst-libs/gst/dshow/gstdshow.cpp:
	  * gst-libs/gst/dshow/gstdshow.h:
	  * gst-libs/gst/dshow/gstdshowfakesink.cpp:
	  * gst-libs/gst/dshow/gstdshowfakesink.h:
	  * gst-libs/gst/dshow/gstdshowfakesrc.cpp:
	  * gst-libs/gst/dshow/gstdshowfakesrc.h:
	  * gst-libs/gst/dshow/gstdshowinterface.cpp:
	  * gst-libs/gst/dshow/gstdshowinterface.h:
	  * win32/common/libgstdshow.def:
	  * win32/vs6/libgstdshow.dsp:
	  Add a new gst library which allow to create internal Direct Show
	  graph (pipelines) to wrap Windows sources, decoders or encoders.
	  It includes a DirectShow fake source and sink and utility functions.
	  * sys/dshowsrcwrapper/gstdshowaudiosrc.c:
	  * sys/dshowsrcwrapper/gstdshowaudiosrc.h:
	  * sys/dshowsrcwrapper/gstdshowsrcwrapper.c:
	  * sys/dshowsrcwrapper/gstdshowsrcwrapper.h:
	  * sys/dshowsrcwrapper/gstdshowvideosrc.c:
	  * sys/dshowsrcwrapper/gstdshowvideosrc.h:
	  * win32/vs6/libdshowsrcwrapper.dsp:
	  Add a new plugin to wrap DirectShow sources on Windows.
	  It gets data from any webcam, dv cam, micro. We could add
	  tv tunner card later.

2007-05-22 11:14:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Depend on gstreamer-0.10.12.1. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _Gs...
	  Original commit message from CVS:
	  * configure.ac:
	  Depend on gstreamer-0.10.12.1.
	  * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN,
	  _GstIirEqualizerBand, object, _GstIirEqualizerBandClass,
	  parent_class, gst_iir_equalizer_band_set_property,
	  gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type,
	  gst_iir_equalizer_child_proxy_get_child_by_index,
	  gst_iir_equalizer_child_proxy_get_children_count,
	  gst_iir_equalizer_child_proxy_interface_init, setup_filter,
	  gst_iir_equalizer_compute_frequencies,
	  gst_iir_equalizer_set_property, gst_iir_equalizer_get_property,
	  plugin_init):
	  * gst/equalizer/gstiirequalizer.h (audiofilter):
	  * gst/equalizer/gstiirequalizernbands.c (ARG_NUM_BANDS,
	  gst_iir_equalizer_nbands_base_init, gst_iir_equalizer_nbands_init,
	  gst_iir_equalizer_nbands_set_property):
	  Use new locking macros.
	  * gst/filter/gstbpwsinc.c (bpwsinc_set_caps):
	  Add fixme.
	  * gst/spectrum/gstspectrum.c (SPECTRUM_WINDOW_BASE,
	  SPECTRUM_WINDOW_LEN, gst_spectrum_init, gst_spectrum_set_property,
	  gst_spectrum_event, gst_spectrum_transform_ip):
	  Use new locking macros. Turn two fixed values into #defines.

2007-05-22 11:03:30 +0000  Edward Hervey <bilboed@bilboed.com>

	  docs/plugins/Makefile.am: Also look for .m (objectivec) files.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  Also look for .m (objectivec) files.
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * sys/osxvideo/osxvideosink.m:
	  Add documentation for element and properties.

2007-05-21 14:01:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBa...
	  Original commit message from CVS:
	  * ChangeLog:
	  ChangeLog surgery.
	  * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN,
	  _GstIirEqualizerBand, object, _GstIirEqualizerBandClass,
	  parent_class, gst_iir_equalizer_band_set_property,
	  gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type,
	  gst_iir_equalizer_child_proxy_get_child_by_index,
	  gst_iir_equalizer_child_proxy_get_children_count,
	  gst_iir_equalizer_child_proxy_interface_init, setup_filter,
	  gst_iir_equalizer_compute_frequencies, plugin_init):
	  * tests/icles/equalizer-test.c:
	  Add fixme and comment for example.

2007-05-21 12:43:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	* gst/spectrum/gstspectrum.c:
	  gst/spectrum/gstspectrum.c (gst_spectrum_set_property, gst_spectrum_event, gst_spectrum_transform_ip):
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c (gst_spectrum_set_property,
	  gst_spectrum_event, gst_spectrum_transform_ip):
	  Use lock to protect from concurrent access.

2007-05-21 11:37:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.c: Specify and use properties as unsigned int that are an unsigned int.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
	  (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
	  Specify and use properties as unsigned int that are an unsigned int.

2007-05-21 11:17:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.*: Fixup docs, make the bitrate property an int as it should be and allow to set the differ...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
	  (gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config),
	  (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
	  * ext/wavpack/gstwavpackenc.h:
	  Fixup docs, make the bitrate property an int as it should be and
	  allow to set the different extra processing modes instead of only
	  allowing none and the default one.

2007-05-21 10:07:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Since we depend on 0.10.13 -core, override the unlock_stop vmethod for safer shutdown.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	  (gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop):
	  Since we depend on 0.10.13 -core, override the unlock_stop vmethod for
	  safer shutdown.

2007-05-21 10:03:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtpdec.*: Added signal for backwards compat.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init):
	  * gst/rtsp/gstrtpdec.h:
	  Added signal for backwards compat.

2007-05-21 09:32:26 +0000  René Stadler <mail@renestadler.de>

	  Use audioconvert for converting from non-native endianness floats in auparse instead of doing it ourself. Fixes #424527.
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler dot de>
	  * configure.ac:
	  * gst/auparse/gstauparse.c: (gst_au_parse_reset),
	  (gst_au_parse_parse_header), (gst_au_parse_chain):
	  * gst/auparse/gstauparse.h:
	  Use audioconvert for converting from non-native endianness floats
	  in auparse instead of doing it ourself. Fixes #424527.
	  This needs the audioconvert from plugins-base CVS.

2007-05-21 09:29:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph263ppay.c: Fix enum registration.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	  (gst_rtp_h263p_pay_flush):
	  Fix enum registration.

2007-05-21 08:57:18 +0000  Antoine Tremblay <hexa00@gmail.com>

	  gst/rtp/gstrtph263ppay.*: Add new fragmentation mode base on GOB headers. Fixes #438940.
	  Original commit message from CVS:
	  Patch by: Antoine Tremblay <hexa00 at gmail dot com>
	  * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	  (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
	  (gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
	  (gst_rtp_h263p_pay_flush):
	  * gst/rtp/gstrtph263ppay.h:
	  Add new fragmentation mode base on GOB headers. Fixes #438940.

2007-05-20 21:31:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.c: Add missing audioconverts in the example pipelines of wavpackenc. As the wavpack stuff n...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c:
	  Add missing audioconverts in the example pipelines of wavpackenc. As
	  the wavpack stuff now needs input with 32 bit width (and random depth)
	  this is needed now. The example pipelines for the parser and decoder
	  are still fine.

2007-05-20 14:59:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/directdraw/gstdirectdrawsink.c: Bunch of small fixes: remove static function that doesn't exist; declare another ...
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c: (gst_ddrawsurface_finalize),
	  (gst_directdraw_sink_buffer_alloc),
	  (gst_directdraw_sink_get_ddrawcaps),
	  (gst_directdraw_sink_surface_create):
	  Bunch of small fixes: remove static function that doesn't exist;
	  declare another one that does; printf format fix; use right macro
	  when specifying debug category; remove a bunch of unused variables;
	  #if 0 out an unused chunk of code (partially fixes #439914).

2007-05-20 14:14:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Printf format fixes (#439910, #439911).
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample):
	  * gst/switch/gstswitch.c: (gst_switch_chain):
	  Printf format fixes (#439910, #439911).

2007-05-20 14:05:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtsp/gstrtspsrc.c: Printf format fix.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
	  Printf format fix.

2007-05-19 10:01:45 +0000  René Stadler <mail@renestadler.de>

	  Add replaygain playback elements (#412710).
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler de>
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * gst/replaygain/Makefile.am:
	  * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init),
	  (gst_rg_analysis_start), (gst_rg_analysis_set_caps),
	  (gst_rg_analysis_transform_ip), (gst_rg_analysis_event),
	  (gst_rg_analysis_stop), (gst_rg_analysis_handle_tags),
	  (gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result),
	  (gst_rg_analysis_album_result):
	  * gst/replaygain/gstrganalysis.h:
	  * gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init),
	  (gst_rg_limiter_class_init), (gst_rg_limiter_init),
	  (gst_rg_limiter_set_property), (gst_rg_limiter_get_property),
	  (gst_rg_limiter_transform_ip):
	  * gst/replaygain/gstrglimiter.h:
	  * gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init),
	  (gst_rg_volume_class_init), (gst_rg_volume_init),
	  (gst_rg_volume_set_property), (gst_rg_volume_get_property),
	  (gst_rg_volume_dispose), (gst_rg_volume_change_state),
	  (gst_rg_volume_sink_event), (gst_rg_volume_tag_event),
	  (gst_rg_volume_reset), (gst_rg_volume_update_gain),
	  (gst_rg_volume_determine_gain):
	  * gst/replaygain/gstrgvolume.h:
	  * gst/replaygain/replaygain.c: (plugin_init):
	  * gst/replaygain/replaygain.h:
	  * gst/replaygain/rganalysis.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/rganalysis.c: (send_eos_event),
	  (GST_START_TEST):
	  * tests/check/elements/rglimiter.c: (setup_rglimiter),
	  (cleanup_rglimiter), (set_playing_state), (create_test_buffer),
	  (verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main):
	  * tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume),
	  (cleanup_rgvolume), (set_playing_state), (set_null_state),
	  (send_eos_event), (send_tag_event), (test_buffer_new),
	  (fail_unless_target_gain), (fail_unless_result_gain),
	  (fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main):
	  Add replaygain playback elements (#412710).

2007-05-18 13:27:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Don't crash when an unsupported transport error was returned by the server, just try to config...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	  Don't crash when an unsupported transport error was returned by the
	  server, just try to configure the next stream. Fixes #439255.

2007-05-18 11:39:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Add TCP timeout property and use it for all TCP connection.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	  (gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	  (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	  (gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
	  * gst/rtsp/gstrtspsrc.h:
	  Add TCP timeout property and use it for all TCP connection.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	  (rtsp_connection_write), (rtsp_connection_next_timeout),
	  (rtsp_connection_reset_timeout):
	  Make connect and writes cancelable and make them use the timeout.

2007-05-18 10:36:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Refactor timeout handling.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
	  (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	  (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	  (gst_rtspsrc_setup_streams):
	  Refactor timeout handling.
	  Also send keep-alive when dealing with TCP transport.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	  (rtsp_connection_free), (rtsp_connection_next_timeout),
	  (rtsp_connection_reset_timeout):
	  * gst/rtsp/rtspconnection.h:
	  Use a timer to handle the session timeouts, add some methods to deal
	  with timeouts.

2007-05-17 14:56:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Ignore streams that fail the setup command, we will retry with a different transport later on.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
	  (gst_rtspsrc_setup_streams):
	  Ignore streams that fail the setup command, we will retry with a
	  different transport later on.
	  * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
	  (rtsp_ext_wms_configure_stream):
	  Fix encoding name case.

2007-05-17 10:59:00 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/osxvideo/osxvideosink.*: Remove the event-loop-in-separate-thread modifications, because MacOSX is $#@(*%$# ! For...
	  Original commit message from CVS:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  Remove the event-loop-in-separate-thread modifications, because MacOSX
	  is $#@(*%$# ! For those wondering, the event handling needs to be done
	  in the main thread after all..

2007-05-17 09:41:48 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/osxvideo/osxvideosink.*: Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now.
	  Original commit message from CVS:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now.
	  Use a separate thread/task for the cocoa event_loop, else it wouldn't
	  stop.

2007-05-16 16:50:23 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/libpng/gstpngdec.c: Fix build on macosx.
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c: (user_endrow_callback), (user_read_data):
	  Fix build on macosx.

2007-05-16 16:30:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/raw1394/gstdv1394src.c: Replace direct comparison of a string with the string literal "" with a comparison of the...
	  Original commit message from CVS:
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri):
	  Replace direct comparison of a string with the string literal "" with
	  a comparison of the first character with '\0'. Fixes #438926.

2007-05-15 17:22:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save and restore the various flags in the directdraw/dir...
	  Original commit message from CVS:
	  * configure.ac:
	  * sys/directdraw/Makefile.am:
	  * sys/directsound/Makefile.am:
	  Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save
	  and restore the various flags in the directdraw/directsound
	  detection section. Apparently improves cross-compiling for win32
	  with mingw32 under some circumstances (#437539).

2007-05-15 11:18:33 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/debug/breakmydata.c (gst_break_my_data_init): One more try. This should be the proper fix now.
	  Original commit message from CVS:
	  * gst/debug/breakmydata.c (gst_break_my_data_init):
	  One more try. This should be the proper fix now.

2007-05-15 06:41:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/debug/breakmydata.c: Ooops, no // comments please.
	  Original commit message from CVS:
	  * gst/debug/breakmydata.c:
	  Ooops, no // comments please.

2007-05-15 06:34:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/debug/breakmydata.c: Fix gst_buffer_is_writable() assertion.
	  Original commit message from CVS:
	  * gst/debug/breakmydata.c: (gst_break_my_data_class_init),
	  (gst_break_my_data_init):
	  Fix gst_buffer_is_writable() assertion.

2007-05-15 02:56:23 +0000  David Schleef <ds@schleef.org>

	  sys/v4l2/gstv4l2src.c: Add support for Bayer images as video/x-raw-bayer.  Fixes #314160.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: Add support for Bayer images as
	  video/x-raw-bayer.  Fixes #314160.

2007-05-14 17:10:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Update theora pay/depayloader in a similar to vorbis.
	  Original commit message from CVS:
	  * gst/rtp/gstrtptheoradepay.c: (decode_base64),
	  (gst_rtp_theora_depay_parse_configuration):
	  * gst/rtp/gstrtptheorapay.c: (encode_base64),
	  (gst_rtp_theora_pay_finish_headers),
	  (gst_rtp_theora_pay_handle_buffer):
	  Update theora pay/depayloader in a similar to vorbis.
	  * gst/rtp/gstrtpvorbisdepay.c:
	  (gst_rtp_vorbis_depay_parse_configuration):
	  Update docs.

2007-05-14 16:19:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: When we try to execute a method that is not supported by the server, don't error out but remov...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
	  When we try to execute a method that is not supported by the server,
	  don't error out but remove the method from the accepted methods so that
	  we never try to perform this method again.

2007-05-14 14:47:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpvorbisdepay.c: Remove annoying _dump_mem.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
	  Remove annoying _dump_mem.

2007-05-14 11:11:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Parse range correctly.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
	  Parse range correctly.
	  * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
	  The baseurl now always has a '/' at the start.

2007-05-14 09:01:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Factor out caps configuration and configure more stuff such as the time ranges and speed/scale...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
	  (gst_rtspsrc_parse_range), (gst_rtspsrc_open),
	  (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	  Factor out caps configuration and configure more stuff such as the time
	  ranges and speed/scale values.
	  * gst/rtsp/rtsptransport.c:
	  Add Copyright after non-trival fixes.

2007-05-13 19:57:45 +0000  David Schleef <ds@schleef.org>

	  gst/replaygain/rganalysis.c: Fix wrong ifdef for visual C++.  Fixes: #437403.
	  Original commit message from CVS:
	  * gst/replaygain/rganalysis.c:
	  Fix wrong ifdef for visual C++.  Fixes: #437403.
	  By Ali Sabil <ali.sabil@gmail.com>.

2007-05-13 15:47:13 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/level/gstlevel.c: Use guint8 * instead of gpointer then vs6 can build in_data += (filter->width / 8).
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_transform_ip):
	  Use guint8 * instead of gpointer then vs6 can build
	  in_data += (filter->width / 8).

2007-05-12 16:37:50 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtsp/: Make channel guint8 where possible.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/rtsp/gstrtspsrc.h:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
	  * gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
	  (rtsp_message_get_header):
	  * gst/rtsp/rtspmessage.h:
	  Make channel guint8 where possible.
	  Make rtsp_message_init_data() take the channel as a guint8.
	  * gst/rtsp/rtspdefs.c:
	  Fixed a typo: Timout -> Timeout
	  * gst/rtsp/rtspdefs.h:
	  Make RTSP_CHECK() behave as a statement.
	  * gst/rtsp/sdpmessage.c:
	  Avoid a compiler warning in INIT_ARRAY().
	  Fixes #437692.

2007-05-12 16:27:51 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtsp/rtspurl.*: Add support for query parameters to RTSP URLs.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
	  (rtsp_url_get_request_uri):
	  * gst/rtsp/rtspurl.h:
	  Add support for query parameters to RTSP URLs.

2007-05-12 16:26:06 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
	  (parse_range), (range_as_text), (rtsp_transport_mode_as_text),
	  (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
	  (rtsp_transport_parse), (rtsp_transport_as_text):
	  * gst/rtsp/rtsptransport.h:
	  Add validation to rtsp_transport_parse().
	  Add rtsp_transport_as_text() to generate an RTSP header from an
	  RTSPTransport.
	  Change ssrc to guint (was a string) since that is what it is, even
	  though it is sent as a hex string.
	  Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
	  incorrect, which can be seen when looking at the examples in the RFC).
	  Fixes #437670.

2007-05-11 16:11:04 +0000  Eric Anholt <anholt@freebsd.org>

	* ChangeLog:
	* sys/ximage/gstximagesrc.c:
	  sys/ximage/gstximagesrc.c (gst_ximage_src_open_display, gst_ximage_src_ximage_get):
	  Original commit message from CVS:
	  Patch by: Eric Anholt
	  * sys/ximage/gstximagesrc.c (gst_ximage_src_open_display,
	  gst_ximage_src_ximage_get):
	  Use union of all damage between frames to make it faster.
	  Fixes bug #342463.
	  Also fix crasher when cursor is at bottom right of window.

2007-05-11 16:01:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: Skip LIST chunks before the fmt chunk (fixes #437499). Also fix streaming mode regression...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Skip LIST chunks before the fmt chunk (fixes #437499). Also fix
	  streaming mode regression for file from #343837 with 'bext' chunk
	  before the 'fmt' chunk.

2007-05-11 15:09:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Preliminary seek support.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	  (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
	  (gst_rtspsrc_handle_src_event),
	  (gst_rtspsrc_stream_configure_manager),
	  (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
	  (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	  * gst/rtsp/gstrtspsrc.h:
	  * gst/rtsp/rtspdefs.h:
	  Preliminary seek support.
	  Activate internal pads so that we can receive events on them.
	  Don't try to parse a range string when it's NULL.

2007-05-11 15:04:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/README: Update README with new RTP variables that will be used for synchronisation.
	  Original commit message from CVS:
	  * gst/rtp/README:
	  Update README with new RTP variables that will be used for
	  synchronisation.
	  * gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
	  (gst_rtp_vorbis_depay_parse_configuration),
	  (gst_rtp_vorbis_depay_process):
	  * gst/rtp/gstrtpvorbispay.c: (encode_base64),
	  (gst_rtp_vorbis_pay_finish_headers),
	  (gst_rtp_vorbis_pay_handle_buffer):
	  Update vorbis pay and depayloader to draft-04.

2007-05-11 11:24:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtsptransport.c: UDP MCAST is actually the default for RTP/AVP.
	  Original commit message from CVS:
	  * gst/rtsp/rtsptransport.c:
	  UDP MCAST is actually the default for RTP/AVP.

2007-05-11 10:31:27 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximage/gstximagesrc.c (gst_ximage_src_start, gst_ximage_src_ximage_get):
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c (gst_ximage_src_start,
	  gst_ximage_src_ximage_get):
	  * sys/ximage/gstximagesrc.h (last_ximage):
	  When using Damage actually keep the last frame, and not assume
	  that the buffer we get already has the last frame on it.
	  Copy the cursor over if we specify a non-zero start x and
	  start y.

2007-05-11 09:12:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtsptransport.c: Make UDP the default transport when not specified.
	  Original commit message from CVS:
	  * gst/rtsp/rtsptransport.c:
	  Make UDP the default transport when not specified.

2007-05-10 14:02:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
	  gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
	  gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
	  gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
	  qtdemux_parse_segments, qtdemux_parse_trak):
	  * gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
	  rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
	  rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
	  rtp_session_get_location, rtp_session_get_tool,
	  rtp_session_process_bye, session_report_blocks):
	  * gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
	  rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
	  More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
	  * gst/switch/Makefile.am:
	  Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).

2007-05-10 01:21:19 +0000  David Schleef <ds@schleef.org>

	  gst/level/gstlevel.c: Revert last change.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c:
	  Revert last change.

2007-05-09 21:30:53 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/level/gstlevel.c: Use guint8 * instead of gpointer then vs6 know the size of data pointed when moving the pointer.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_calculate_##TYPE),
	  (gst_level_transform_ip):
	  Use guint8 * instead of gpointer then vs6 know the size of data
	  pointed when moving the pointer.
	  * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
	  Move instructions after variables declaration.
	  * win32/vs6/autogen.dsp:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgstrtsp.dsp:
	  Update vs6 project files.

2007-05-09 11:23:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Add code to parse time ranges.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query),
	  (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open):
	  * gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range),
	  (parse_clock_range), (parse_smpte_range), (rtsp_range_parse),
	  (rtsp_range_free):
	  * gst/rtsp/rtsprange.h:
	  Add code to parse time ranges.
	  Report DURATION on the stream when possible.

2007-05-08 15:49:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videomixer/videomixer.c: Fix strides calculation for AYUV (it's just width*4) (#436910).
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
	  (gst_videomixer_fill_checker), (gst_videomixer_fill_color),
	  (gst_videomixer_collected):
	  Fix strides calculation for AYUV (it's just width*4) (#436910).

2007-05-06 21:32:40 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Sync the GObject properties before each processing step to properly work with the controller.
	  Original commit message from CVS:
	  * gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
	  * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
	  * gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
	  Sync the GObject properties before each processing step to properly
	  work with the controller.

2007-05-04 15:17:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Let more error state trickle down so that we can catch more error cases.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	  (gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	  (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	  (gst_rtspsrc_change_state):
	  Let more error state trickle down so that we can catch more error
	  cases.
	  Handle keep-alive a little smarter by selecting a method the server
	  actually supports.
	  Fix a race in UDP streaming shutdown.

2007-05-04 13:04:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Ignore errors when trying to use the keep-alive messages.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
	  Ignore errors when trying to use the keep-alive messages.

2007-05-04 12:31:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP connection.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
	  (gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
	  (gst_rtspsrc_stream_configure_manager),
	  (gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
	  (gst_rtspsrc_stream_configure_mcast),
	  (gst_rtspsrc_stream_configure_udp),
	  (gst_rtspsrc_stream_configure_udp_sink),
	  (gst_rtspsrc_stream_configure_transport):
	  Send RTCP messages back to the server over the TCP connection.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_write),
	  (rtsp_connection_send), (rtsp_connection_read), (read_body),
	  (rtsp_connection_receive):
	  * gst/rtsp/rtspconnection.h:
	  Factor out and expose lowlevel _write and _read methods.
	  Implement sending data messages to the server.

2007-05-03 15:55:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/multipart/multipartmux.c: Fix timestamps on outgoing buffers.
	  Original commit message from CVS:
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads),
	  (gst_multipart_mux_collected):
	  Fix timestamps on outgoing buffers.

2007-05-03 14:39:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/multipart/multipartmux.c: Emit NEWSEGMENT events before pushing the first buffer.
	  Original commit message from CVS:
	  * gst/multipart/multipartmux.c:
	  (gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected),
	  (gst_multipart_mux_change_state):
	  Emit NEWSEGMENT events before pushing the first buffer.

2007-05-03 13:48:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Refactor transport configuration code.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	  (gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
	  (gst_rtspsrc_handle_src_query),
	  (gst_rtspsrc_stream_configure_manager),
	  (gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
	  (gst_rtspsrc_stream_configure_mcast),
	  (gst_rtspsrc_stream_configure_udp),
	  (gst_rtspsrc_stream_configure_udp_sink),
	  (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
	  (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	  (gst_rtspsrc_pause):
	  Refactor transport configuration code.
	  Create internal pads for TCP transport so that we can implement events
	  and queries.
	  Handle events and queries.
	  Parse range from the SDP.
	  Fix race in pause handler where the connection could still be flushing.

2007-05-02 19:32:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
	  (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
	  (gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
	  (gst_rtspsrc_play), (gst_rtspsrc_handle_message),
	  (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Fix race when multiple udp sources post timeouts, just act on the first
	  received timeout.
	  Protect stream list with a recursive lock to fix some races.
	  Flush connection when we need to do a reconnect or stop.
	  Make state lock recursive.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	  (rtsp_connection_close):
	  Some small cleanups.

2007-05-02 18:31:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpack.c: Call bindtextdomain() to get localized strings.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpack.c: (plugin_init):
	  Call bindtextdomain() to get localized strings.
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
	  (gst_wavpack_parse_handle_seek_event),
	  (gst_wavpack_parse_push_buffer), (gst_wavpack_parse_chain):
	  * ext/wavpack/gstwavpackparse.h:
	  Handle DISCONT buffers by correctly setting the DISCONT flag
	  on outgoing buffers when necessary.
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event)
	  Send newsegment from the streaming thread.

2007-05-02 18:25:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Only set DISCONT when there actually is a discont or when we just started.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	  Only set DISCONT when there actually is a discont or when we just
	  started.

2007-05-02 18:01:52 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflac.c: Call bindtextdomain() to get localized strings.
	  Original commit message from CVS:
	  * ext/flac/gstflac.c: (plugin_init):
	  Call bindtextdomain() to get localized strings.

2007-05-02 17:19:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.*: Be a bit more clever when dealing with VBR files with FACT tags, we don't want to timesta...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
	  (gst_wavparse_stream_data):
	  * gst/wavparse/gstwavparse.h:
	  Be a bit more clever when dealing with VBR files with FACT tags, we
	  don't want to timestamp buffers in that case but the estimated BPS can
	  be used for seeking.
	  Only send close segment in the streaming thread.

2007-05-02 17:08:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/flac/gstflacdec.c: Correctly post an error on the bus if something went wrong in the loop function. This fixes a ...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_loop):
	  Correctly post an error on the bus if something went wrong in the loop
	  function. This fixes a few cases where the task was paused and nothing
	  happened anymore.

2007-05-02 16:58:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Remove old workaround that was needed when seeking after the last sample. With the fix...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_handle_seek_event):
	  Remove old workaround that was needed when seeking after the last
	  sample. With the fixed error handling this works now as expected
	  without pushing the last sample although it wasn't requested.

2007-05-02 16:45:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Handle segment seeks in the seek event handler, correctly work with stop position == -...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_handle_seek_event):
	  Handle segment seeks in the seek event handler, correctly work with
	  stop position == -1 and instead of stopping the task on seek just
	  pause it.

2007-05-02 16:19:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Add handling for segment seeks.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_loop):
	  Add handling for segment seeks.

2007-05-02 15:13:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Correctly handle errors, especially in the loop function. Before it was easy to get th...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_pull_buffer),
	  (gst_wavpack_parse_create_src_pad),
	  (gst_wavpack_parse_resync_loop), (gst_wavpack_parse_loop),
	  (gst_wavpack_parse_chain):
	  Correctly handle errors, especially in the loop function. Before it
	  was easy to get the task paused but no error being posted on the bus.

2007-05-02 14:27:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/test.c: Fix compilation of deprecated test just because I'm too lazy to delete it.
	  Original commit message from CVS:
	  * gst/rtsp/test.c: (main):
	  Fix compilation of deprecated test just because I'm too lazy to delete
	  it.

2007-05-02 13:32:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
	  (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
	  (gst_rtspsrc_open), (gst_rtspsrc_handle_message):
	  * gst/rtsp/gstrtspsrc.h:
	  Fix sending RTCP to the right place.
	  Fix bug in reffing the wrong UDP element.
	  Use new pad names for the session manager.
	  Implement handling server requests in interleaved and UDP modes.
	  Handle session keep-alive in UDP modes.
	  Remove GCond for handling UDP timeouts.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	  (rtsp_connection_send), (rtsp_connection_read), (read_body),
	  (rtsp_connection_receive), (rtsp_connection_close):
	  * gst/rtsp/rtspconnection.h:
	  Store connection IP address for later.
	  Add timeout args to all operations that might block forever.
	  Parse session timeout.
	  Only close sockets when not already closed.
	  * gst/rtsp/rtspdefs.c:
	  * gst/rtsp/rtspdefs.h:
	  Add timeout return value and error string.
	  * gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
	  Add small comment.

2007-05-01 16:13:58 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/rtp/gstrtpmp4vpay.*: Handle NEWSEGMENT and FLUSH events. Fixes #434824.
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
	  (gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event):
	  * gst/rtp/gstrtpmp4vpay.h:
	  Handle NEWSEGMENT and FLUSH events. Fixes #434824.

2007-04-30 11:15:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/gst-plugins-good-plugins-docs.sgml: Remove v4l2src from docs, since it breaks the docs build, and the pl...
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  Remove v4l2src from docs, since it breaks the docs build, and the
	  plugin is only built if --enable-experimental is used anyway.
	  * docs/plugins/Makefile.am:
	  Spaces => tab.

2007-04-29 14:43:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.c: Add code to drop membership of a multicast group.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (leave_multicast),
	  (gst_multiudpsink_add), (gst_multiudpsink_remove):
	  Add code to drop membership of a multicast group.
	  * gst/udp/gstudpsink.c: (gst_udpsink_update_uri),
	  (gst_udpsink_set_uri):
	  Implement URI handler.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_parse_rtpinfo):
	  Use URI handler to make udpsink instace.
	  Improve code to configure port and destination.

2007-04-29 13:56:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* sys/directdraw/gstdirectdrawsink.c:
	* sys/osxvideo/osxvideosink.m:
	  80 char police
	  Original commit message from CVS:
	  80 char police

2007-04-29 13:53:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  autogen.sh: Require automake 1.7
	  Original commit message from CVS:
	  * autogen.sh:
	  Require automake 1.7
	  * ext/alsaspdif/Makefile.am:
	  * ext/divx/Makefile.am:
	  * ext/ivorbis/Makefile.am:
	  * ext/musicbrainz/Makefile.am:
	  * ext/neon/Makefile.am:
	  * ext/sdl/Makefile.am:
	  * ext/swfdec/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/wavpack/Makefile.am:
	  * ext/xvid/Makefile.am:
	  * gst/modplug/Makefile.am:
	  Fix up Makefile.am accordingly.

2007-04-29 13:49:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  docs/plugins/inspect/: Add jack and update.
	  Original commit message from CVS:
	  * docs/plugins/inspect/plugin-alsaspdif.xml:
	  * docs/plugins/inspect/plugin-bz2.xml:
	  * docs/plugins/inspect/plugin-cdxaparse.xml:
	  * docs/plugins/inspect/plugin-dfbvideosink.xml:
	  * docs/plugins/inspect/plugin-faac.xml:
	  * docs/plugins/inspect/plugin-faad.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-freeze.xml:
	  * docs/plugins/inspect/plugin-glimagesink.xml:
	  * docs/plugins/inspect/plugin-gsm.xml:
	  * docs/plugins/inspect/plugin-h264parse.xml:
	  * docs/plugins/inspect/plugin-jack.xml:
	  * docs/plugins/inspect/plugin-mms.xml:
	  * docs/plugins/inspect/plugin-modplug.xml:
	  * docs/plugins/inspect/plugin-musepack.xml:
	  * docs/plugins/inspect/plugin-musicbrainz.xml:
	  * docs/plugins/inspect/plugin-neon.xml:
	  * docs/plugins/inspect/plugin-nsfdec.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-sdl.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-tta.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * docs/plugins/inspect/plugin-xingheader.xml:
	  * docs/plugins/inspect/plugin-xvid.xml:
	  Add jack and update.

2007-04-29 12:19:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.c: Fix multicast detection.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
	  Fix multicast detection.
	  Don't try to join a multicast group if the address is not multicast.
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri):
	  Small debug improvement.

2007-04-27 16:44:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Ignore ASYNC state messages from the udpsink, it's irrelevant for the parent.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	  (gst_rtspsrc_handle_message):
	  Ignore ASYNC state messages from the udpsink, it's irrelevant for the
	  parent.

2007-04-27 15:30:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpilbcdepay.h: Fix mode property when specified as an arg.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpilbcdepay.h:
	  Fix mode property when specified as an arg.

2007-04-26 15:08:20 +0000  Edward Hervey <bilboed@bilboed.com>

	  docs/plugins/: Add documentation for osxaudio plugin.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-osxaudio.xml:
	  Add documentation for osxaudio plugin.

2007-04-26 14:31:32 +0000  Edward Hervey <bilboed@bilboed.com>

	  docs/plugins/: Add documentation for osxvideo
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-osxvideo.xml:
	  Add documentation for osxvideo

2007-04-26 10:08:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Protect state changes with a lock.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_open), (gst_rtspsrc_close),
	  (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	  (gst_rtspsrc_pause):
	  * gst/rtsp/gstrtspsrc.h:
	  Protect state changes with a lock.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	  (parse_line):
	  * gst/rtsp/rtspconnection.h:
	  Remove some unused stuff.

2007-04-26 08:48:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Handle the case where there are exactly 0 bytes to read and the ioctl did not report an error. F...
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Handle the case where there are exactly 0 bytes to read and the ioctl
	  did not report an error. Fixes #433530.

2007-04-26 08:39:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.*: Apply DISCONT to buffers.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	  * gst/wavparse/gstwavparse.h:
	  Apply DISCONT to buffers.
	  Only apply timestamp to the first sample after a DISCONT, too many VBR
	  files cause random jitter in the timestamps. Fixes #433119.

2007-04-25 15:55:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtpdec.*: Add dummy latency property to be backwards compat with rtpbin.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
	  (gst_rtp_dec_init), (gst_rtp_dec_set_property),
	  (gst_rtp_dec_get_property):
	  * gst/rtsp/gstrtpdec.h:
	  Add dummy latency property to be backwards compat with rtpbin.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	  (gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_parse_rtpinfo):
	  * gst/rtsp/gstrtspsrc.h:
	  Add latency property and configure in the session manager.
	  Don't set invalid clock-base and seqnum-base on caps, some servers
	  sometimes don't send them.

2007-04-25 15:31:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/alpha/gstalphacolor.c: Double-check that RGB input caps are really RGBA caps (apparently the core doesn't always ...
	  Original commit message from CVS:
	  * gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
	  (gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
	  Double-check that RGB input caps are really RGBA caps (apparently
	  the core doesn't always catch it if those caps aren't a subset of
	  our template caps, also see #421543). Fixes #429319 in a way.
	  Also, don't leak the pad template in the transform_caps function.
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/alphacolor.c: (setup_alphacolor),
	  (cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
	  (create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
	  (GST_START_TEST), (alphacolor_suite):
	  Add some basic unit tests for alphacolor.

2007-04-25 15:08:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/libpng/gstpngdec.c: If we get a fatal flow return in the loop function, first post the error message and only the...
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  If we get a fatal flow return in the loop function, first post the
	  error message and only then send the EOS event downstream, otherwise
	  applications might get an eos message before the error message and
	  think everything was ok (related to #429319).

2007-04-25 10:07:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspconnection.c: Read the channel byte as an unsigned byte.
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
	  Read the channel byte as an unsigned byte.

2007-04-25 09:47:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Make sure we configure the clock_rate in the baseclass in the setcaps function. Fixes #431282.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
	  (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
	  * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
	  (gst_rtp_gsm_depay_setcaps):
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	  * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
	  (gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
	  (gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
	  (gst_ilbc_depay_get_property):
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	  * gst/rtp/gstrtpmp4adepay.c:
	  * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
	  (gst_rtp_pcma_depay_setcaps):
	  * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
	  (gst_rtp_pcmu_depay_setcaps):
	  Make sure we configure the clock_rate in the baseclass in the setcaps
	  function. Fixes #431282.

2007-04-25 08:36:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Parse server address from SDP.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	  (gst_rtspsrc_stream_free), (request_pt_map),
	  (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
	  * gst/rtsp/gstrtspsrc.h:
	  Parse server address from SDP.
	  Hook up a udpsink to send RTCP back to the server.
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * gst/rtsp/rtsptransport.h:
	  Add some docs.

2007-04-25 06:52:09 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: Make header field check conditional. Fixes #433135
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Make header field check conditional. Fixes #433135

2007-04-24 09:12:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add minimal docs blurb to alphacolor; split out headers into separate header file for gtk-doc.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * gst/alpha/Makefile.am:
	  * gst/alpha/gstalphacolor.c:
	  * gst/alpha/gstalphacolor.h:
	  Add minimal docs blurb to alphacolor; split out headers into
	  separate header file for gtk-doc.

2007-04-20 17:25:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/debug/progressreport.c: Don't try to post NULL message (in case we can't query upstream position or duration).
	  Original commit message from CVS:
	  * gst/debug/progressreport.c: (gst_progress_report_report):
	  Don't try to post NULL message (in case we can't query upstream
	  position or duration).

2007-04-18 12:36:37 +0000  Michael Smith <msmith@xiph.org>

	  gst/cutter/gstcutter.*: Fix some of the most obvious bugs in cutter. Now doesn't leak everything if input is silent.
	  Original commit message from CVS:
	  * gst/cutter/gstcutter.c: (gst_cutter_init), (gst_cutter_chain),
	  (gst_cutter_get_caps):
	  * gst/cutter/gstcutter.h:
	  Fix some of the most obvious bugs in cutter. Now doesn't leak
	  everything if input is silent.

2007-04-18 09:48:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavenc/gstwavenc.*: everything else results in a invalid block align and invalid files.
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	  (gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
	  * gst/wavenc/gstwavenc.h:
	  Wav apparently only supports width==GST_ROUND_UP(depth), everything
	  else results in a invalid block align and invalid files.

2007-04-17 16:39:02 +0000  Snaik <snaik32@gmail.com>

	  gst/smpte/barboxwipes.c: Add missing break statement for BOX_HORIZONTAL case.
	  Original commit message from CVS:
	  Patch by: Snaik <snaik32 gmail com>
	  * gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw):
	  Add missing break statement for BOX_HORIZONTAL case.

2007-04-17 10:14:43 +0000  Vincent Torri <vtorri@univ-evry.fr>

	  gst/wavparse/gstwavparse.c: Use correct format strings for integer types.
	  Original commit message from CVS:
	  Patch by: Vincent Torri <vtorri at univ-evry dot fr>
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Use correct format strings for integer types.

2007-04-17 02:51:02 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavparse/gstwavparse.c: Use gst_riff_create_audio_template_caps () instead of the local caps.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	  (gst_wavparse_create_sourcepad):
	  Use gst_riff_create_audio_template_caps () instead of the local caps.
	  This makes updates of the local caps unecessary whenever libgstriff
	  gets support for new formats.

2007-04-16 21:29:40 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/: Fix and/or update copyright attributions (#430228).
	  Original commit message from CVS:
	  Patch by: Brian Cameron  <brian.cameron at sun dot com>
	  * sys/sunaudio/gstsunaudio.c:
	  * sys/sunaudio/gstsunaudiomixer.c:
	  * sys/sunaudio/gstsunaudiomixer.h:
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  * sys/sunaudio/gstsunaudiomixerctrl.h:
	  * sys/sunaudio/gstsunaudiomixertrack.h:
	  * sys/sunaudio/gstsunaudiosink.c:
	  * sys/sunaudio/gstsunaudiosink.h:
	  * sys/sunaudio/gstsunaudiosrc.c:
	  * sys/sunaudio/gstsunaudiosrc.h:
	  Fix and/or update copyright attributions (#430228).

2007-04-14 17:18:14 +0000  Sébastien Moutte <sebastien@moutte.net>

	  docs/plugins/inspect/: Add xml doc files for Windows sinks
	  Original commit message from CVS:
	  * docs/plugins/inspect/plugin-directdraw.xml:
	  * docs/plugins/inspect/plugin-directsound.xml:
	  * docs/plugins/inspect/plugin-waveform.xml:
	  Add xml doc files for Windows sinks
	  * win32/vs6/libgstqtdemux.dsp:
	  * win32/vs6/libgstmpegvideoparse.dsp:
	  * win32/vs6/gst_plugins_bad.dsw:
	  Update projects files.

2007-04-13 09:32:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Fix docs.
	  * gst/rtsp/URLS:
	  Add some more example urls.
	  * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
	  (gst_rtp_dec_chain_rtp):
	  Better debugging.
	  * gst/rtsp/gstrtspsrc.c: (request_pt_map),
	  (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_parse_rtpinfo):
	  Remove unused code.

2007-04-13 08:19:35 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: Relax the audio/mpeg caps again and add FIXME: comment.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	  (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	  (gst_wavparse_stream_data):
	  Relax the audio/mpeg caps again and add FIXME: comment.

2007-04-13 06:20:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.*: More sanity check for the header fields. Fix type for 'rate' header field.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	  (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	  (gst_wavparse_stream_data):
	  * gst/wavparse/gstwavparse.h:
	  More sanity check for the header fields. Fix type for 'rate' header
	  field.

2007-04-12 16:06:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/icydemux/gsticydemux.c: If the metadata strings we get in the stream are not UTF-8, try to interpret them accordi...
	  Original commit message from CVS:
	  * gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8),
	  (gst_icydemux_unicodify):
	  If the metadata strings we get in the stream are not UTF-8, try to
	  interpret them according to the character encodings specified in the
	  GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and
	  only fall back to locale/ISO-8859-1 if those aren't set or don't
	  work. Should fix #428901.

2007-04-12 14:20:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph264depay.c: Use the proper sync word for SPS and PPS.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264depay.c:
	  Use the proper sync word for SPS and PPS.

2007-04-12 11:41:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME, fnv1_hash_32_new, fnv1_hash_...
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
	  fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
	  * gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
	  Add a simple hashing implementation that we can use to generate
	  a 24-bit ident value based on the codebooks for vorbis and theora.
	  * gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
	  gst_rtp_theora_pay_handle_buffer):
	  * gst/rtp/gstrtpvorbisdepay.c
	  (gst_rtp_vorbis_depay_parse_configuration,
	  gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
	  * gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
	  gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
	  gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
	  Use the hashing function, ensuring that the same codebooks result
	  in the same ident and thus the same SDP description.
	  Various log fixes/changes.

2007-04-12 11:37:50 +0000  jerry tan <jerry.tan@sun.com>

	  sys/sunaudio/gstsunaudiosrc.c: it is the application's responsibility to make sure it open the device once.
	  Original commit message from CVS:
	  Patch by: jerry tan <jerry dot tan at sun dot com>
	  * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	  remove the call of  ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the
	  application's responsibility to make sure it open the device once.
	  Remove a careless error if AUDIODEV is set. Fixes #392620.

2007-04-12 10:52:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the pts_offset calculations.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c:
	  Make timescale 32 bits again so we don't screw up the pts_offset
	  calculations.

2007-04-12 08:21:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
	  (gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
	  * gst/rtsp/gstrtpdec.h:
	  Make backward compat with rtpbin by adding the request-pt-map signals.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	  (new_session_pad), (request_pt_map),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_stream_configure_caps),
	  (gst_rtspsrc_activate_streams):
	  * gst/rtsp/gstrtspsrc.h:
	  Implement request-pt-map signals instead of setting caps on the buffers
	  for the session manager.

2007-04-11 10:25:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudp.c: Register GstNetBuffer in plugin_init so that the type can be used from multiple threads without races.
	  Original commit message from CVS:
	  * gst/udp/gstudp.c: (plugin_init):
	  Register GstNetBuffer in plugin_init so that the type can be used from
	  multiple threads without races.

2007-04-11 10:19:06 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  update to spec file
	  Original commit message from CVS:
	  update to spec file

2007-04-11 09:53:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/: Handle version 1 mdhd atoms to get extended precision durations.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
	  (qtdemux_parse_samples), (qtdemux_parse_segments),
	  (qtdemux_parse_trak), (qtdemux_parse_tree):
	  * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mdhd):
	  Handle version 1 mdhd atoms to get extended precision durations.
	  Fixes #426972.

2007-04-10 17:06:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	  (gst_rtp_amr_depay_process):
	  Fix depayloader clock_rate and some cleanups.
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
	  (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	  * gst/rtp/gstrtph264depay.h:
	  Don't push codec_data in the adapter because it might get flushed when
	  we get a discont.
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	  Handle multiple AU per packet.
	  * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
	  (gst_rtp_sv3v_depay_plugin_init):
	  Disable rank, this one does not work.
	  Remove timestamping, base class does that.

2007-04-10 12:01:33 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/auparse/gstauparse.c: limit caps to the formats we announce in the template
	  Original commit message from CVS:
	  * gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
	  limit caps to the formats we announce in the template
	  * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	  (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	  (gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
	  fix some crashers/asserts when dealing with broken files

2007-04-10 10:01:14 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/: Fix some compiler warnings. Fixes #428182.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
	  * gst/rtp/gstrtpL16depay.c:
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	  * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
	  (gst_rtp_speex_depay_setcaps):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
	  Fix some compiler warnings. Fixes #428182.

2007-04-06 12:54:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
	  (free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
	  (gst_rtp_dec_init), (gst_rtp_dec_finalize),
	  (gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
	  (gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
	  (gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
	  (gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
	  (create_rtcp), (gst_rtp_dec_request_new_pad),
	  (gst_rtp_dec_release_pad):
	  * gst/rtsp/gstrtpdec.h:
	  * gst/rtsp/gstrtsp.c: (plugin_init):
	  Morph RTPDec into something compatible with RTPBin as a fallback.
	  Various other style fixes.
	  * gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
	  (find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
	  (gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
	  (new_session_pad), (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
	  (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Implement RTPBin session manager handling.
	  Don't try to add empty properties to caps.
	  Implement fallback session manager, handling.
	  Don't combine errors from RTCP streams, just ignore them.
	  * gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
	  * gst/rtsp/rtsptransport.h:
	  Implement fallback session manager.
	  Make RTPBin the default one when available.

2007-04-05 15:05:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/gstrtpxqtdepay.*: Try to recover from packet loss a little better.
	  Original commit message from CVS:
	  * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
	  (gst_rtp_xqt_depay_change_state):
	  * gst/qtdemux/gstrtpxqtdepay.h:
	  Try to recover from packet loss a little better.

2007-04-05 13:56:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4adepay.c: This element is ready to be autoplugged.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
	  (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_plugin_init):
	  This element is ready to be autoplugged.

2007-04-05 11:26:25 +0000  Julien Moutte <julien@moutte.net>

	  gst/avi/gstavidemux.c: Don't leave the offsets defined by upstream element on the compressed data buffer we are pushi...
	  Original commit message from CVS:
	  2007-04-05  Julien MOUTTE  <julien@moutte.net>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
	  Don't leave the offsets defined by upstream element on the
	  compressed data buffer we are pushing downstream. Make them
	  GST_BUFFER_OFFSET_NONE.

2007-04-04 12:39:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/: Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
	  Original commit message from CVS:
	  * gst/avi/README:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	  (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
	  (gst_avi_demux_stream_index), (gst_avi_demux_sync),
	  (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_stream_header_push),
	  (gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
	  Don't abort on out-of-memory. Use stream-nr as unsigned integer only.

2007-04-03 09:55:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/smpte/barboxwipes.c:
	  Original commit message from CVS:
	  * gst/smpte/barboxwipes.c:
	  Fix error as spotted by Snaik <snaik32 at gmail dot com>

2007-03-30 17:19:34 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavparse/gstwavparse.c: Support audio/x-raw-float in wav files. This only works with plugins-base CVS, using an o...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Support audio/x-raw-float in wav files. This only works with
	  plugins-base CVS, using an older version doesn't have any
	  disadvantages though.

2007-03-30 15:59:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Revert last change as we don't want plugins-good to depend on plugins-base CVS now.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/auparse/gstauparse.c: (gst_au_parse_reset),
	  (gst_au_parse_parse_header), (gst_au_parse_chain):
	  * gst/auparse/gstauparse.h:
	  Revert last change as we don't want plugins-good to depend on
	  plugins-base CVS now.

2007-03-30 04:50:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept th...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	  (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps),
	  (gst_wavpack_dec_clip_outgoing_buffer),
	  (gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain):
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
	  (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
	  (gst_wavpack_enc_chain):
	  * ext/wavpack/gstwavpackenc.h:
	  * ext/wavpack/gstwavpackparse.c:
	  Don't play audioconvert. As wavpack wants/outputs all samples with
	  width==32 and depth=[1,32] accept this and let audioconvert convert
	  to accepted formats instead of doing it in the element for n*8 depths.
	  This also adds support for non-n*8 depths and prevents some useless
	  memory allocations. Fixes #421598
	  Also add a workaround for bug #421542 in wavpackenc for now...
	  * tests/check/elements/wavpackdec.c: (GST_START_TEST):
	  * tests/check/elements/wavpackenc.c: (GST_START_TEST):
	  * tests/check/elements/wavpackparse.c: (GST_START_TEST):
	  Consider the change above in the unit tests and test if the correct
	  caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in
	  the wavpackparse unit test.
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
	  (gst_wavpack_dec_sink_set_caps):
	  Set caps on the src pad as soon as possible.
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackcommon.h:
	  * ext/wavpack/gstwavpackenc.h:
	  * ext/wavpack/gstwavpackparse.h:
	  Fix indention. gst-indent is now called by cicl.

2007-03-29 18:51:33 +0000  René Stadler <mail@renestadler.de>

	  configure.ac: Require gst-plugins-base CVS for audioconvert with non-native float support and width/depth fix in libg...
	  Original commit message from CVS:
	  * configure.ac:
	  Require gst-plugins-base CVS for audioconvert with non-native
	  float support and width/depth fix in libgstriff.
	  Patch by: René Stadler <mail at renestadler dot de>
	  * gst/auparse/gstauparse.c: (gst_au_parse_reset),
	  (gst_au_parse_parse_header), (gst_au_parse_chain):
	  * gst/auparse/gstauparse.h:
	  Don't swap the floats ourself if they're not in native endianness.
	  Instead let audioconvert handle this. Fixes #339838.

2007-03-29 14:40:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Flush adapter on disconts.
	  Original commit message from CVS:
	  * gst/rtp/gstasteriskh263.h:
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process),
	  (gst_rtp_h263p_depay_change_state):
	  * gst/rtp/gstrtph263pdepay.h:
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	  (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
	  (gst_rtp_h264_depay_change_state):
	  * gst/rtp/gstrtph264depay.h:
	  * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
	  (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process):
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	  Flush adapter on disconts.

2007-03-29 14:03:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Use more efficient adapter and rtpbuffer methods when possible.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process):
	  * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process):
	  * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process):
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush):
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	  (gst_rtp_mp4v_depay_process):
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush):
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process):
	  * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush):
	  * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
	  * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
	  * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
	  * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process):
	  Use more efficient adapter and rtpbuffer methods when possible.

2007-03-29 12:14:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavenc/gstwavenc.c: Correctly handle width!=depth input.
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	  (gst_wavenc_sink_setcaps):
	  Correctly handle width!=depth input.
	  * gst/wavparse/gstwavparse.c:
	  Already export in the caps that width==8 uses unsigned samples and
	  everything else uses signed samples.

2007-03-29 09:59:23 +0000  Laurent Glayal <spglegle@yahoo.fr>

	  gst/udp/: Rework the socket allocation a bit based on the sockfd argument so that it becomes usable.
	  Original commit message from CVS:
	  Patch by: Laurent Glayal <spglegle at yahoo dot fr>
	  * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
	  (gst_dynudpsink_init), (gst_dynudpsink_set_property),
	  (gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
	  (gst_dynudpsink_close):
	  * gst/udp/gstdynudpsink.h:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	  (gst_udpsrc_create), (gst_udpsrc_set_property),
	  (gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
	  * gst/udp/gstudpsrc.h:
	  Rework the socket allocation a bit based on the sockfd argument so that
	  it becomes usable.
	  Add a closefd property to instruct the udp elements to close the custom
	  file descriptors when going to READY. Fixes #423304.
	  API:GstUDPSrc::closefd property
	  API:GstDynUDPSink::closefd property

2007-03-29 08:08:49 +0000  Laurent Glayal <spglegle@yahoo.fr>

	  gst/rtp/: Added H264 payloader. Fixes #423782.
	  Original commit message from CVS:
	  Patch by: Laurent Glayal <spglegle at yahoo dot fr>
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
	  (gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
	  (gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
	  (gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
	  (gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
	  (gst_rtp_h264_pay_plugin_init):
	  * gst/rtp/gstrtph264pay.h:
	  Added H264 payloader. Fixes #423782.
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	  (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	  Small fixes.

2007-03-28 22:27:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavparse/gstwavparse.c: Actually support depths from 1 to 32, not only 8 to 32.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Actually support depths from 1 to 32, not only 8 to 32.

2007-03-28 22:23:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/wavparse/gstwavparse.c: Add support for wav files containing audio/x-raw-int with random depths between 1 and 32 ...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Add support for wav files containing audio/x-raw-int with random
	  depths between 1 and 32 bits.

2007-03-28 18:40:12 +0000  Stefan Kost <ensonic@users.sf.net>

	  gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792.
	  Original commit message from CVS:
	  Based on patch by: Stefan Kost  <ensonic@users.sf.net>
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
	  (gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
	  (gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
	  (gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
	  (gst_rtp_mp4a_depay_get_property),
	  (gst_rtp_mp4a_depay_change_state),
	  (gst_rtp_mp4a_depay_plugin_init):
	  * gst/rtp/gstrtpmp4adepay.h:
	  Added MP4A-LATM depayloader. Fixes #417792.
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	  (gst_rtp_mp4v_depay_process):
	  Fixup depayloader, setting codec_data, using more efficient adaptor and
	  rtpbuffer handling.
	  * gst/rtsp/URLS:
	  Add url to test above.

2007-03-28 15:17:27 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov files with h264 video).
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
	  (gst_qtdemux_chain), (qtdemux_parse_samples):
	  * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_ctts):
	  * gst/qtdemux/qtdemux_dump.h:
	  * gst/qtdemux/qtdemux_fourcc.h:
	  * gst/qtdemux/qtdemux_types.c:
	  Process 'ctts' atoms, which are present in AVC ISO files (.mov files
	  with h264 video).
	  Use the offset present in 'ctts' to calculate the PTS for each packet
	  and set the PTS on outgoing buffers.
	  Fixes #423283

2007-03-25 15:34:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field ...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
	  (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
	  (get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
	  (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_stream_configure_caps),
	  (gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
	  * gst/rtsp/gstrtspsrc.h:
	  Handle default clock-rates for static payload types, rearrange stuff so
	  that the rtpmap field in the sdp can override the defaults.
	  Parse RTP-Info field to get the seqnum and timebase fields that should
	  go in the caps.
	  Delay configuring caps after we got the RTP-Info from the PLAY reply from
	  the server.

2007-03-24 19:46:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging input caps into 1-channel output caps (I...
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps):
	  Remove 'channel-positions' field when munging input caps into
	  1-channel output caps (I guess technically we should set the
	  position for each channel on the output caps if it's non-NONE,
	  but I'll save that as a task for another day).

2007-03-22 22:14:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/interleave/deinterleave.c: Don't leak input buffer in chain function; maintain our own list of source pads - ther...
	  Original commit message from CVS:
	  * gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
	  (gst_deinterleave_remove_pads), (gst_deinterleave_process),
	  (gst_deinterleave_chain):
	  Don't leak input buffer in chain function; maintain our own list of
	  source pads - there are no guarantees about the order of the list
	  in the GstElement struct, and we want a very specific order; lastly,
	  some more debugging.

2007-03-22 16:25:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Revert last commit, preventing infinite plugging loops with ranks is no clean solution...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
	  Revert last commit, preventing infinite plugging loops with ranks
	  is no clean solution and in general there's no reason why one wants
	  to parse framed wavpack data again.

2007-03-22 15:52:51 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.c: Send the new segment event in time format instead of bytes. This allows "wavpackenc ! wa...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
	  Send the new segment event in time format instead of bytes. This
	  allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines.
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
	  Accept framed and non-framed input, wavpackparse doesn't care. To
	  prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the
	  rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse !
	  ..." pipelines.

2007-03-22 11:08:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackdec.c: Revert to use gst_pad_alloc_buffer() here. We can and should use it.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  Revert to use gst_pad_alloc_buffer() here. We can and should use it.
	  Thanks to Jan and Mike for noticing my mistake.

2007-03-22 09:44:17 +0000  Christophe Dehais <christophe.dehais@gmail.com>

	  ext/gconf/gconf.c: Accept complex pipeline descriptions as an audio profile instead of just a single element. Fixes #...
	  Original commit message from CVS:
	  Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>
	  * ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	  Accept complex pipeline descriptions as an audio profile instead of just
	  a single element. Fixes #420658.

2007-03-22 00:17:41 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.*: Put the write helpers into the GstWavpackEnc struct directly and not as a pointer to sav...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
	  (gst_wavpack_enc_init), (gst_wavpack_enc_chain),
	  (gst_wavpack_enc_rewrite_first_block):
	  * ext/wavpack/gstwavpackenc.h:
	  Put the write helpers into the GstWavpackEnc struct directly and not
	  as a pointer to save two small, but useless mallocs. This also makes
	  it possible to drop the finalize method.
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_push_buffer):
	  For consistency reasons also set GST_BUFFER_OFFSET_END on the outgoing
	  buffers the same way wavpackenc does it.

2007-03-21 23:50:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackdec.c: Don't use gst_pad_alloc_buffer() as we might clip the buffer later and
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  Don't use gst_pad_alloc_buffer() as we might clip the buffer later and
	  BaseTransform-based elements will likely break because of wrong
	  unit-size. Also plug a possible memleak that happens when decoding
	  fails for some reason.

2007-03-21 12:53:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/lame/gstlame.c: Disable the bitrate checking when the user has requested
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_setup):
	  Disable the bitrate checking when the user has requested
	  Free Format mode, as all bitrates less than the maximum
	  are valid then.

2007-03-21 11:49:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/apetag/gsttagdemux.c: Rename registered type in preparation of GstTagDemux moving to
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_get_type):
	  Rename registered type in preparation of GstTagDemux moving to
	  -base at some point in the future.

2007-03-19 10:29:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: Streaming mode fixes: don't unref buffer we don't own any longer; remove bogus adapter fl...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Streaming mode fixes: don't unref buffer we don't own any longer;
	  remove bogus adapter flush. Fixes #419338.

2007-03-18 04:21:28 +0000  David Schleef <ds@schleef.org>

	  REQUIREMENTS: Change the format to key/value, add a bunch of information, remove a bunch of requirements that are for...
	  Original commit message from CVS:
	  * REQUIREMENTS: Change the format to key/value, add a bunch of
	  information, remove a bunch of requirements that are for
	  other GStreamer packages.

2007-03-18 02:00:54 +0000  David Schleef <ds@schleef.org>

	  REQUIREMENTS: Fix a few things.  This file really needs a good once-over.
	  Original commit message from CVS:
	  * REQUIREMENTS: Fix a few things.  This file really needs a
	  good once-over.

2007-03-16 18:38:18 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/osxvideo/osxvideosink.m: Fix previous commit, we want to pass the NSView in the message.
	  Original commit message from CVS:
	  * sys/osxvideo/osxvideosink.m:
	  Fix previous commit, we want to pass the NSView in the message.

2007-03-16 16:27:20 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/osxvideo/osxvideosink.m: Emit 'have-ns-view' message when working in embedded mode. The message will contain a po...
	  Original commit message from CVS:
	  * sys/osxvideo/osxvideosink.m:
	  Emit 'have-ns-view' message when working in embedded mode. The message
	  will contain a pointer to the newly created NSView.

2007-03-16 09:57:40 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/gstiirequalizer10bands.c: A 10 band EQ should be initialized to 1 bands and not to 3.
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer10bands.c:
	  (gst_iir_equalizer_10bands_init):
	  A 10 band EQ should be initialized to 1 bands and not to 3.

2007-03-15 12:05:01 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/Makefile.am: Don't forget to distribute the sys/osxaudio/ directory.
	  Original commit message from CVS:
	  * sys/Makefile.am:
	  Don't forget to distribute the sys/osxaudio/ directory.

2007-03-15 11:39:53 +0000  Edward Hervey <bilboed@bilboed.com>

	  Activate osxaudio in gst-plugins-good with proper build setup.
	  Original commit message from CVS:
	  * configure.ac:
	  * sys/Makefile.am:
	  * sys/osxaudio/Makefile.am:
	  * sys/osxaudio/gstosxaudio.c:
	  * sys/osxaudio/gstosxaudiosink.c:
	  (gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_init),
	  (gst_osx_audio_sink_getcaps),
	  (gst_osx_audio_sink_create_ringbuffer), (plugin_init):
	  * sys/osxaudio/gstosxaudiosrc.c:
	  (gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_init),
	  (gst_osx_audio_src_create_ringbuffer):
	  * sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_get_type),
	  (gst_osx_ring_buffer_class_init), (gst_osx_ring_buffer_init),
	  (gst_osx_ring_buffer_acquire), (gst_osx_ring_buffer_start),
	  (gst_osx_ring_buffer_pause), (gst_osx_ring_buffer_stop):
	  * sys/osxaudio/gstosxringbuffer.h:
	  Activate osxaudio in gst-plugins-good with proper build setup.
	  Add inlined documentation.
	  Fix debug statements
	  Fix ringbuffer when pausing.
	  Fixes #323471

2007-03-14 22:21:26 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst/rtp/: Ported mulaw and alaw payloaders to use new base class
	  Original commit message from CVS:
	  * gst/rtp/gstrtppcmapay.c:
	  * gst/rtp/gstrtppcmapay.h:
	  * gst/rtp/gstrtppcmupay.c:
	  * gst/rtp/gstrtppcmupay.h:
	  Ported mulaw and alaw payloaders to use new base class

2007-03-14 16:30:19 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/osxvideo/: Fix leaks when running a NSApp.
	  Original commit message from CVS:
	  * sys/osxvideo/cocoawindow.h:
	  * sys/osxvideo/cocoawindow.m:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  Fix leaks when running a NSApp.
	  Accept any kind of resolutions.
	  Works in fullscreen. Can maximize.
	  Only thing left before being able to move this to -good is documentation
	  and embedded window support.

2007-03-14 15:25:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/: Update translations.
	  Original commit message from CVS:
	  * po/af.po:
	  * po/az.po:
	  * po/cs.po:
	  * po/en_GB.po:
	  * po/it.po:
	  * po/nl.po:
	  * po/or.po:
	  * po/sq.po:
	  * po/sr.po:
	  * po/sv.po:
	  * po/uk.po:
	  * po/vi.po:
	  Update translations.

2007-03-14 14:49:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Fix string replace error (AG_AG_GST_* => AG_GST_*).
	  Original commit message from CVS:
	  * configure.ac:
	  Fix string replace error (AG_AG_GST_* => AG_GST_*).

2007-03-14 14:48:08 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/: Add 3 and 10 band version and add missing gst_object_sync_values.
	  Original commit message from CVS:
	  * gst/equalizer/Makefile.am:
	  * gst/equalizer/gstiirequalizer.c: (_do_init),
	  (gst_iir_equalizer_band_set_property),
	  (gst_iir_equalizer_band_class_init),
	  (gst_iir_equalizer_band_get_type),
	  (gst_iir_equalizer_child_proxy_get_child_by_index),
	  (gst_iir_equalizer_child_proxy_get_children_count),
	  (gst_iir_equalizer_child_proxy_interface_init), (setup_filter),
	  (gst_iir_equalizer_compute_frequencies),
	  (gst_iir_equalizer_transform_ip), (plugin_init):
	  * gst/equalizer/gstiirequalizer10bands.c:
	  (gst_iir_equalizer_10bands_base_init),
	  (gst_iir_equalizer_10bands_class_init),
	  (gst_iir_equalizer_10bands_init),
	  (gst_iir_equalizer_10bands_set_property),
	  (gst_iir_equalizer_10bands_get_property):
	  * gst/equalizer/gstiirequalizer10bands.h:
	  * gst/equalizer/gstiirequalizer3bands.c:
	  (gst_iir_equalizer_3bands_base_init),
	  (gst_iir_equalizer_3bands_class_init),
	  (gst_iir_equalizer_3bands_init),
	  (gst_iir_equalizer_3bands_set_property),
	  (gst_iir_equalizer_3bands_get_property):
	  * gst/equalizer/gstiirequalizer3bands.h:
	  * gst/equalizer/gstiirequalizernbands.c:
	  (gst_iir_equalizer_nbands_base_init),
	  (gst_iir_equalizer_nbands_init):
	  Add 3 and 10 band version and add missing gst_object_sync_values.
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_event),
	  (gst_spectrum_transform_ip):
	  Add some comments about float support.

2007-03-12 17:56:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/apetag/gsttagdemux.c: Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END her...
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
	  Fix handling of -1 values for start and stop values when seeking,
	  and SEEK_CUR+SEEK_END here as well.

2007-03-12 17:24:23 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/gstid3demux.c: Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END.
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
	  Fix handling of -1 values for start and stop values when seeking,
	  and SEEK_CUR+SEEK_END.

2007-03-12 15:49:02 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  I'm too lazy to comment this
	  Original commit message from CVS:
	  Add Patch by: line for wim, since he's away

2007-03-12 13:28:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is the image format a vari...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
	  the image format a variable-length NUL-terminated string; in
	  versions before that the image format is a fixed-length string of
	  3 characters (see #348644 for a sample tag).
	  Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.

2007-03-11 22:23:04 +0000  Sébastien Moutte <sebastien@moutte.net>

	  sys/directdraw/gstdirectdrawsink.*: Handle display mode changes during playback.
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawsink.c:
	  * sys/directdraw/gstdirectdrawsink.h:
	  Handle display mode changes during playback.

2007-03-10 16:07:31 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/MANIFEST: Add new project files to MANIFEST.
	  Original commit message from CVS:
	  * win32/MANIFEST:
	  Add new project files to MANIFEST.
	  * win32/vs6/libgstaudiofx.dsp:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgstrtsp.dsp:
	  Update project files.

2007-03-10 12:30:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Printf format fixes; also add some missing quotes in translated strings. Fixes #416728 and #416727.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
	  (gst_avi_demux_parse_index):
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  Printf format fixes; also add some missing quotes in translated
	  strings. Fixes #416728 and #416727.

2007-03-09 20:12:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/autodetect/gstautoaudiosink.c: Tim and I can't think of any reason the child audio sink needs to be set back to N...
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
	  Tim and I can't think of any reason the child audio sink needs to
	  be set back to NULL after successfully determining that it can
	  reach READY - it gets immediately set back to READY by the caller
	  anyway, causing an unnecessary close/open of any audio devices
	  involved.

2007-03-09 19:51:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/: Add ja.po file from #377306.
	  Original commit message from CVS:
	  * po/LINGUAS:
	  * po/ja.po:
	  Add ja.po file from #377306.

2007-03-09 19:44:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/sunaudio/: Actually translate sunaudio mixer track labels instead of just marking the strings as translatable (#3...
	  Original commit message from CVS:
	  * sys/sunaudio/gstsunaudio.c: (plugin_init):
	  * sys/sunaudio/gstsunaudiomixertrack.c:
	  (gst_sunaudiomixer_track_new):
	  Actually translate sunaudio mixer track labels instead of just
	  marking the strings as translatable (#377306); clean up weird
	  label string mapping code that serves no apparent purpose. Also
	  set the 'untranslated-label' property when creating mixer tracks
	  if the GstMixerTrack base class supports this.
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/sunaudio.c: (GST_START_TEST),
	  (sunaudio_suite):
	  Very minimalistic unit test for sunaudiomixer element (compiles, but not
	  actually tested on a system where sunaudiomixer is available).

2007-03-09 18:49:37 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Re-enable the states test and see if it works on the buildbots.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Re-enable the states test and see if it works on the buildbots.

2007-03-09 17:32:32 +0000  Wim Taymans <wim@fluendo.com>

	  ext/dv/gstdvdec.*: Infer pixel-aspect-ratio from the video frame format if it isn't provided by the container, as hap...
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
	  (gst_dvdec_src_negotiate), (gst_dvdec_chain),
	  (gst_dvdec_change_state):
	  * ext/dv/gstdvdec.h:
	  Infer pixel-aspect-ratio from the video frame format if it isn't
	  provided by the container, as happens when playing DV from AVI
	  or Quicktime containers.
	  Patch by: Wim Taymans <wim@fluendo.com>
	  Fixes #380944

2007-03-09 17:05:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: When activated, remove the udpsrc timeout, we have dataflow and timeouts will later be handled...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
	  When activated, remove the udpsrc timeout, we have dataflow and timeouts
	  will later be handled by the jitterbuffer.

2007-03-09 16:53:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/taglib/gstid3v2mux.cc: Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc:
	  Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	  Fixes #414496.

2007-03-09 15:04:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Fix stream position reporting after a seek. Fixes #416445.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_push_event), (gst_avi_demux_do_seek),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	  (gst_avi_demux_chain):
	  Fix stream position reporting after a seek. Fixes #416445.

2007-03-09 08:58:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/equalizer/: Refactor plugin into a base class and a first subclass (nband eq). The nband eq uses GstChildProxy an...
	  Original commit message from CVS:
	  * gst/equalizer/Makefile.am:
	  * gst/equalizer/gstiirequalizer.c: (_do_init),
	  (gst_iir_equalizer_band_set_property),
	  (gst_iir_equalizer_band_get_property),
	  (gst_iir_equalizer_band_class_init),
	  (gst_iir_equalizer_band_get_type),
	  (gst_iir_equalizer_child_proxy_get_child_by_index),
	  (gst_iir_equalizer_child_proxy_get_children_count),
	  (gst_iir_equalizer_child_proxy_interface_init),
	  (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
	  (gst_iir_equalizer_finalize), (setup_filter),
	  (gst_iir_equalizer_compute_frequencies),
	  (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
	  (gst_iir_equalizer_setup), (plugin_init):
	  * gst/equalizer/gstiirequalizer.h:
	  * gst/equalizer/gstiirequalizernbands.c:
	  (gst_iir_equalizer_nbands_base_init),
	  (gst_iir_equalizer_nbands_class_init),
	  (gst_iir_equalizer_nbands_init),
	  (gst_iir_equalizer_nbands_set_property),
	  (gst_iir_equalizer_nbands_get_property):
	  * gst/equalizer/gstiirequalizernbands.h:
	  Refactor plugin into a base class and a first subclass (nband eq). The
	  nband eq uses GstChildProxy and is controlable. More subclasses will
	  follow.

2007-03-08 16:01:42 +0000  René Stadler <mail@renestadler.de>

	  gst/avi/gstavidemux.c: Make avidemux accept optional header chunks in any order.
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler dot de>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
	  (gst_avi_demux_stream_data), (gst_avi_demux_chain):
	  Make avidemux accept optional header chunks in any order.
	  Fixes #415446.

2007-03-08 12:23:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Disable the states check until the remaining Valgrind errors are fixed or suppressed.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Disable the states check until the remaining Valgrind errors
	  are fixed or suppressed.

2007-03-08 10:24:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/elements/.cvsignore: Add audiodynamic check to .cvsignore
	  Original commit message from CVS:
	  * tests/check/elements/.cvsignore:
	  Add audiodynamic check to .cvsignore

2007-03-08 10:02:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Add new audiodynamic element which can act as a compressor or expander. Supported are hard-knee and sof...
	  Original commit message from CVS:
	  reviewed by: Stefan Kost  <ensonic@users.sf.net>
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiodynamic.c:
	  (gst_audio_dynamic_characteristics_get_type),
	  (gst_audio_dynamic_mode_get_type),
	  (gst_audio_dynamic_set_process_function),
	  (gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
	  (gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
	  (gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
	  (gst_audio_dynamic_transform_hard_knee_compressor_int),
	  (gst_audio_dynamic_transform_hard_knee_compressor_float),
	  (gst_audio_dynamic_transform_soft_knee_compressor_int),
	  (gst_audio_dynamic_transform_soft_knee_compressor_float),
	  (gst_audio_dynamic_transform_hard_knee_expander_int),
	  (gst_audio_dynamic_transform_hard_knee_expander_float),
	  (gst_audio_dynamic_transform_soft_knee_expander_int),
	  (gst_audio_dynamic_transform_soft_knee_expander_float),
	  (gst_audio_dynamic_transform_ip):
	  * gst/audiofx/audiodynamic.h:
	  * gst/audiofx/audiofx.c: (plugin_init):
	  Add new audiodynamic element which can act as a compressor or
	  expander. Supported are hard-knee and soft-knee operation modes with
	  user-specified ratio and threshold.
	  Attack and release parameters are not yet implemented but will follow.
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  Integrate audiodynamic into the docs.
	  * tests/check/Makefile.am:
	  * tests/check/elements/audiodynamic.c: (setup_dynamic),
	  (cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
	  Add unit test for audiodynamic.

2007-03-07 19:48:03 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/raw1394/gstdv1394src.c: Free handles that we allocated when exiting via the error paths.
	  Original commit message from CVS:
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
	  Free handles that we allocated when exiting via the error paths.

2007-03-07 12:07:07 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: Use a general wavpack debug category for common code.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpack.c: (plugin_init):
	  * ext/wavpack/gstwavpackcommon.c:
	  Use a general wavpack debug category for common code.
	  * ext/wavpack/gstwavpackstreamreader.c:
	  (gst_wavpack_stream_reader_set_pos_abs),
	  (gst_wavpack_stream_reader_set_pos_rel),
	  (gst_wavpack_stream_reader_write_bytes):
	  Use the general wavpack debug category here too and add debug
	  output to the functions that should not be called at all by
	  the wavpack library.
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_plugin_init):
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_plugin_init):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
	  Change debugging category names to conform to the conventions.

2007-03-07 11:37:23 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.*: Share qtdemux debug category across all files, otherwise all debugging in files other than qtd...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c:
	  * gst/qtdemux/qtdemux.h:
	  Share qtdemux debug category across all files, otherwise all debugging
	  in files other than qtdemux.c would end up in the default category.

2007-03-07 11:24:05 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/level/gstlevel.*: Resolve message timestamps against the playback segment.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_class_init),
	  (gst_level_set_caps), (gst_level_start), (gst_level_event),
	  (gst_level_transform_ip):
	  * gst/level/gstlevel.h:
	  Resolve message timestamps against the playback segment.

2007-03-07 11:23:20 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.*: One FIXME less, by resolving message timestamps against the playback segment.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_start),
	  (gst_spectrum_event), (gst_spectrum_transform_ip):
	  * gst/spectrum/gstspectrum.h:
	  One FIXME less, by resolving message timestamps against the playback
	  segment.

2007-03-06 23:21:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  Fix ChangeLog message
	  Original commit message from CVS:
	  Fix ChangeLog message

2007-03-06 23:19:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/gstid3demux.c: Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the caps passed to ...
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
	  (gst_id3demux_sink_activate):
	  Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
	  caps passed to it (previouslly one code path assumes it takes ownership
	  while another one assumes it doesn't).
	  * configure.ac:
	  * tests/files/Makefile.am:
	  * tests/files/id3-407349-1.tag:
	  * tests/files/id3-407349-2.tag:
	  Add directory where data for unit tests can be stored.
	  * tests/Makefile.am:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
	  (read_tags_from_file), (run_check_for_file),
	  (check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
	  Add unit test for id3demux, and in particular for bug #407349. Only
	  testing pull-mode for now; push mode doesn't work yet because the test
	  files are smaller than ID3_TYPE_FIND_MIN_SIZE.

2007-03-06 22:14:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Add missing backslash at end of line.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Add missing backslash at end of line.

2007-03-06 18:36:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* common:
	  Trigger rebuild.
	  Original commit message from CVS:
	  Trigger rebuild.

2007-03-06 18:16:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/: Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise the four-digit number will be interp...
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
	  * gst/id3demux/id3tags.h:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	  (parse_obsolete_tdat_frame):
	  Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
	  the four-digit number will be interpreted as a year, whereas it is
	  month and day in DDMM format. Instead, parse TDAT frames and fix up
	  the date in the GST_TAG_DATE tag later if we also extracted a year.
	  Fixes #407349.

2007-03-06 14:53:04 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/gconf/gstswitchsink.c: Fix up the dispose logic so it doesn't leak, and fix setting of the child state so that we...
	  Original commit message from CVS:
	  * ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
	  (gst_switch_commit_new_kid):
	  Fix up the dispose logic so it doesn't leak, and fix setting of
	  the child state so that we don't set a child to our current state
	  just as we are changing it to something else.

2007-03-06 13:57:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/spectrum/gstspectrum.c: Fix and cleanup default property values.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
	  (gst_spectrum_init), (gst_spectrum_set_property),
	  (gst_spectrum_transform_ip):
	  Fix and cleanup default property values.
	  Add FIXMEs for stuff that looks rather wrong.

2007-03-06 13:21:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/goom/gstgoom.*: Document, fix and improve goom adapter behaviour.
	  Original commit message from CVS:
	  * gst/goom/gstgoom.c: (gst_goom_src_setcaps), (get_buffer),
	  (gst_goom_chain):
	  * gst/goom/gstgoom.h:
	  Document, fix and improve goom adapter behaviour.
	  Fixes #407006.

2007-03-05 18:43:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/esd/esdsink.c: Unref static pad template after using it.
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_open):
	  Unref static pad template after using it.

2007-03-05 17:17:04 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/gconf/gstswitchsink.c: Fix up the reference counting of the child elements.
	  Original commit message from CVS:
	  * ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
	  (gst_switch_commit_new_kid):
	  Fix up the reference counting of the child elements.

2007-03-05 17:08:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Fix encoding-name case.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
	  * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_finish_headers):
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
	  Fix encoding-name case.

2007-03-05 16:39:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Fix speex (de)payloader. Fixes #358040.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init),
	  (gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps),
	  (gst_rtp_speex_depay_process):
	  * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init),
	  (gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps),
	  (gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer),
	  (gst_rtp_speex_pay_change_state):
	  * gst/rtp/gstrtpspeexpay.h:
	  Fix speex (de)payloader. Fixes #358040.

2007-03-05 15:42:58 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/gconf/gstswitchsink.c: Install fakesink in NULL by fixing some broken logic. This obviates the need to manually s...
	  Original commit message from CVS:
	  * ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
	  (gst_switch_commit_new_kid), (gst_switch_sink_set_child):
	  Install fakesink in NULL by fixing some broken logic. This obviates
	  the need to manually set _IS_SINK.
	  Add some comments and remove a little cruft while I'm at it.

2007-03-05 14:46:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/gconf/gstswitchsink.c: Mark us as a sink when we have no fakesink in NULL. Fixes #414887.
	  Original commit message from CVS:
	  * ext/gconf/gstswitchsink.c: (gst_switch_sink_reset):
	  Mark us as a sink when we have no fakesink in NULL. Fixes #414887.

2007-03-05 08:30:52 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/: Remove two obsolete and confusing comments.
	  Original commit message from CVS:
	  * gst/spectrum/demo-audiotest.c: (message_handler):
	  * gst/spectrum/demo-osssrc.c: (message_handler):
	  Remove two obsolete and confusing comments.

2007-03-04 18:52:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/POTFILES.in: Update.
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  Update.

2007-03-04 17:33:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Gah! Also disable gconfvideosink from the tests, otherwise it will instantiate autovideosink...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Gah! Also disable gconfvideosink from the tests, otherwise
	  it will instantiate autovideosink, and dfbvideosink and
	  leak on the buildbots.

2007-03-04 17:13:19 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/cdio/gstcdiocddasrc.c: Make sure we always destroy our libcdio handle.
	  Original commit message from CVS:
	  * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open),
	  (gst_cdio_cdda_src_finalize):
	  Make sure we always destroy our libcdio handle.

2007-03-04 17:05:58 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Disable autovideosink so the buildbots don't barf over memory leaked in the directfb sink.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Disable autovideosink so the buildbots don't barf over memory
	  leaked in the directfb sink.

2007-03-04 15:28:30 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/ximage/gstximagesrc.c: Chain up in dispose
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_dispose):
	  Chain up in dispose

2007-03-04 15:07:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/multipart/multipartdemux.c: Use gst_pad_new_from_static_template instead of static_pad_template_get+pad_new.
	  Original commit message from CVS:
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
	  (gst_multipart_find_pad_by_mime):
	  Use gst_pad_new_from_static_template instead of
	  static_pad_template_get+pad_new.

2007-03-04 14:56:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/ximage/gstximagesrc.c: Catch the case where no clock has been set.
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_create):
	  Catch the case where no clock has been set.

2007-03-04 13:52:03 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Fix a bunch of leaks shown by the newly-added states test.
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flac_enc_finalize):
	  * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init),
	  (gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize):
	  * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
	  (gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose),
	  (gst_gconf_audio_src_finalize), (do_toggle_element):
	  * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init),
	  (gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize),
	  (do_toggle_element):
	  * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
	  (gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose),
	  (gst_gconf_video_src_finalize), (do_toggle_element):
	  * ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init),
	  (gst_switch_sink_reset), (gst_switch_sink_set_child):
	  * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
	  * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
	  * ext/shout2/gstshout2.c: (gst_shout2send_class_init),
	  (gst_shout2send_init), (gst_shout2send_finalize):
	  * gst/debug/testplugin.c: (gst_test_class_init),
	  (gst_test_finalize):
	  * gst/flx/gstflxdec.c: (gst_flxdec_class_init),
	  (gst_flxdec_dispose):
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_finalize):
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize):
	  * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context):
	  * gst/rtsp/rtspextwms.h:
	  * gst/smpte/gstsmpte.c: (gst_smpte_class_init),
	  (gst_smpte_finalize):
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize):
	  * gst/udp/gstudpsink.c: (gst_udpsink_class_init),
	  (gst_udpsink_finalize):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_dispose),
	  (gst_wavparse_sink_activate):
	  * sys/oss/gstosssink.c: (gst_oss_sink_finalise):
	  * sys/oss/gstosssrc.c: (gst_oss_src_class_init),
	  (gst_oss_src_finalize):
	  * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy):
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	  (gst_v4l2src_finalize):
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
	  Fix a bunch of leaks shown by the newly-added states test.

2007-03-04 13:41:00 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/dv/gstdvdec.c: Use gst_pad_new_from_static_template instead of static_pad_template_get+pad_new.
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c: (gst_dvdec_init):
	  Use gst_pad_new_from_static_template instead of
	  static_pad_template_get+pad_new.

2007-03-03 13:06:21 +0000  Loïc Minier <lool+gnome@via.ecp.fr>

	  Don't mix tabs and spaces (#414168).
	  Original commit message from CVS:
	  Patch by: Loïc Minier <lool+gnome at via ecp fr>
	  * ext/libcaca/Makefile.am:
	  * gst/debug/Makefile.am:
	  Don't mix tabs and spaces (#414168).

2007-03-02 21:35:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/generic/.cvsignore: Ignore files to please buildbot.
	  Original commit message from CVS:
	  * tests/check/generic/.cvsignore:
	  Ignore files to please buildbot.

2007-03-02 21:01:19 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.c: Unbreak my previous commit (swapped nominator & denominator). Tim, thanks for spotting.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
	  (gst_wavparse_stream_data):
	  Unbreak my previous commit (swapped nominator & denominator). Tim,
	  thanks for spotting.

2007-03-02 16:08:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/cdio/gstcdiocddasrc.c: Small code cleanups.
	  Original commit message from CVS:
	  * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_probe_devices),
	  (gst_cdio_cdda_src_read_sector), (gst_cdio_cdda_src_open),
	  (gst_cdio_cdda_src_finalize):
	  Small code cleanups.
	  Don't use pad_alloc as the base class cannot deal with the error codes.

2007-03-02 13:40:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Fix doc.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	  (gst_udpsrc_create):
	  Fix doc.

2007-03-02 13:29:25 +0000  René Stadler <mail@renestadler.de>

	  gst/wavparse/gstwavparse.c: Handle rounding better to not drop last sample frame. Fixes #356692
	  Original commit message from CVS:
	  Patch by: René Stadler <mail@renestadler.de>
	  * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	  (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	  (gst_wavparse_stream_data):
	  Handle rounding better to not drop last sample frame. Fixes #356692

2007-03-02 13:19:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Disable cacasink from the states check too - it also calls exit(1) on us when it can't find ...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Disable cacasink from the states check too - it also calls exit(1)
	  on us when it can't find a terminal to talk to.

2007-03-02 12:56:13 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/udp/gstudpsrc.*: Add support to strip proprietary headers. Fixes #350296.
	  Original commit message from CVS:
	  Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	  (gst_udpsrc_create), (gst_udpsrc_set_property),
	  (gst_udpsrc_get_property):
	  * gst/udp/gstudpsrc.h:
	  Add support to strip proprietary headers. Fixes #350296.

2007-03-02 12:52:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp2tdepay.c: Fix compilation.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
	  Fix compilation.

2007-03-02 12:16:16 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/rtp/gstrtpmp2tdepay.*: Add support to strip off proprietary headers. Fixes #350278.
	  Original commit message from CVS:
	  Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_class_init),
	  (gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_process),
	  (gst_rtp_mp2t_depay_set_property),
	  (gst_rtp_mp2t_depay_get_property):
	  * gst/rtp/gstrtpmp2tdepay.h:
	  Add support to strip off proprietary headers. Fixes #350278.

2007-03-02 11:22:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/hal/hal.c: Fix compilation.
	  Original commit message from CVS:
	  * ext/hal/hal.c:
	  Fix compilation.

2007-03-02 10:54:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/sunaudio/gstsunaudiosrc.*: Remove device-name from GstSunAudioSrc. Fixes #412597.
	  Original commit message from CVS:
	  * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_class_init),
	  (gst_sunaudiosrc_init), (gst_sunaudiosrc_get_property),
	  (gst_sunaudiosrc_open):
	  * sys/sunaudio/gstsunaudiosrc.h:
	  Remove device-name from GstSunAudioSrc. Fixes #412597.

2007-03-01 21:50:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/hal/: Having NULL as UDI previously selected the default sink/src. Change this back but mention it in the debug o...
	  Original commit message from CVS:
	  * ext/hal/gsthalaudiosink.c: (do_toggle_element):
	  * ext/hal/gsthalaudiosrc.c: (do_toggle_element):
	  Having NULL as UDI previously selected the default sink/src. Change
	  this back but mention it in the debug output.
	  * ext/hal/hal.c: (gst_hal_get_alsa_element),
	  (gst_hal_get_oss_element), (gst_hal_get_string),
	  (gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
	  (gst_hal_get_audio_src):
	  * ext/hal/hal.h:
	  Refactor a bit, check all error conditions, greatly improve debugging
	  and fix some possible memory leaks. Also implement OSS support
	  and allow specifying an UDI that points to a real device. For this the
	  child device which supports ALSA (preferred) or OSS is used.
	  As a side effect this makes it impossible now to get a alsasink in
	  halaudiosrc and a alsasrc in halaudiosink.

2007-03-01 18:47:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Errors from the udp sources are not fatal unless all of them are in error.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
	  (find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
	  Errors from the udp sources are not fatal unless all of them are in
	  error.

2007-03-01 18:14:42 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Disable aasink in the states test. I suspect this is the element that is calling exit(1) whe...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Disable aasink in the states test. I suspect this is the element that
	  is calling exit(1) when it can't proceed.

2007-03-01 17:26:30 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Draw plugins in from the build tree sys/ dir, rather than picking up the already installed v...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Draw plugins in from the build tree sys/ dir, rather than picking
	  up the already installed versions.

2007-03-01 10:44:36 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximage/gstximagesrc.c: Error out correctly when getting xcontext fails.
	  Original commit message from CVS:
	  2007-03-01  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_open_display):
	  Error out correctly when getting xcontext fails.

2007-03-01 09:29:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's what it will be in the future and rtspsrc...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
	  Make state change to PAUSED NO_PREROLL because that's what it will be in
	  the future and rtspsrc relies on it.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_change_state):
	  Don't error out when we don't get an error from the state change
	  function.

2007-03-01 01:48:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/hal/: Check if the device UDI is set before trying to query HAL about it and give a useful error message if it wa...
	  Original commit message from CVS:
	  * ext/hal/gsthalaudiosink.c: (do_toggle_element):
	  * ext/hal/gsthalaudiosrc.c: (do_toggle_element):
	  Check if the device UDI is set before trying to query HAL
	  about it and give a useful error message if it wasn't set.
	  * ext/hal/hal.c: (gst_hal_get_string):
	  Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL
	  gives an assertion failure in D-Bus when running with
	  DBUS_FATAL_WARNINGS=1.

2007-02-28 19:29:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* win32/common/config.h:
	  update config to trunk
	  Original commit message from CVS:
	  update config to trunk

2007-02-28 19:29:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: Convert to new AG_GST style.
	  Original commit message from CVS:
	  * configure.ac:
	  Convert to new AG_GST style.

2007-02-28 18:41:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/lame/gstlame.c: Display sensible defaults and limits for the vbr-min/max/mean properties. Fix the 'hard-limit' VB...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init),
	  (gst_lame_setup):
	  Display sensible defaults and limits for the
	  vbr-min/max/mean properties. Fix the 'hard-limit' VBR min
	  property - it's supposed to be a boolean 0/1 value.

2007-02-28 16:01:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/lame/gstlame.c: Initialise the variables so gcc doesn't complain about possibly uninitialised uses, even though t...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c:
	  Initialise the variables so gcc doesn't complain about possibly
	  uninitialised uses, even though they can't actually happen.

2007-02-28 12:59:43 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/check/: add test for states
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/generic/states.c: (GST_START_TEST), (states_suite):
	  add test for states

2007-02-28 10:58:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/elements/.cvsignore: Add new videofilter check to .cvsignore.
	  Original commit message from CVS:
	  * tests/check/elements/.cvsignore:
	  Add new videofilter check to .cvsignore.

2007-02-28 10:54:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Fix combined flow return. Fixes #412608.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	  (gst_avi_demux_loop), (gst_avi_demux_chain):
	  Fix combined flow return. Fixes #412608.

2007-02-28 10:41:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videofilter/Makefile.am: Dist header..
	  Original commit message from CVS:
	  * gst/videofilter/Makefile.am:
	  Dist header..

2007-02-28 10:29:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videofilter/gstgamma.h: Add header too.
	  Original commit message from CVS:
	  * gst/videofilter/gstgamma.h:
	  Add header too.

2007-02-28 10:17:15 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/videofilter/: Port gamma filter to 0.10. Fixes #412704.
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw at skynet be>
	  * gst/videofilter/Makefile.am:
	  * gst/videofilter/gstgamma.c: (gst_gamma_base_init),
	  (gst_gamma_class_init), (gst_gamma_init), (gst_gamma_set_property),
	  (gst_gamma_get_property), (gst_gamma_calculate_tables),
	  (oil_tablelookup_u8), (gst_gamma_set_caps),
	  (gst_gamma_planar411_ip), (gst_gamma_transform_ip), (plugin_init):
	  Port gamma filter to 0.10. Fixes #412704.
	  * tests/check/Makefile.am:
	  * tests/check/elements/videofilter.c: (setup_filter),
	  (cleanup_filter), (check_filter), (GST_START_TEST),
	  (videobalance_suite), (videoflip_suite), (gamma_suite), (main):
	  Add unit tests for videofilters.

2007-02-28 10:06:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/URLS: Add another interesting test url.
	  Original commit message from CVS:
	  * gst/rtsp/URLS:
	  Add another interesting test url.
	  * gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
	  Don't allow getting header fields from data packets.

2007-02-27 23:43:08 +0000  Michael Smith <msmith@xiph.org>

	  ext/shout2/gstshout2.*: Add a property for username.
	  Original commit message from CVS:
	  * ext/shout2/gstshout2.c: (gst_shout2send_class_init),
	  (gst_shout2send_init), (gst_shout2send_start),
	  (gst_shout2send_set_property), (gst_shout2send_get_property):
	  * ext/shout2/gstshout2.h:
	  Add a property for username.

2007-02-27 12:02:03 +0000  Christian Schaller <uraeus@gnome.org>

	* sys/directdraw/gstdirectdrawplugin.c:
	* sys/directdraw/gstdirectdrawsink.c:
	* sys/directdraw/gstdirectdrawsink.h:
	* sys/directsound/gstdirectsoundplugin.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  update copyright statements
	  Original commit message from CVS:
	  update copyright statements

2007-02-27 11:59:21 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxaudio/gstosxaudioelement.c:
	* sys/osxaudio/gstosxaudioelement.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	  update copyright statement
	  Original commit message from CVS:
	  update copyright statement

2007-02-27 11:30:19 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/osxvideo/: Disable the cocoa event loop since it's a huge memory leak. Should only matter if the sink isn't used ...
	  Original commit message from CVS:
	  * sys/osxvideo/cocoawindow.h:
	  * sys/osxvideo/cocoawindow.m:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  Disable the cocoa event loop since it's a huge memory leak. Should only
	  matter if the sink isn't used within an NSApp (which has already got
	  a coca event loop).
	  Remove all unused code.

2007-02-26 12:07:14 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/rtsp/Makefile.am: Fix make check too.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  Fix make check too.

2007-02-26 10:00:28 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/rtsp/base64.*: Commit missing files for base64 encoding.
	  Original commit message from CVS:
	  * gst/rtsp/base64.c: (util_base64_encode):
	  * gst/rtsp/base64.h:
	  Commit missing files for base64 encoding.

2007-02-24 22:57:49 +0000  Loïc Minier <lool+gnome@via.ecp.fr>

	  Fix build with LDFLAGS='-Wl,-z,defs' (#410997)
	  Original commit message from CVS:
	  Patch by: Loïc Minier <lool+gnome at via ecp fr>
	  * configure.ac:
	  * ext/annodex/Makefile.am:
	  * ext/jpeg/Makefile.am:
	  * ext/speex/Makefile.am:
	  * gst/alpha/Makefile.am:
	  * gst/cutter/Makefile.am:
	  * gst/debug/Makefile.am:
	  * gst/effectv/Makefile.am:
	  * gst/goom/Makefile.am:
	  * gst/level/Makefile.am:
	  * gst/smpte/Makefile.am:
	  * gst/videofilter/Makefile.am:
	  Fix build with LDFLAGS='-Wl,-z,defs' (#410997)

2007-02-24 22:52:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Fix build with LDFLAGS='-Wl,-z,defs'.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/gsm/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/wavpack/Makefile.am:
	  * gst/equalizer/Makefile.am:
	  * gst/filter/Makefile.am:
	  * gst/mve/Makefile.am:
	  * gst/nsf/Makefile.am:
	  * gst/replaygain/Makefile.am:
	  * gst/speed/Makefile.am:
	  Fix build with LDFLAGS='-Wl,-z,defs'.

2007-02-23 19:12:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken from icecast to replace it. Relicensed fr...
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/rtspconnection.c: (append_auth_header),
	  (rtsp_connection_send), (rtsp_connection_set_auth):
	  g_base64_encode is a GLib 2.12 function. Use an equivalent taken
	  from icecast to replace it. Relicensed from GPL courtesy of Mike
	  Smith.

2007-02-23 18:12:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	  (gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
	  (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
	  (gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
	  (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	  (gst_rtspsrc_uri_set_uri):
	  * gst/rtsp/gstrtspsrc.h:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	  (append_auth_header), (rtsp_connection_send),
	  (rtsp_connection_free), (rtsp_connection_set_auth):
	  * gst/rtsp/rtspconnection.h:
	  * gst/rtsp/rtspdefs.h:
	  * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
	  * gst/rtsp/rtspurl.h:
	  Implement simple Basic Authentication support so that urls like
	  rtsp://user:pass@hostname/rtspstream work on hosts that require
	  authentication.

2007-02-22 17:53:26 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/v4l2_calls.c:
	  Fix segfault when oppening a radio device.
	  Original commit message from CVS:
	  Fix segfault when oppening a radio device.

2007-02-22 14:35:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix level for multi-channel case.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_set_caps),
	  (gst_level_transform_ip):
	  * sys/v4l2/README:
	  * tests/check/elements/level.c: (GST_START_TEST):
	  Fix level for multi-channel case.

2007-02-21 16:02:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/lame/gstlame.c: Fix up bitrate checking macro.  Make it give us a
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_setcaps),
	  (gst_lame_set_property), (gst_lame_setup):
	  Fix up bitrate checking macro.  Make it give us a
	  GST_ELEMENT_WARNING message so the application has a chance of
	  reporting this to the user.  Move the checking to _setup, so we
	  are sure it runs in the READY state, when we hope to have a pipeline
	  and a bus that is not flushing.
	  This fixes e.g. using 96 kbit/sec as a bitrate.

2007-02-21 10:18:12 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/level/gstlevel.*: Use function pointer for process function and add process functions for float audio.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
	  (gst_level_transform_ip):
	  * gst/level/gstlevel.h:
	  Use function pointer for process function and add process functions
	  for float audio.

2007-02-20 21:34:00 +0000  Sébastien Moutte <sebastien@moutte.net>

	  sys/directsound/gstdirectsoundsink.*: Remove include of unused headers.
	  Original commit message from CVS:
	  * sys/directsound/gstdirectsoundsink.c:
	  * sys/directsound/gstdirectsoundsink.h:
	  Remove include of unused headers.
	  * sys/waveform/gstwaveformplugin.c:
	  * sys/waveform/gstwaveformsink.c:
	  * sys/waveform/gstwaveformsink.h:
	  * win32/vs6/libgstwaveform.dsp:
	  Add a new waveform plugin which includes an audio sink
	  element using the WaveForm win32 API.
	  * win32/MANIFEST:
	  Add the new project file form waveform plugin.

2007-02-19 12:22:43 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/v4l2src_calls.c: Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO, fixes #407369
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	  (gst_v4l2src_capture_init):
	  Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO,
	  fixes #407369

2007-02-18 18:00:51 +0000  Sébastien Moutte <sebastien@moutte.net>

	  sys/directdraw/: Prepare the plugin to move to good:
	  Original commit message from CVS:
	  * sys/directdraw/gstdirectdrawplugin.c:
	  * sys/directdraw/gstdirectdrawsink.c:
	  * sys/directdraw/gstdirectdrawsink.h:
	  Prepare the plugin to move to good:
	  Remove unused/untested code (rendering to an extern surface,
	  yuv format rendering).Use GST_(DEBUG/*)_OBJECT macros
	  Rename all functions from gst_directdrawsink to gst_directdraw_sink.
	  Add gtk doc section
	  Fix a bug in gst_directdraw_sink_show_frame, memcpy line by line
	  respecting destination surface stride.
	  * sys/directsound/gstdirectsoundplugin.c:
	  * sys/directsound/gstdirectsoundsink.c:
	  * sys/directsound/gstdirectsoundsink.h:
	  Prepare the plugin to move to good:
	  Rename all functions from gst_directsoundsink to gst_directsound_sink.
	  Add gtk doc section
	  * win32/common/config.h.in:
	  * win32/MANIFEST:
	  Add config.h.in

2007-02-18 13:24:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added simple mpeg transport stream payloader.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_base_init),
	  (gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init),
	  (gst_rtp_mp2t_pay_setcaps), (gst_rtp_mp2t_pay_handle_buffer),
	  (gst_rtp_mp2t_pay_plugin_init):
	  * gst/rtp/gstrtpmp2tpay.h:
	  Added simple mpeg transport stream payloader.

2007-02-16 12:32:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/URLS: Add example H264 rtsp url.
	  Original commit message from CVS:
	  * gst/rtsp/URLS:
	  Add example H264 rtsp url.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	  Don't convert values to lowercase or we might mess up base64 encoded
	  properties.

2007-02-16 12:30:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/README: Fix case of string params.
	  Original commit message from CVS:
	  * gst/rtp/README:
	  Fix case of string params.
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	  (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	  Fix depayloader, support more packet types.
	  Add sync codes to make sure the packetizer can do its job.
	  * gst/rtp/gstrtpmp4gdepay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
	  Fix caps case again.

2007-02-15 12:26:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph264depay.c: Set right caps on output buffers.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
	  Set right caps on output buffers.

2007-02-14 17:04:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/sdpmessage.c: Clear stack allocated SDPMedia struct before calling _init() on it.
	  Original commit message from CVS:
	  * gst/rtsp/sdpmessage.c: (sdp_parse_line):
	  As spotted by: Peter Kjellerstedt  <pkj at axis com>:
	  Clear stack allocated SDPMedia struct before calling _init() on it.
	  Clarify this in the docs as well.

2007-02-14 17:01:25 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/gconf/gstgconfaudiosink.c: Don't reset the profile when going switching states, as it makes the element non-reusa...
	  Original commit message from CVS:
	  * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
	  (do_change_child):
	  Don't reset the profile when going switching states, as it makes
	  the element non-reusable.

2007-02-14 15:24:50 +0000  jp.liu <jp_liu@astrocom.cn>

	  gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793.
	  Original commit message from CVS:
	  * gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init),
	  (sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init),
	  (sdp_key_init), (sdp_attribute_init), (sdp_message_init),
	  (sdp_message_uninit), (sdp_message_free), (sdp_media_init),
	  (sdp_media_uninit), (sdp_media_free), (sdp_message_add_media),
	  (sdp_parse_line):
	  * gst/rtsp/sdpmessage.h:
	  Based on patch by: jp.liu <jp_liu at astrocom dot cn>
	  Fix memory management of SDP messages. Fixes #407793.

2007-02-14 12:07:01 +0000  zhangfei gao <gaozhangfei@yahoo.com.cn>

	  gst/avi/gstavimux.c: Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
	  Original commit message from CVS:
	  Patch by: zhangfei gao <gaozhangfei@yahoo.com.cn>
	  * gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
	  Allow muxing video/x-h264 (was already in the caps). Fixes #407780.

2007-02-14 10:09:12 +0000  jp.liu <jp_liu@astrocom.cn>

	  gst/rtsp/rtspurl.c: Fix parsing of password field in url. Fixes #407797.
	  Original commit message from CVS:
	  Patch by: jp.liu <jp_liu at astrocom dot cn>
	  * gst/rtsp/rtspurl.c: (rtsp_url_parse):
	  Fix parsing of password field in url. Fixes #407797.

2007-02-14 09:55:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.*: Update docs.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
	  (gst_wavparse_reset), (gst_wavparse_init),
	  (gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
	  (gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
	  (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
	  (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
	  (gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
	  (gst_wavparse_loop), (gst_wavparse_chain),
	  (gst_wavparse_pad_convert), (gst_wavparse_pad_query),
	  (gst_wavparse_srcpad_event), (gst_wavparse_change_state),
	  (plugin_init):
	  * gst/wavparse/gstwavparse.h:
	  Update docs.
	  Use boilerplate.
	  Various code cleanups.
	  When the bitrate is not known (bps == 0 or compressed formats) let
	  downstream element guestimate the duration and position and don't
	  generate timestamps or durations. Fixes #405213.
	  Fix EOS and ERROR conditions in chain mode, we just need to forward the
	  error flowreturn upstream.

2007-02-13 16:01:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child that implements the GConf key monitoring. ...
	  Original commit message from CVS:
	  * ext/gconf/Makefile.am:
	  * ext/gconf/gconf.c: (gst_gconf_get_string),
	  (gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
	  (gst_gconf_render_bin_with_default):
	  * ext/gconf/gconf.h:
	  * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
	  (gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
	  (gst_gconf_audio_sink_dispose), (do_change_child),
	  (gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
	  (cb_change_child), (gst_gconf_audio_sink_change_state):
	  * ext/gconf/gstgconfaudiosink.h:
	  * ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
	  (gst_switch_sink_class_init), (gst_switch_sink_reset),
	  (gst_switch_sink_init), (gst_switch_sink_dispose),
	  (gst_switch_commit_new_kid), (gst_switch_sink_set_child),
	  (gst_switch_sink_set_property), (gst_switch_sink_handle_event),
	  (gst_switch_sink_get_property), (gst_switch_sink_change_state):
	  * ext/gconf/gstswitchsink.h:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
	  (gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
	  (gst_auto_audio_sink_detect):
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
	  (gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
	  (gst_auto_video_sink_detect):
	  Re-factor the gconfaudiosink into a "GstSwitchSink" base class
	  and a child that implements the GConf key monitoring. The end goal of
	  this is an audio sink that can be changed on the fly, but at the
	  moment it still only changes on the next READY transition.

2007-02-13 11:57:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	  (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	  (gst_avi_demux_sync), (gst_avi_demux_massage_index),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	  (gst_avi_demux_loop):
	  Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif

2007-02-13 09:46:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add crossreferences to glib/gobject/gstream docs.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  Add crossreferences to glib/gobject/gstream docs.

2007-02-12 23:35:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/monoscope/: Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS (but no LIBS, since we only use define...
	  Original commit message from CVS:
	  * gst/monoscope/Makefile.am:
	  * gst/monoscope/gstmonoscope.c:
	  Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS
	  (but no LIBS, since we only use defines from the headers).

2007-02-12 23:27:31 +0000  Jonathan Matthew <jonathan@kaolin.wh9.net>

	  gst/wavparse/gstwavparse.c: Fix massive memory leak when operating in streaming mode due to
	  Original commit message from CVS:
	  Based on patch by: Jonathan Matthew  <jonathan at kaolin wh9 net>
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
	  (gst_wavparse_stream_data):
	  Fix massive memory leak when operating in streaming mode due to
	  GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
	  Fixes #407057.

2007-02-12 15:29:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.*: Save some memory (8%) by repacking the index entry structure (more to come). Add more FIXMEs t...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	  (gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
	  (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
	  (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	  (gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
	  (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
	  (gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
	  (gst_avi_demux_stream_data), (gst_avi_demux_loop):
	  * gst/avi/gstavidemux.h:
	  Save some memory (8%) by repacking the index entry structure (more to
	  come). Add more FIXMEs to questionable parts.

2007-02-12 12:57:22 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/: More FIXME comments and messaging changes.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps),
	  (gst_v4l2src_get_caps):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	  (gst_v4l2src_capture_init):
	  More FIXME comments and messaging changes.

2007-02-12 12:43:00 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/goom/gstgoom.*: Improved docs and use GST_DEBUG_FUNCPTR.
	  Original commit message from CVS:
	  * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
	  (gst_goom_change_state):
	  * gst/goom/gstgoom.h:
	  Improved docs and use GST_DEBUG_FUNCPTR.
	  * gst/level/gstlevel.c: (gst_level_class_init):
	  Use GST_DEBUG_FUNCPTR.
	  * gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
	  (gst_monoscope_chain), (gst_monoscope_change_state):
	  Improved docs source cleanups.

2007-02-12 10:29:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/debug/: Add code for a pushfilesrc element that implements a pushfile:// URI handler, to make debugging push-mode...
	  Original commit message from CVS:
	  * gst/debug/Makefile.am:
	  * gst/debug/gstdebug.c: (plugin_init):
	  * gst/debug/gstpushfilesrc.c:
	  * gst/debug/gstpushfilesrc.h:
	  Add code for a pushfilesrc element that implements a pushfile:// URI
	  handler, to make debugging push-mode operation of demuxer/decoders
	  that support both easier in connection with seek/playbin/etc.
	  The element isn't registered at the moment.

2007-02-11 15:26:49 +0000  Sébastien Moutte <sebastien@moutte.net>

	  Makefile.am: Add win32 MANIFEST
	  Original commit message from CVS:
	  * Makefile.am:
	  Add win32 MANIFEST
	  * sys/directdraw/gstdirectdrawsink.c:
	  * sys/directdraw/gstdirectdrawsink.h:
	  Clear unused code and add comments.
	  Remove yuv from template caps, it only supports RGB
	  actually.
	  Implement XOverlay interface and remove window and fullscreen
	  properties.
	  Add debug logs.
	  Test for blit capabilities to return only the current colorspace if
	  the hardware can't blit for one colorspace to another.
	  * sys/directsound/gstdirectsoundsink.c:
	  Add some debugs.
	  * win32/MANIFEST:
	  Add VS7 project files and solution.
	  * win32/vs6/gst_plugins_bad.dsw:
	  * win32/vs6/libgstdirectdraw.dsp:
	  * win32/vs6/libgstdirectsound.dsp:
	  * win32/vs6/libgstqtdemux.dsp:
	  Update project files.

2007-02-11 12:57:47 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  Comment a #if 0 in caps template definition as VS6 seems to
	  do not support it.
	  * gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
	  Use gst_guint64_to_gdouble for conversion.
	  * gst/rtsp/rtspconnection.c:(rtsp_connection_send):
	  Move variables declaration before the first instruction.
	  * gst/rtsp/rtspdefs.c:(rtsp_strresult):
	  Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
	  And don't include netdb.h for G_OS_WIN32
	  * gst/rtsp/sdpmessage.c:(sdp_parse_line):
	  This initialization SDPMedia nmedia = {.media = NULL }; is not supported
	  by VS6 then use an other way to initialize SDPMedia structure.
	  * gst/udp/gstdynudpsink.h:
	  * gst/udp/gstdynudpnetutils.h:
	  Do not include <sys/time.h> for G_OS_WIN32
	  * gst/udp/gstudpsrc.c:
	  Define socklen_t as int for G_OS_WIN32
	  * win/common/config.h.in:
	  Undef HAVE_NETINET_IN_H
	  * win32/vs6/gst_plugins_good.dsw:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgstrtsp.dsp:
	  * win32/vs6/libgstautogen.dsp:
	  * win32/vs6/libgstaudiofx.dsp:
	  * win32/vs6/libgstudp.dsp:
	  Add and update project files.
	  * win32/common/gstudp-enumtypes.c:
	  * win32/common/gstudp-enumtypes.h:
	  Add a copy of udp enumtypes to win32/common as in core
	  and base.

2007-02-11 10:53:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Activate monoscope when building with --enable-experimental. Fix
	  Original commit message from CVS:
	  * configure.ac:
	  Activate monoscope when building with --enable-experimental. Fix
	  --enable-external configure switch description.
	  * sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init):
	  * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose):
	  Help gst-indent.

2007-02-09 16:24:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.*: On receiving EOS, we try to push a last buffer with the remaining samples. Don't do that if we go...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain),
	  (gst_lame_change_state):
	  * ext/lame/gstlame.h:
	  On receiving EOS, we try to push a last buffer with the remaining
	  samples. Don't do that if we got an unclean flow return on the last
	  gst_pad_push(), downstream might not handle this very gracefully
	  (see #403168).
	  * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
	  Pass flow returns upstream (helps #403168).

2007-02-09 09:24:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavimux.c: Explicitly cast result of pointer arithmetic to integer in order to avoid compiler warnings on s...
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
	  Explicitly cast result of pointer arithmetic to integer in order to
	  avoid compiler warnings on some 64-bit systems. Should fix #406018.

2007-02-08 11:09:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/debug/progressreport.c: Some more docs.
	  Original commit message from CVS:
	  * gst/debug/progressreport.c:
	  Some more docs.

2007-02-07 21:09:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/inspect/plugin-rtp.xml: Update for new elements.
	  Original commit message from CVS:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  Update for new elements.
	  * gst/debug/progressreport.h:
	  Commit newly-created header file as well.

2007-02-07 20:39:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Make progressreport element post messages with the current progress on the bus. Also add some basic docs for it.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * gst/debug/Makefile.am:
	  * gst/debug/progressreport.c: (gst_progress_report_post_progress),
	  (gst_progress_report_do_query), (gst_progress_report_report):
	  Make progressreport element post messages with the current progress
	  on the bus. Also add some basic docs for it.

2007-02-07 13:08:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/hal/hal.*: Some small cleanups; deal with errors when parsing the HAL ALSA capabilities a bit better.
	  Original commit message from CVS:
	  * ext/hal/hal.c: (gst_hal_get_string):
	  * ext/hal/hal.h:
	  Some small cleanups; deal with errors when parsing the HAL ALSA
	  capabilities a bit better.

2007-02-06 16:29:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/smpte/gstsmpte.c: Let's try this again and use the right cast this time.
	  Original commit message from CVS:
	  * gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
	  Let's try this again and use the right cast this time.

2007-02-06 16:24:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/smpte/gstsmpte.c: Add cast to avoid compiler warnings with older GLib versions where the nick/name members in GEn...
	  Original commit message from CVS:
	  * gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
	  Add cast to avoid compiler warnings with older GLib versions
	  where the nick/name members in GEnumValue are not declared as
	  constant strings.

2007-02-06 15:56:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gconf/: In gconfaudiosink, get the right key as the old key in do_toggle (ie. one dependent on the profile select...
	  Original commit message from CVS:
	  * ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
	  (gst_gconf_render_bin_from_key),
	  (gst_gconf_get_default_audio_sink):
	  * ext/gconf/gconf.h:
	  * ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
	  (do_toggle_element), (gst_gconf_audio_sink_set_property),
	  (gst_gconf_audio_sink_get_property):
	  In gconfaudiosink, get the right key as the old key in do_toggle
	  (ie. one dependent on the profile selected). Log some more stuff so
	  we can see what's actually going on.

2007-02-06 11:16:49 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Some small cleanups and port both elements to the new GstAudioFilter base class to save a few lines of ...
	  Original commit message from CVS:
	  * gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init),
	  (gst_audio_amplify_class_init), (gst_audio_amplify_init),
	  (gst_audio_amplify_set_process_function),
	  (gst_audio_amplify_setup):
	  * gst/audiofx/audioamplify.h:
	  * gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
	  (gst_audio_invert_class_init), (gst_audio_invert_setup):
	  * gst/audiofx/audioinvert.h:
	  Some small cleanups and port both elements to the new GstAudioFilter
	  base class to save a few lines of common code.
	  * gst/audiofx/Makefile.am:
	  Link against libgstaudio for the above changes

2007-02-03 23:35:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Fix up to use the newly ported (actually working) GstAudioFilter.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/equalizer/Makefile.am:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init),
	  (gst_iir_equalizer_class_init), (gst_iir_equalizer_init),
	  (setup_filter), (gst_iir_equalizer_compute_frequencies),
	  (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
	  (gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup),
	  (plugin_init):
	  * gst/equalizer/gstiirequalizer.h:
	  Fix up to use the newly ported (actually working) GstAudioFilter.
	  Bump core/base requirements to CVS for this.
	  * tests/icles/.cvsignore:
	  * tests/icles/Makefile.am:
	  * tests/icles/equalizer-test.c: (check_bus),
	  (equalizer_set_band_value), (equalizer_set_all_band_values),
	  (equalizer_set_band_value_and_wait),
	  (equalizer_set_all_band_values_and_wait), (do_slider_fiddling),
	  (main):
	  Add brain-dead interactive test for equalizer.

2007-02-02 18:36:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/equalizer/gstiirequalizer.c: Rename "values" property to "band-values" and change type into a
	  Original commit message from CVS:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_class_init),
	  (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
	  (gst_iir_equalizer_filter_inplace):
	  Rename "values" property to "band-values" and change type into a
	  GValueArray, so it's more easily bindable and the range of the
	  values passed in is defined and checked etc.; also do some
	  locking.

2007-02-02 17:39:21 +0000  James Doc Livingston <doclivingston@gmail.com>

	  Port equalizer plugin to 0.10 (#403572).
	  Original commit message from CVS:
	  Patch by: James "Doc" Livingston  <doclivingston at gmail com>
	  * configure.ac:
	  * gst/equalizer/Makefile.am:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
	  (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
	  (gst_iir_equalizer_compute_frequencies),
	  (gst_iir_equalizer_set_property),
	  (gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
	  (plugin_init):
	  Port equalizer plugin to 0.10 (#403572).

2007-01-31 08:32:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Fix a off by one that leads to the duration reported as one sample less than it is
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query),
	  (gst_wavpack_parse_handle_seek_event),
	  (gst_wavpack_parse_create_src_pad):
	  Fix a off by one that leads to the duration reported as one
	  sample less than it is

2007-01-30 17:19:33 +0000  Edward Hervey <bilboed@bilboed.com>

	  configure.ac: Check for an Objective C compiler
	  Original commit message from CVS:
	  * configure.ac:
	  Check for an Objective C compiler
	  * sys/Makefile.am:
	  * sys/osxvideo/Makefile.am:
	  * sys/osxvideo/cocoawindow.h:
	  * sys/osxvideo/cocoawindow.m:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  Port of osxvideo plugin to 0.10. Do NOT consider 100% stable !
	  Fixes #402470

2007-01-29 10:59:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/elements/.cvsignore: Some more ignores.
	  Original commit message from CVS:
	  * tests/check/elements/.cvsignore:
	  Some more ignores.

2007-01-28 18:28:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videocrop/gstvideocrop.c: Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY.
	  Original commit message from CVS:
	  * gst/videocrop/gstvideocrop.c:
	  (gst_video_crop_get_image_details_from_caps),
	  (gst_video_crop_transform_packed_complex):
	  Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY.
	  * tests/icles/videocrop-test.c: (check_bus_for_errors),
	  (test_with_caps), (main):
	  Block streaming thread before changing filter caps while the
	  pipeline is running so that we don't get random not-negotiated
	  errors just because GStreamer can't handle that yet.

2007-01-27 16:08:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/videocrop-test.c: Catch errors while the test is running.
	  Original commit message from CVS:
	  * tests/icles/videocrop-test.c: (test_with_caps):
	  Catch errors while the test is running.

2007-01-26 12:21:41 +0000  charles <charlesg3@gmail.com>

	  ext/shout2/gstshout2.*: Properly handle tags in shout2send. Fixes #399825.
	  Original commit message from CVS:
	  Patch by: charles <charlesg3 at gmail dot com>
	  * ext/shout2/gstshout2.c: (gst_shout2send_init),
	  (set_shout_metadata), (gst_shout2send_event):
	  * ext/shout2/gstshout2.h:
	  Properly handle tags in shout2send. Fixes #399825.

2007-01-25 23:27:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Fix the SEEKING query. We can seek if we are in pull mode, not the other way around. A...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query):
	  Fix the SEEKING query. We can seek if we are in pull mode, not the
	  other way around. Also set the correct format in the seeking query and
	  handle the case where the headers are not read yet and we can't say
	  anything about our seeking capabilities.

2007-01-25 21:55:49 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: Fix spelling in 2 places: It's called Wavpack, not WavePack.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
	  Fix spelling in 2 places: It's called Wavpack, not WavePack.

2007-01-25 14:40:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the rules in the SDP to caps document.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_activate_streams):
	  Convert SDP fields to upper/lowercase following the rules in the SDP to
	  caps document.

2007-01-25 14:22:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Fix case of encoding-name and key/value pairs to match the document.
	  Original commit message from CVS:
	  * gst/rtp/README:
	  * gst/rtp/gstrtpilbcdepay.c:
	  * gst/rtp/gstrtpilbcpay.c:
	  * gst/rtp/gstrtpmp4gdepay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpspeexdepay.c:
	  * gst/rtp/gstrtpspeexpay.c:
	  * gst/rtp/gstrtpsv3vdepay.c:
	  * gst/rtp/gstrtptheoradepay.c:
	  * gst/rtp/gstrtptheorapay.c:
	  * gst/rtp/gstrtpvorbisdepay.c:
	  * gst/rtp/gstrtpvorbispay.c:
	  Fix case of encoding-name and key/value pairs to match the document.
	  This is to make interoperation with SDP case-insensitive as required by
	  the relevant RFCs.

2007-01-25 12:05:11 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/: Use proper print statements.
	  Original commit message from CVS:
	  * gst/multifile/gstmultifilesink.c:
	  (gst_multi_file_sink_class_init):
	  * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init):
	  * gst/mve/gstmvedemux.c: (gst_mve_video_create_buffer),
	  (gst_mve_video_palette), (gst_mve_video_code_map),
	  (gst_mve_audio_init), (gst_mve_audio_data), (gst_mve_timer_create),
	  (gst_mve_demux_chain):
	  * gst/mve/gstmvemux.c: (gst_mve_mux_push_chunk):
	  * gst/mve/mveaudioenc.c: (mve_compress_audio):
	  * gst/mve/mvevideodec16.c: (ipvideo_copy_block):
	  * gst/mve/mvevideodec8.c: (ipvideo_copy_block):
	  * gst/mve/mvevideoenc16.c: (mve_encode_frame16):
	  * gst/mve/mvevideoenc8.c: (mve_encode_frame8):
	  Use proper print statements.
	  Fixes build on mac os x.
	  <wingo> oo look at me my name is edward i'm hacking on macos wooo

2007-01-25 11:02:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  configure.ac: Bump required -core/-base to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Bump required -core/-base to CVS

2007-01-25 10:54:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpL16pay.*: Fill up to MTU using adapter.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
	  (gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
	  * gst/rtp/gstrtpL16pay.h:
	  Fill up to MTU using adapter.
	  Timestamp rtp packets.

2007-01-25 10:36:35 +0000  Edward Hervey <bilboed@bilboed.com>

	  Use G_GSIZE_FORMAT in print statements for portability.
	  Original commit message from CVS:
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	  * sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls):
	  Use G_GSIZE_FORMAT in print statements for portability.
	  Fixes build on macosx.

2007-01-24 18:20:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Port and enable raw audio payloader/depayloader. Needs a bit more work on the payloader side.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
	  (gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
	  (gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
	  (gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
	  (gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
	  (gst_rtp_L16_depay_plugin_init):
	  * gst/rtp/gstrtpL16depay.h:
	  * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
	  (gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
	  (gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
	  (gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
	  (gst_rtp_L16_pay_plugin_init):
	  * gst/rtp/gstrtpL16pay.h:
	  Port and enable raw audio payloader/depayloader. Needs a bit more work
	  on the payloader side.

2007-01-24 16:25:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Only unblock the udp pads when we linked and activated them all.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (pad_blocked),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp):
	  * gst/rtsp/gstrtspsrc.h:
	  Only unblock the udp pads when we linked and activated them all.
	  Fixes #395688.

2007-01-24 15:18:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added simple AC3 depayloader (RFC 4184).
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init),
	  (gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init),
	  (gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process),
	  (gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property),
	  (gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init):
	  * gst/rtp/gstrtpac3depay.h:
	  Added simple AC3 depayloader (RFC 4184).
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	  Fix a leak.

2007-01-24 12:41:03 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" eleme...
	  Original commit message from CVS:
	  reviewed by: Stefan Kost  <ensonic@users.sf.net>
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audioamplify.c:
	  (gst_audio_amplify_clipping_method_get_type),
	  (gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
	  (gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
	  (gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
	  (gst_audio_amplify_set_caps),
	  (gst_audio_amplify_transform_int_clip),
	  (gst_audio_amplify_transform_int_wrap_negative),
	  (gst_audio_amplify_transform_int_wrap_positive),
	  (gst_audio_amplify_transform_float_clip),
	  (gst_audio_amplify_transform_float_wrap_negative),
	  (gst_audio_amplify_transform_float_wrap_positive),
	  (gst_audio_amplify_transform_ip):
	  * gst/audiofx/audioamplify.h:
	  * gst/audiofx/audiofx.c: (plugin_init):
	  Add new element "audioamplify". This allows scaling of raw audio
	  samples, similar to the "volume" element, but provides different modes
	  for clipping and allows unlimited amplification. It's mainly targeted
	  for creative sound design and not as a replacement of the "volume"
	  element. Fixes #397162
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  Add docs for audioamplify and integrate them into the build system
	  * tests/check/Makefile.am:
	  * tests/check/elements/audioamplify.c: (setup_amplify),
	  (cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
	  Add fairly extensive unit test suite for audioamplify

2007-01-24 12:26:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Unblock pads after adding the pads to the element so that autopluggers get a change to link so...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
	  Unblock pads after adding the pads to the element so that autopluggers
	  get a change to link something. Possibly fixes #395688.

2007-01-24 12:22:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Fix caps with payload numbers.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrdepay.c:
	  * gst/rtp/gstrtpgsmdepay.c:
	  * gst/rtp/gstrtph263pdepay.c:
	  * gst/rtp/gstrtph263ppay.c:
	  * gst/rtp/gstrtph264depay.c:
	  * gst/rtp/gstrtpilbcdepay.c:
	  * gst/rtp/gstrtpmp2tdepay.c:
	  * gst/rtp/gstrtpmp4gdepay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	  * gst/rtp/gstrtpmp4vpay.c:
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init),
	  (gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps),
	  (gst_rtp_mpa_depay_process):
	  * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init),
	  (gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process):
	  * gst/rtp/gstrtppcmadepay.c:
	  * gst/rtp/gstrtppcmudepay.c:
	  * gst/rtp/gstrtpspeexdepay.c:
	  * gst/rtp/gstrtpspeexpay.c:
	  * gst/rtp/gstrtpsv3vdepay.c:
	  * gst/rtp/gstrtptheoradepay.c:
	  * gst/rtp/gstrtptheorapay.c:
	  * gst/rtp/gstrtpvorbisdepay.c:
	  * gst/rtp/gstrtpvorbispay.c:
	  Fix caps with payload numbers.
	  Add some fixed payload numbers to caps when possible.

2007-01-24 11:29:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/gstrtpxqtdepay.c: Fix caps on the depayloader.
	  Original commit message from CVS:
	  * gst/qtdemux/gstrtpxqtdepay.c:
	  Fix caps on the depayloader.

2007-01-23 18:16:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/: Add new audiofx element "audioinvert". This element swaps the upper and lower half of samples and can b...
	  Original commit message from CVS:
	  reviewed by: Stefan Kost  <ensonic@users.sf.net>
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofx.c: (plugin_init):
	  * gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
	  (gst_audio_invert_class_init), (gst_audio_invert_init),
	  (gst_audio_invert_set_property), (gst_audio_invert_get_property),
	  (gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
	  (gst_audio_invert_transform_float),
	  (gst_audio_invert_transform_ip):
	  * gst/audiofx/audioinvert.h:
	  Add new audiofx element "audioinvert". This element swaps the upper
	  and lower half of samples and can be used for example for a
	  wide-stereo effect. Fixes #396057
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  Add docs for the audioinvert element and add them to the build system.
	  * tests/check/Makefile.am:
	  * tests/check/elements/audioinvert.c: (setup_invert),
	  (cleanup_invert), (GST_START_TEST), (invert_suite), (main):
	  Add unit test suite for the audioinvert element.

2007-01-23 17:36:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4gdepay.c: Parse config params as string and int.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int),
	  (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process):
	  Parse config params as string and int.
	  Parse and use AU header length

2007-01-23 17:27:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/smpte/: constify some static structs.
	  Original commit message from CVS:
	  * gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw),
	  (gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw):
	  * gst/smpte/gstmask.c: (_gst_mask_register):
	  * gst/smpte/gstmask.h:
	  * gst/smpte/gstsmpte.c: (gst_smpte_update_mask):
	  * gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line),
	  (gst_smpte_paint_triangle_clock):
	  constify some static structs.
	  Don't update the mask if nothing changed to the params.
	  Make sure we never draw outside of the picture. Fixes #398325.

2007-01-22 13:06:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Error out properly when pull_range fails while we're reading the headers, instead of just paus...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull):
	  Error out properly when pull_range fails while we're reading the
	  headers, instead of just pausing the task silently. Fixes #399338.

2007-01-19 13:06:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/smpte/gstsmpte.c: Some more sanity checks to make sure the input formats match and the input pads are actually ne...
	  Original commit message from CVS:
	  * gst/smpte/gstsmpte.c: (gst_smpte_collected):
	  Some more sanity checks to make sure the input formats match and the
	  input pads are actually negotiated, in case someone tries to feed
	  buffers from fakesrc or filesrc. Fixes #398299.
	  Also const-ify an array, just because we can.

2007-01-19 10:35:13 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/smpte/gstsmpte.c: Ignore previous commit, that was only valid for widths and heights that are multiples of 4.
	  Original commit message from CVS:
	  * gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected):
	  Ignore previous commit, that was only valid for widths and heights
	  that are multiples of 4.
	  Copy over size/stride macros from jpegdec. This allows the element
	  to work with any width,height...
	  ... but puts in evidence that the actual transformations only work
	  with width/height that are multiples of 4.

2007-01-19 09:48:47 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/smpte/gstsmpte.c: Allocate buffers of the right size.
	  Original commit message from CVS:
	  * gst/smpte/gstsmpte.c: (gst_smpte_collected):
	  Allocate buffers of the right size.
	  The proper size of a I420 buffer in bytes is:
	  width * height * 3
	  ------------------
	  2

2007-01-18 18:37:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/smpte/gstsmpte.c: Proxy getcaps on sink pads too, so that we either end up with the same dimensions on all pads o...
	  Original commit message from CVS:
	  * gst/smpte/gstsmpte.c: (gst_smpte_init):
	  Proxy getcaps on sink pads too, so that we either end up with the
	  same dimensions on all pads or error out if that's not possible
	  (seems to work even!). Fixes #398086, I think.

2007-01-18 11:29:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Remove ladspa from docs; add hierarchy info for GstAudioPanorama; fix integer properties with -1 as mi...
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  Remove ladspa from docs; add hierarchy info for GstAudioPanorama;
	  fix integer properties with -1 as minimum value.
	  * docs/plugins/inspect/plugin-1394.xml:
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-shout2send.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update to CVS.

2007-01-18 11:23:36 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiofx/audiopanorama.c: Fix doc section name (Fixes #397946)
	  Original commit message from CVS:
	  * gst/audiofx/audiopanorama.c:
	  Fix doc section name (Fixes #397946)

2007-01-18 10:33:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  Remove bogus ChangeLog entry
	  Original commit message from CVS:
	  Remove bogus ChangeLog entry

2007-01-17 14:30:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/: Fix EIO handing when capturing. Add new property to specify the number of buffers to enque (and remove the...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2object.c:
	  (gst_v4l2_object_install_properties_helper),
	  (gst_v4l2_object_set_property_helper),
	  (gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	  (gst_v4l2src_init), (gst_v4l2src_set_property),
	  (gst_v4l2src_get_property), (gst_v4l2src_set_caps):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	  (gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
	  (gst_v4l2src_capture_deinit):
	  Fix EIO handing when capturing. Add new property to specify the number of
	  buffers to enque (and remove the borked num-buffers usage).

2007-01-16 08:29:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/audiopanorama.c: Use a function array for process methods, add more docs and define the startindex of enums.
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo circular-chaos org>
	  * gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init),
	  (gst_audio_panorama_set_process_function):
	  Use a function array for process methods, add more docs and define the
	  startindex of enums.

2007-01-14 17:55:33 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  Add support for more than one audio stream; write better AVIX header; refactor code a bit; don't announce vorbis caps...
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw at skynet be>
	  * gst/avi/gstavimux.c: (gst_avi_mux_finalize),
	  (gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
	  (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
	  (gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
	  (gst_avi_mux_riff_get_avi_header),
	  (gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
	  (gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
	  (gst_avi_mux_bigfile), (gst_avi_mux_start_file),
	  (gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
	  (gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
	  (gst_avi_mux_change_state):
	  * gst/avi/gstavimux.h:
	  * tests/check/elements/avimux.c: (teardown_src_pad):
	  Add support for more than one audio stream; write better AVIX
	  header; refactor code a bit; don't announce vorbis caps on our audio
	  sink pads since we don't support it anyway. Closes #379298.

2007-01-13 19:12:32 +0000  Andy Wingo <wingo@pobox.com>

	  gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads): Use fixed caps on src pads.
	  Original commit message from CVS:
	  2007-01-13  Andy Wingo  <wingo@pobox.com>
	  * gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads):
	  Use fixed caps on src pads.
	  (gst_deinterleave_remove_pads): Remove src pads, not sink pads. I
	  seem to have reverse midas disease!
	  (gst_deinterleave_process): Proxy timestamps, offsets, durations,
	  and set caps on outgoing buffers. Fixes #395597, I think.

2007-01-13 18:01:41 +0000  Andy Wingo <wingo@pobox.com>

	  gst/interleave/interleave.c (gst_interleave_init): Init the activation mode properly.
	  Original commit message from CVS:
	  2007-01-13  Andy Wingo  <wingo@pobox.com>
	  * gst/interleave/interleave.c (gst_interleave_init): Init the
	  activation mode properly.
	  (gst_interleave_src_setcaps, gst_interleave_src_getcaps)
	  (gst_interleave_init): Set a setcaps and getcaps function on the
	  src pad, so that we can implement pull-mode negotiation.
	  (gst_interleave_sink_setcaps): Renamed from
	  gst_interleave_setcaps, as it only does the sink logic now.
	  Implement both for pull-mode and push-mode.
	  (gst_interleave_process): Set caps on our outgoing buffer.
	  (gst_interleave_src_activate_pull): Fix some more bogus casts.
	  What is up with this.

2007-01-13 15:52:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiofx/audiopanorama.*: Add 'method' property and provide a simple (non-psychoacustic) processing method (#394859).
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo circular-chaos org>
	  * gst/audiofx/audiopanorama.c:
	  (gst_audio_panorama_method_get_type),
	  (gst_audio_panorama_class_init), (gst_audio_panorama_init),
	  (gst_audio_panorama_set_process_function),
	  (gst_audio_panorama_set_property),
	  (gst_audio_panorama_get_property), (gst_audio_panorama_set_caps),
	  (gst_audio_panorama_transform_m2s_int_simple),
	  (gst_audio_panorama_transform_s2s_int_simple),
	  (gst_audio_panorama_transform_m2s_float_simple),
	  (gst_audio_panorama_transform_s2s_float_simple):
	  * gst/audiofx/audiopanorama.h:
	  Add 'method' property and provide a simple (non-psychoacustic)
	  processing method (#394859).
	  * tests/check/elements/audiopanorama.c: (GST_START_TEST),
	  (panorama_suite):
	  Tests for new method.

2007-01-12 18:28:13 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  comment out LADSPA plugin for now
	  Original commit message from CVS:
	  comment out LADSPA plugin for now

2007-01-12 17:16:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/: Add X-QT depayloader that will eventually share code with the demuxer.
	  Original commit message from CVS:
	  * gst/qtdemux/Makefile.am:
	  * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_base_init),
	  (gst_rtp_xqt_depay_class_init), (gst_rtp_xqt_depay_init),
	  (gst_rtp_xqt_depay_finalize), (gst_rtp_quicktime_parse_sd),
	  (gst_rtp_xqt_depay_setcaps), (gst_rtp_xqt_depay_process),
	  (gst_rtp_xqt_depay_set_property), (gst_rtp_xqt_depay_get_property),
	  (gst_rtp_xqt_depay_change_state), (gst_rtp_xqt_depay_plugin_init):
	  * gst/qtdemux/gstrtpxqtdepay.h:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_base_init),
	  (gst_qtdemux_loop_state_header), (gst_qtdemux_loop),
	  (qtdemux_parse_moov), (qtdemux_parse_container),
	  (qtdemux_parse_node), (gst_qtdemux_add_stream),
	  (qtdemux_parse_trak), (qtdemux_audio_caps):
	  * gst/qtdemux/qtdemux.h:
	  * gst/qtdemux/quicktime.c: (plugin_init):
	  Add X-QT depayloader that will eventually share code with the demuxer.
	  Make new plugin entry point with quicktime releated stuff.

2007-01-12 12:10:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/Makefile.am: Dist all new files.
	  Original commit message from CVS:
	  * gst/qtdemux/Makefile.am:
	  Dist all new files.

2007-01-12 10:27:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Activate docs for jack, sdl and qtdemux.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-qtdemux.xml:
	  Activate docs for jack, sdl and qtdemux.

2007-01-12 10:22:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/: Cleanup and refactor to make the code more readable.
	  Original commit message from CVS:
	  * gst/qtdemux/Makefile.am:
	  * gst/qtdemux/qtdemux.c: (extract_initial_length_and_fourcc),
	  (gst_qtdemux_loop_state_header), (gst_qtdemux_combine_flows),
	  (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop),
	  (gst_qtdemux_chain), (qtdemux_sink_activate_pull),
	  (qtdemux_inflate), (qtdemux_parse_moov), (qtdemux_parse_container),
	  (qtdemux_parse_node), (qtdemux_tree_get_child_by_type),
	  (qtdemux_tree_get_sibling_by_type), (gst_qtdemux_add_stream),
	  (qtdemux_parse_samples), (qtdemux_parse_segments),
	  (qtdemux_parse_trak), (qtdemux_tag_add_str), (qtdemux_tag_add_num),
	  (qtdemux_tag_add_date), (qtdemux_tag_add_gnre),
	  (qtdemux_parse_udta), (qtdemux_redirects_sort_func),
	  (qtdemux_process_redirects), (qtdemux_parse_redirects),
	  (qtdemux_parse_tree), (gst_qtdemux_handle_esds),
	  (qtdemux_video_caps), (qtdemux_audio_caps):
	  * gst/qtdemux/qtdemux.h:
	  * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mvhd),
	  (qtdemux_dump_tkhd), (qtdemux_dump_elst), (qtdemux_dump_mdhd),
	  (qtdemux_dump_hdlr), (qtdemux_dump_vmhd), (qtdemux_dump_dref),
	  (qtdemux_dump_stsd), (qtdemux_dump_stts), (qtdemux_dump_stss),
	  (qtdemux_dump_stsc), (qtdemux_dump_stsz), (qtdemux_dump_stco),
	  (qtdemux_dump_co64), (qtdemux_dump_dcom), (qtdemux_dump_cmvd),
	  (qtdemux_dump_unknown), (qtdemux_node_dump_foreach),
	  (qtdemux_node_dump):
	  * gst/qtdemux/qtdemux_dump.h:
	  * gst/qtdemux/qtdemux_fourcc.h:
	  * gst/qtdemux/qtdemux_types.c: (qtdemux_type_get):
	  * gst/qtdemux/qtdemux_types.h:
	  * gst/qtdemux/qtpalette.h:
	  Cleanup and refactor to make the code more readable.
	  Move debugging/tables into separate files.
	  Add 2/4/16 color palletee support.
	  Fix raw 15 bit RGB handling.
	  Use more FOURCC constants.
	  Add some docs.

2007-01-11 19:51:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.c: Minor clean-up: use enum values instead of hardcoded constants (#395536).
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge  <slomo@circular-chaos.org>
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type),
	  (gst_wavpack_enc_correction_mode_get_type),
	  (gst_wavpack_enc_joint_stereo_mode_get_type):
	  Minor clean-up: use enum values instead of hardcoded constants (#395536).

2007-01-11 16:59:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Set correct caps on outgoing pulled buffers, or things blow up after recent core changes.
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_read_range):
	  Set correct caps on outgoing pulled buffers, or things blow up
	  after recent core changes.

2007-01-11 11:05:04 +0000  Jonas Holmberg <jonas.holmberg@axis.com>

	  gst/multipart/multipartmux.c: Return FLOW errors ASAP. Fixes #394977.
	  Original commit message from CVS:
	  Based on patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_init),
	  (gst_multipart_mux_request_new_pad),
	  (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
	  (gst_multipart_mux_change_state):
	  Return FLOW errors ASAP. Fixes #394977.
	  Misc cleanups.

2007-01-11 09:30:59 +0000  Lutz Mueller <lutz@topfrose.de>

	  gst/rtsp/gstrtspsrc.c: Check for stream pad before activating.
	  Original commit message from CVS:
	  Patch by: Lutz Mueller <lutz at topfrose dot de>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
	  Check for stream pad before activating.

2007-01-10 15:19:48 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/rtsp/: Allow url to be NULL to be able to use it for server connections.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/rtsp/COPYING.MIT:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	  (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
	  (gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
	  (gst_rtspsrc_parse_methods),
	  (gst_rtspsrc_create_transports_string),
	  (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
	  (gst_rtspsrc_open), (gst_rtspsrc_close):
	  * gst/rtsp/gstrtspsrc.h:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	  (rtsp_connection_connect), (rtsp_connection_send), (read_line),
	  (parse_request_line), (parse_line), (rtsp_connection_read),
	  (rtsp_connection_close):
	  * gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
	  (rtsp_method_as_text), (rtsp_header_as_text),
	  (rtsp_status_as_text), (rtsp_find_header_field),
	  (rtsp_find_method):
	  * gst/rtsp/rtspdefs.h:
	  * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
	  (rtsp_ext_wms_configure_stream):
	  * gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
	  (rtsp_message_new_request), (rtsp_message_init_request),
	  (rtsp_message_new_response), (rtsp_message_init_response),
	  (rtsp_message_init_data), (rtsp_message_unset),
	  (rtsp_message_free), (rtsp_message_add_header),
	  (rtsp_message_get_header), (rtsp_message_set_body),
	  (rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
	  * gst/rtsp/rtspmessage.h:
	  * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
	  (sdp_media_get_attribute_val_n), (read_string), (read_string_del),
	  (sdp_parse_line), (sdp_message_parse_buffer), (print_media),
	  (sdp_message_dump):
	  Allow url to be NULL to be able to use it for server connections.
	  Can now send responses as well as requests.
	  No longer hangs in an endless loop if EOF is received.
	  Can now convert a status code to a text string.
	  Return RTSP_HDR_INVALID for unknown headers.
	  Return RTSP_INVALID for unknown methods.
	  Copy CSeq and Session headers from the request.
	  Only free memory corresponding to the currently set message type.
	  Added const to function arguments as appropriate.
	  Avoid a compiler warning when initializing nmedia.
	  Use guint rather than gint to avoid compiler warnings.
	  Fix crasher in wms extension.
	  Factor out stream setup from open_connection.
	  Delay activation of streams when actual data is received from the
	  server, this prepares us to do proper protocol switching.
	  Added new license.
	  Fixes #380895.

2007-01-10 09:47:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Some small docs fixes (#394851).
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo ubuntu com>
	  * docs/plugins/Makefile.am:
	  * gst/audiofx/audiopanorama.c:
	  Some small docs fixes (#394851).

2007-01-09 12:25:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Fix docs.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  Fix docs.

2007-01-09 12:23:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added RFC 2250 MPEG Video Depayloader.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_base_init),
	  (gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init),
	  (gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process),
	  (gst_rtp_mpv_depay_set_property), (gst_rtp_mpv_depay_get_property),
	  (gst_rtp_mpv_depay_change_state), (gst_rtp_mpv_depay_plugin_init):
	  * gst/rtp/gstrtpmpvdepay.h:
	  Added RFC 2250 MPEG Video Depayloader.
	  * gst/rtp/gstrtpL16depay.h:
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	  (gst_rtp_h263p_depay_process):
	  Fix Header file. Small cleanups.
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init),
	  (gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize),
	  (gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state):
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init),
	  (gst_rtp_mp4v_depay_init), (gst_rtp_mp4v_depay_finalize),
	  (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process),
	  (gst_rtp_mp4v_depay_change_state):
	  Remove usused code. Remove Adapter from state Change. Added debug.
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_base_init),
	  (gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init),
	  (gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process):
	  * gst/rtp/gstrtpmpadepay.h:
	  Subclass base depayloader.
	  Added debug.
	  Support static payload type assignment as well.
	  * gst/rtp/gstrtpmpapay.c:
	  Fix caps.

2007-01-08 12:45:10 +0000  Vincent Torri <vtorri@univ-evry.fr>

	  ext/jpeg/: These libjpeg callbacks should return a 'boolean' (unsigned char apparently) and not a 'gboolean' (which m...
	  Original commit message from CVS:
	  Patch by: Vincent Torri  <vtorri at univ-evry fr>
	  * ext/jpeg/gstjpegdec.c:
	  * ext/jpeg/gstjpegenc.c:
	  * ext/jpeg/smokecodec.c:
	  These libjpeg callbacks should return a 'boolean' (unsigned char
	  apparently) and not a 'gboolean' (which maps to gint). Fixes
	  warnings when compiling with MingW (#393427).
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	  Use ioctlsocket on win32.
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Some printf format fixes for win32.

2007-01-07 22:03:54 +0000  Andy Wingo <wingo@pobox.com>

	  New elements interleave and deinterleave, implement channel interleaving and deinterleaving.
	  Original commit message from CVS:
	  2007-01-07  Andy Wingo  <wingo@pobox.com>
	  * configure.ac:
	  * gst/interleave/Makefile.am:
	  * gst/interleave/plugin.h:
	  * gst/interleave/plugin.c:
	  * gst/interleave/interleave.c:
	  * gst/interleave/deinterleave.c: New elements interleave and
	  deinterleave, implement channel interleaving and deinterleaving.
	  The interleaver can operate in pull or push mode but the
	  deinterleaver is more like a demuxer and can only operate in push
	  mode.

2007-01-07 10:44:12 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/cutter/gstcutter.c: Use gst_guint64_to_gdouble for conversion.
	  Original commit message from CVS:
	  * gst/cutter/gstcutter.c: (gst_cutter_chain):
	  Use gst_guint64_to_gdouble for conversion.
	  * win32/vs6/libgstmatroska.dsp:
	  Add zlib to the link.
	  * win32/vs6/libgstvideobox.dsp:
	  Update liboil library name (project is linked to liboil-0.3-0.lib now).

2007-01-05 18:32:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Check for zlib and if available pass it explicitly to the linker when linking qtdemux. If not available (or --disable...
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/qtdemux/Makefile.am:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_moov):
	  Check for zlib and if available pass it explicitly to the linker
	  when linking qtdemux. If not available (or --disable-external has
	  been specified!), disable the bits in qtdemux that use it. Fixes
	  build on MingW (#392856).

2007-01-05 17:23:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/Makefile.am: If zlib is available and used, we must link it explicitly for things to work on MingW (fixe...
	  Original commit message from CVS:
	  * gst/matroska/Makefile.am:
	  If zlib is available and used, we must link it explicitly for
	  things to work on MingW (fixes #392855).

2007-01-05 16:07:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/videocrop-test.c: Call g_thread_init() right at the beginning. Remove superfluous gst_init() - we've alre...
	  Original commit message from CVS:
	  * tests/icles/videocrop-test.c: (main):
	  Call g_thread_init() right at the beginning. Remove superfluous
	  gst_init() - we've already been inited via the GOption stuff.

2007-01-04 11:02:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/esd/esdsink.c: Don't return bogus values when esd_get_delay() fails for some reason (#392189).
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_delay):
	  Don't return bogus values when esd_get_delay() fails for some
	  reason (#392189).

2007-01-04 09:44:57 +0000  Vincent Torri <vtorri@univ-evry.fr>

	  Add directsoundsink to build and dist it, so it gets built when compiling with MingW on win32 and the required header...
	  Original commit message from CVS:
	  Patch by: Vincent Torri  <vtorri at univ-evry fr>
	  * configure.ac:
	  * sys/Makefile.am:
	  * sys/directsound/Makefile.am:
	  * sys/directsound/gstdirectsoundsink.c:
	  (gst_directsoundsink_reset):
	  Add directsoundsink to build and dist it, so it gets built when
	  compiling with MingW on win32 and the required headers and libraries
	  are available (fixes: #392638). Also simplify DirectDraw check a bit.
	  * tests/check/elements/.cvsignore:
	  Fix CVS ignore for neonhttpsrc test binary.

2007-01-03 19:54:33 +0000  Vincent Torri <vtorri@univ-evry.fr>

	  Add directdrawsink to build and dist it, so it gets built when compiling with MingW on win32 and the required headers...
	  Original commit message from CVS:
	  Patch by: Vincent Torri  <vtorri at univ-evry fr>
	  * configure.ac:
	  * sys/Makefile.am:
	  * sys/directdraw/Makefile.am:
	  Add directdrawsink to build and dist it, so it gets built when
	  compiling with MingW on win32 and the required headers and libraries
	  are available (fixes: #392313).
	  * sys/directdraw/gstdirectdrawsink.c:
	  (gst_directdrawsink_center_rect), (gst_directdrawsink_show_frame),
	  (gst_directdrawsink_setup_ddraw),
	  (gst_directdrawsink_surface_create):
	  Comment out some unused things and fix some printf format issues in
	  order to avoid warnings when buildling with MingW (#392313).

2007-01-03 16:41:10 +0000  Jens Granseuer <jensgr@gmx.net>

	  Fix build with gcc-2.x (declare variables at the beginning of a block etc.). Fixes #391971.
	  Original commit message from CVS:
	  Patch by: Jens Granseuer  <jensgr at gmx net>
	  * ext/xvid/gstxvidenc.c: (gst_xvidenc_encode),
	  (gst_xvidenc_get_property):
	  * gst/filter/gstbpwsinc.c: (bpwsinc_transform_ip):
	  * gst/filter/gstfilter.c: (plugin_init):
	  * gst/filter/gstiir.c: (iir_transform_ip):
	  * gst/filter/gstlpwsinc.c: (lpwsinc_transform_ip):
	  * gst/modplug/gstmodplug.cc:
	  * gst/nuvdemux/gstnuvdemux.c: (gst_nuv_demux_header_load),
	  (gst_nuv_demux_stream_extend_header):
	  Fix build with gcc-2.x (declare variables at the beginning of a
	  block etc.). Fixes #391971.

2006-12-30 20:01:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/lame/gstlame.c: warn when outgoing sample rate is different from incoming
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_setcaps), (gst_lame_chain):
	  warn when outgoing sample rate is different from incoming

2006-12-30 12:44:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/videocrop.c: When we can't create an element needed for the test, print a message detailing whic...
	  Original commit message from CVS:
	  * tests/check/elements/videocrop.c: (GST_START_TEST),
	  (videocrop_test_cropping_init_context):
	  When we can't create an element needed for the test, print a message
	  detailing which element it actually is that's missing (#390673).

2006-12-24 11:36:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/ximage/gstximagesrc.c: Fix presumably copy'n'pasto for 16bpp depth.
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c: (composite_pixel):
	  Fix presumably copy'n'pasto for 16bpp depth.

2006-12-24 11:24:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-mux.c: The "signed" field in audio caps is of boolean type, trying to use gst_structure_get_int...
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_audio_pad_setcaps):
	  The "signed" field in audio caps is of boolean type, trying to use
	  gst_structure_get_int() to extract it will fail. Fixing this makes
	  matroskamux accept raw audio input (#387121) (use at your own risk
	  though, due to the matroska spec being not entirely useful in this
	  respect).
	  Also fix up raw audio structures in template caps so that they
	  represent what our setcaps function will actually accept, so that
	  converters know what to convert to.
	  Finally, don't fail if there isn't an "endianness" field in 8-bit
	  PCM caps.

2006-12-22 10:15:24 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/: reapply consistent pad (de)activation
	  Original commit message from CVS:
	  * tests/check/elements/mpeg2enc.c: (setup_mpeg2enc),
	  (cleanup_mpeg2enc):
	  * tests/check/elements/rganalysis.c: (cleanup_rganalysis):
	  * tests/check/elements/wavpackdec.c: (setup_wavpackdec),
	  (cleanup_wavpackdec):
	  * tests/check/elements/wavpackenc.c: (setup_wavpackenc),
	  (cleanup_wavpackenc):
	  * tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc):
	  reapply consistent pad (de)activation

2006-12-22 10:15:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/: reapply consistent pad (de)activation
	  Original commit message from CVS:
	  * tests/check/elements/audiopanorama.c: (cleanup_panorama):
	  * tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	  * tests/check/elements/cmmldec.c: (setup_cmmldec),
	  (teardown_cmmldec):
	  * tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	  (teardown_cmmlenc):
	  * tests/check/elements/level.c: (setup_level), (cleanup_level):
	  reapply consistent pad (de)activation

2006-12-21 17:03:39 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS
	  * gst-plugins-good.doap:
	  Add 0.10.5 doap entry

=== release 0.10.4 ===

2006-12-21 15:45:02 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: releasing 0.10.4, "Black Bugs"
	  Original commit message from CVS:
	  === release 0.10.4 ===
	  2006-12-21  Jan Schmidt <thaytan@mad.scientist.com>
	  * configure.ac:
	  releasing 0.10.4, "Black Bugs"

=== release 0.10.5 ===

2006-12-21 15:40:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: releasing 0.10.5, "The Path of Thorns"
	  Original commit message from CVS:
	  === release 0.10.5 ===
	  2006-12-21  Jan Schmidt <thaytan@mad.scientist.com>
	  * configure.ac:
	  releasing 0.10.5, "The Path of Thorns"

2006-12-21 14:03:42 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/mpeg2enc.c: (setup_mpeg2enc)
	  Original commit message from CVS:
	  * tests/check/elements/mpeg2enc.c: (setup_mpeg2enc)
	  (cleanup_mpeg2enc):
	  * tests/check/elements/rganalysis.c: (cleanup_rganalysis):
	  * tests/check/elements/wavpackdec.c: (setup_wavpackdec),
	  (cleanup_wavpackdec):
	  * tests/check/elements/wavpackenc.c: (setup_wavpackenc),
	  (cleanup_wavpackenc):
	  * tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc):
	  revert my freeze breakage

2006-12-21 12:48:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/: revert my freeze breakage
	  Original commit message from CVS:
	  * tests/check/elements/audiopanorama.c: (cleanup_panorama):
	  * tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	  * tests/check/elements/cmmldec.c: (setup_cmmldec),
	  (teardown_cmmldec):
	  * tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	  (teardown_cmmlenc):
	  * tests/check/elements/level.c: (setup_level), (cleanup_level):
	  revert my freeze breakage

2006-12-21 08:20:10 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/: consistent pad (de)activation
	  Original commit message from CVS:
	  * tests/check/elements/mpeg2enc.c: (setup_mpeg2enc),
	  (cleanup_mpeg2enc):
	  * tests/check/elements/rganalysis.c: (cleanup_rganalysis):
	  * tests/check/elements/wavpackdec.c: (setup_wavpackdec),
	  (cleanup_wavpackdec):
	  * tests/check/elements/wavpackenc.c: (setup_wavpackenc),
	  (cleanup_wavpackenc):
	  * tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc):
	  consistent pad (de)activation

2006-12-21 08:15:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/: consistent pad (de)activation
	  Original commit message from CVS:
	  * tests/check/elements/audiopanorama.c: (cleanup_panorama):
	  * tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	  * tests/check/elements/cmmldec.c: (setup_cmmldec),
	  (teardown_cmmldec):
	  * tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	  (teardown_cmmlenc):
	  * tests/check/elements/level.c: (setup_level), (cleanup_level):
	  consistent pad (de)activation

2006-12-18 17:11:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Don't post BUFFERING messages in streaming mode if the stream headers are behind the movie dat...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_post_progress),
	  (gst_qtdemux_chain):
	  Don't post BUFFERING messages in streaming mode if the stream
	  headers are behind the movie data; instead, post "progress" element
	  messages as a temporary solution. Apps might get confused and do
	  silly things to the pipeline state if they see buffering messages
	  from different sources and don't realize they come from different
	  sources (#387160).

2006-12-18 16:46:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Disable LADPSA, as it has moved to the -bad module for the duration.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/Makefile.am:
	  Disable LADPSA, as it has moved to the -bad module for the duration.

2006-12-18 15:51:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ladspa/gstsignalprocessor.c: Reset flow_state back to _OK after a flush stop so that we exit our error state afte...
	  Original commit message from CVS:
	  * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
	  (gst_signal_processor_event):
	  Reset flow_state back to _OK after a flush stop so that we exit our
	  error state after the flush. Fixes #374213

2006-12-18 15:49:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ChangeLog surgery on one of Stefan's commits from August:
	  Original commit message from CVS:
	  ChangeLog surgery on one of Stefan's commits from August:
	  * ext/Makefile.am:
	  Quietly (accidentally) enable LADSPA for building by default,
	  despite the fact that it doesn't meet the plugin checklist.
	  -- Added by Jan Schmidt 18 Dec 2006

2006-12-18 13:40:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/qtdemux/qtdemux.c: Don't output g_warning for an unsupported format, just send a
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_chain),
	  (gst_qtdemux_add_stream):
	  Don't output g_warning for an unsupported format, just send a
	  GST_ELEMENT_WARNING and don't add the pad.
	  Fix the case where it doesn't check for a NULL pad in streaming mode.
	  Fixes #387137

2006-12-18 12:27:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Fix crash dereferencing NULL pointer if there's no stco atom.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  Fix crash dereferencing NULL pointer if there's no stco atom.
	  Fixes #387122.

2006-12-18 10:02:56 +0000  Sebastian Dröge <slomo@ubuntu.com>

	  ext/wavpack/gstwavpackenc.h: Use local copy of md5.h, as it disappeared in recent wavpack installs.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.h:
	  Use local copy of md5.h, as it disappeared in recent wavpack
	  installs.
	  Patch by: Sebastian Dröge <slomo at ubuntu dot com>
	  Fixes: #387076

2006-12-17 19:42:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-12-17 06:11:39 +0000  David Schleef <ds@schleef.org>

	  sys/osxvideo/osxvideosink.*: Decent effort at porting to 0.10.  Needs cleanup on OS/X.
	  Original commit message from CVS:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  Decent effort at porting to 0.10.  Needs cleanup on OS/X.

2006-12-17 05:07:07 +0000  Vijay Santhanam <vijay@santhanam.gmail.com>

	  sys/osxvideo/: Preliminary patch for porting osxvideosink
	  Original commit message from CVS:
	  Patch by: Vijay Santhanam <vijay santhanam gmail com>
	  * sys/osxvideo/Makefile.am:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  Preliminary patch for porting osxvideosink

2006-12-16 16:21:26 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/videomixer/videomixer.c: Introduce some locking around the videomixer state so that it does not crash when adding...
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
	  (gst_videomixer_set_master_geometry),
	  (gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free),
	  (gst_videomixer_reset), (gst_videomixer_init),
	  (gst_videomixer_finalize), (gst_videomixer_request_new_pad),
	  (gst_videomixer_release_pad), (gst_videomixer_collected),
	  (gst_videomixer_change_state):
	  Introduce some locking around the videomixer state so that it does not
	  crash when adding/removing pads. Fixes #383043.

2006-12-16 15:25:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: We don't support seeking in streaming mode, so don't even try.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types),
	  (gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event):
	  We don't support seeking in streaming mode, so don't even try.
	  Implement seeking query so apps can query seekability properly
	  (see #365414). Fix duration query.

2006-12-16 11:42:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Make sure libcaca can actually be used instead of just checking for /usr/bin/caca-config, so we don't w...
	  Original commit message from CVS:
	  * configure.ac:
	  Make sure libcaca can actually be used instead of just checking for
	  /usr/bin/caca-config, so we don't wrongly try to build cacasink when
	  cross-compiling (fixes #384587).

2006-12-15 10:54:28 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  adding doap file
	  Original commit message from CVS:
	  * Makefile.am:
	  * gst-plugins-good.doap:
	  * gst-plugins-good.spec.in:
	  adding doap file

2006-12-14 16:20:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: libflac-1.1.3 changed API again, but we can't build against it yet, so make sure our check doesn't use ...
	  Original commit message from CVS:
	  * configure.ac:
	  libflac-1.1.3 changed API again, but we can't build against it yet,
	  so make sure our check doesn't use libflac-1.1.3 and add a comment
	  to this effect.

2006-12-14 14:25:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/effectv/gstquark.c: Add some NULL pointer checks (possibly related to #385623).
	  Original commit message from CVS:
	  * gst/effectv/gstquark.c: (gst_quarktv_transform),
	  (gst_quarktv_planetable_clear):
	  Add some NULL pointer checks (possibly related to #385623).

2006-12-14 10:15:24 +0000  Roland Kay <roland.kay@ox.compsoc.net>

	  ext/lame/gstlame.*: Fix leak (by calling lame_init_params() before lame_close()); handle
	  Original commit message from CVS:
	  Based on patch by: Roland Kay  <roland.kay at ox compsoc net>
	  * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain),
	  (gst_lame_setup):
	  * ext/lame/gstlame.h:
	  Fix leak (by calling lame_init_params() before lame_close()); handle
	  NULL return from lame_init() more gracefully. Fixes #385311.

2006-12-13 17:12:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Add AMR-WB to the list of supported formats.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
	  (gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
	  (qtdemux_audio_caps):
	  Add AMR-WB to the list of supported formats.

2006-12-12 18:45:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: In streaming mode, if the first buffer we get doesn't have an offset, fix it up to be 0, otherwise trimming won...
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag),
	  (gst_tag_demux_chain):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  In streaming mode, if the first buffer we get doesn't have an
	  offset, fix it up to be 0, otherwise trimming won't work later on
	  and we'll be typefinding application/x-id3, which may result in
	  decodebin plugging an endless number of id3demux elements as a
	  consequence. Fixes #385031.

2006-12-11 21:21:16 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/sunaudio/gstsunaudiosink.c: Ignore the buffer_time the sound device reports. Turns out it is sometimes completely...
	  Original commit message from CVS:
	  * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
	  Ignore the buffer_time the sound device reports. Turns out it is
	  sometimes completely bogus and we're better off without it.

2006-12-11 17:33:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Fix non-working redirects from inetfilm.com (handle 'alis' reference data type as well). Fixes...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_tree):
	  Fix non-working redirects from inetfilm.com (handle 'alis' reference
	  data type as well). Fixes #378613.

2006-12-11 13:59:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Try harder to extract the framerate for video tracks correctly and save it directly instead of convert...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_video_caps):
	  * gst/matroska/matroska-ids.c:
	  (gst_matroska_track_init_video_context):
	  * gst/matroska/matroska-ids.h:
	  Try harder to extract the framerate for video tracks correctly and
	  save it directly instead of converting it back and forth a few
	  times. Mostly makes a difference for very small framerates (<1).
	  Fixes #380199.

2006-12-11 11:41:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gconf/gstgconfaudiosrc.*: Remove gconf notify hook when the gconfaudiosrc element is destroyed, otherwise the cal...
	  Original commit message from CVS:
	  * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_init),
	  (gst_gconf_audio_src_dispose), (do_toggle_element):
	  * ext/gconf/gstgconfaudiosrc.h:
	  Remove gconf notify hook when the gconfaudiosrc element is
	  destroyed, otherwise the callback may be called on an
	  already-destroyed instance and bad things happen. Should fix
	  #378184.
	  Also ignore gconf key changes when the source is already running.

2006-12-09 19:27:28 +0000  Sebastian Dröge <mail@slomosnail.de>

	  gst/apetag/gstapedemux.c: We need to be able to read and parse any possible floating point string format ("1,234" or ...
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge  <mail at slomosnail de>
	  * gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	  We need to be able to read and parse any possible floating point string
	  format ("1,234" or "1.234") irrespective of the current locale. g_strod()
	  will parse the former only in certain locales though, so we really need
	  to canonicalise the separator to '.' and then use g_ascii_strtod() to
	  make sure we can parse either version at all times.
	  Fixes #382982 for real.

2006-12-09 16:17:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/sunaudio/: Use the sunaudio debug category.
	  Original commit message from CVS:
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  * sys/sunaudio/gstsunaudiosrc.c:
	  Use the sunaudio debug category.
	  * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_finalize),
	  (gst_sunaudiosink_class_init), (gst_sunaudiosink_init),
	  (gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property),
	  (gst_sunaudiosink_open), (gst_sunaudiosink_close),
	  (gst_sunaudiosink_prepare), (gst_sunaudio_sink_do_delay),
	  (gst_sunaudiosink_write), (gst_sunaudiosink_delay),
	  (gst_sunaudiosink_reset):
	  * sys/sunaudio/gstsunaudiosink.h:
	  Uses the sunaudio debug category for all debug output
	  Implements the _delay() callback to synchronise video playback better
	  Change the segtotal and segsize values back to the parent class
	  defaults (taken from buffer_time and latency_times of 200ms and 10ms
	  respectively)
	  Measure the samples written to the device vs. played.
	  Keep track of segments in the device by writing empty eof frames, and
	  sleep using a GCond when we get too far ahead and risk overrunning the
	  sink's ringbuffer.
	  Fixes: #360673

2006-12-08 21:12:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Correct the attribution of the previous commit. The patch in question was written by Brian Cameron.
	  Original commit message from CVS:
	  Correct the attribution of the previous commit. The patch in
	  question was written by Brian Cameron.

2006-12-08 17:06:43 +0000  René Stadler <mail@renestadler.de>

	  gst/qtdemux/qtdemux.c: Fix caps for 24 bit raw PCM audio (2).
	  Original commit message from CVS:
	  Patch by: René Stadler  <mail at renestadler de>
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
	  (gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
	  (qtdemux_audio_caps):
	  Fix caps for 24 bit raw PCM audio (2).
	  Fixes #383471.

2006-12-08 16:38:18 +0000  Sebastian Dröge <mail@slomosnail.de>

	  gst/audiofx/audiopanorama.*: Fix audiopanorame with float samples. Fixes #383726.
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge  <mail at slomosnail de >
	  * gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
	  (gst_audio_panorama_set_caps), (gst_audio_panorama_transform):
	  * gst/audiofx/audiopanorama.h:
	  Fix audiopanorame with float samples. Fixes #383726.

2006-12-08 15:12:01 +0000  Padraig O'Briain <padraig.obriain@sun.com>

	  sys/sunaudio/: Implement reset functions to unblock the src/sink more quickly on state change requests.
	  Original commit message from CVS:
	  * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_reset):
	  * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open),
	  (gst_sunaudiosrc_reset):
	  Implement reset functions to unblock the src/sink more quickly on
	  state change requests.
	  Patch by: Padraig O'Briain <padraig dot obriain at sun dot com>

2006-12-08 14:42:42 +0000  Jerry Tan <jerry.tan@sun.com>

	  sys/sunaudio/gstsunaudiomixer.c: Construct the correct mixer device name when the AUDIODEV env var is set.
	  Original commit message from CVS:
	  * sys/sunaudio/gstsunaudiomixer.c:
	  (gst_sunaudiomixer_change_state):
	  Construct the correct mixer device name when the AUDIODEV env var
	  is set.
	  Patch by: Jerry Tan <jerry.tan at sun dot com>
	  Fixes: #383596

2006-12-08 14:32:51 +0000  Jerry Tan <jerry.tan@sun.com>

	  sys/sunaudio/gstsunaudiosrc.c: Apply patch to open the mixer control and set the MULTIPLE_OPEN ioctl. On solaris, the...
	  Original commit message from CVS:
	  * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	  Apply patch to open the mixer control and set the MULTIPLE_OPEN
	  ioctl. On solaris, the mixer device doesn't need opening non-blocking
	  - it can be opened by multiple processes by default, but needs the ioctl 	for multiple opens within 1 process.
	  Patch by: Jerry Tan <jerry.tan at sun dot com>
	  Fixes: #349015

2006-12-07 17:30:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/smpte/: Port to 0.10 some more.
	  Original commit message from CVS:
	  * gst/smpte/gstmask.h:
	  * gst/smpte/gstsmpte.c: (gst_smpte_class_init),
	  (gst_smpte_setcaps), (gst_smpte_init), (gst_smpte_reset),
	  (gst_smpte_collected), (gst_smpte_set_property),
	  (gst_smpte_get_property), (gst_smpte_change_state), (plugin_init):
	  * gst/smpte/gstsmpte.h:
	  Port to 0.10 some more.
	  Added duration property to specify the duration of the transition.
	  Make framerate a fraction.
	  Deprecate fps property, we only use negotiated fps.
	  Added docs.
	  Fix collectpad usage.
	  Reset state in READY.
	  Send NEWSEGMENT event.
	  Fix racy updates of object properties.
	  Added debug category.
	  Fixes #383323.

2006-12-07 11:35:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Handle more H263 variants.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
	  (gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
	  (qtdemux_video_caps):
	  Handle more H263 variants.

2006-12-06 15:06:04 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/videomixer/videomixer.c: Don't reset xpos and ypos in the setcaps function because causes unexpected behaviour.
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * gst/videomixer/videomixer.c:
	  (gst_videomixer_set_master_geometry),
	  (gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free):
	  Don't reset xpos and ypos in the setcaps function because causes
	  unexpected behaviour.
	  Fixes #382179.

2006-12-06 14:45:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/multipart/multipartmux.c: Keep track of the buffer timestamp in the collectdata member instead of modifying the b...
	  Original commit message from CVS:
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_compare_pads),
	  (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected):
	  Keep track of the buffer timestamp in the collectdata member instead
	  of modifying the buffer without making the metadata writable first.
	  Fixes #382277.

2006-12-06 14:33:54 +0000  Rob Taylor <robtaylor@floopily.org>

	  gst/udp/gstudpsrc.c: If using multicast in udpsrc, bind to the multicast address rather than
	  Original commit message from CVS:
	  Patch by: Rob Taylor <robtaylor at floopily dot org>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  If using multicast in udpsrc, bind to the multicast address rather than
	  IN_ADDR_ANY.
	  This allows the simultanous use of multiple udpsrcs listening on
	  different multicat addresses. Without this all udpsrcs will receive all
	  packets from all subscribed multicast addresses.
	  Fixes #383001.

2006-12-06 13:35:52 +0000  Jonathan Matthew <jonathan@0kaolin.wh9.net>

	  ext/taglib/gstid3v2mux.cc: Don't attempt to write a NULL frame into the ID3 tag set when the createFrame method retur...
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc:
	  Don't attempt to write a NULL frame into the ID3 tag set when the
	  createFrame method returned NULL.
	  Fixes: #381857
	  Patch by: Jonathan Matthew <jonathan at 0kaolin wh9 net >

2006-12-06 13:16:59 +0000  Sebastian Dröge <mail@slomosnail.de>

	  gst/apetag/gstapedemux.c: Use g_strtod() instead of sscanf to parse doubles, so that it will try parsing in the C loc...
	  Original commit message from CVS:
	  * gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	  Use g_strtod() instead of sscanf to parse doubles, so that it will
	  try parsing in the C locale if the current locale fails.
	  Fixes: #382982
	  Patch by: Sebastian Dröge  <mail at slomosnail de >

2006-12-01 10:31:46 +0000  Sergey Scobich <sergey.scobich@gmail.com>

	  win32/MANIFEST: Fix compilation on win32 under VS8
	  Original commit message from CVS:
	  * win32/MANIFEST:
	  Fix compilation on win32 under VS8
	  Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
	  Partially fixes #381175

2006-11-30 16:48:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: accept all mpegversions,fixes #380825 spotted by: Jerome Alet
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c:
	  accept all mpegversions,fixes #380825
	  spotted by: Jerome Alet

2006-11-30 16:46:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/v4l2src_calls.c: cleanup the error message a bit more
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
	  (gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
	  (gst_v4l2src_capture_init), (gst_v4l2src_buffer_finalize):
	  cleanup the error message a bit more

2006-11-30 15:08:08 +0000  René Stadler <mail@renestadler.de>

	  gst/replaygain/gstrganalysis.c: Call the base class handler.  Fixes #380610.
	  Original commit message from CVS:
	  Patch by: René Stadler  <mail at renestadler de>
	  * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_event):
	  Call the base class handler.  Fixes #380610.

2006-11-28 12:30:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libcaca/gstcacasink.c: Fix width and height properties.
	  Original commit message from CVS:
	  * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
	  Fix width and height properties.
	  * ext/libcaca/gstcacasink.h:
	  Fix compilation on newer libcaca that require us to include a new
	  header. Fixes #379918.

2006-11-28 11:52:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Add method so that extensions can choose to disable the setup of a stream.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	  * gst/rtsp/gstrtspsrc.h:
	  * gst/rtsp/rtspext.h:
	  * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream),
	  (rtsp_ext_wms_get_context):
	  Add method so that extensions can choose to disable the setup of
	  a stream.
	  Make the WMS extension skip setup of x-wms-rtx streams. Fixes #377792.

2006-11-27 17:16:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Remove some asserts and replace them with a proper error message. Fixes #379261.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
	  (gst_qtdemux_handle_src_query), (qtdemux_parse_trak):
	  Remove some asserts and replace them with a proper error
	  message. Fixes #379261.

2006-11-27 16:30:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ChangeLog:
	  mention bug fix
	  Original commit message from CVS:
	  mention bug fix

2006-11-27 16:29:07 +0000  Jonas Holmberg <jonas.holmberg@axis.com>

	  gst/multipart/multipartmux.c: Push header in a separate buffer instead of memcpy:ing all data
	  Original commit message from CVS:
	  Patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	  Push header in a separate buffer instead of memcpy:ing all data
	  Change LF => CRLF in headers
	  Move trailing LF to header

2006-11-27 16:26:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmpadepay.c: Small buffer overflow fix and improve debugging.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_chain):
	  Small buffer overflow fix and improve debugging.

2006-11-24 08:58:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/esd/: remove obsolete _factory_init protos
	  Original commit message from CVS:
	  * ext/esd/esdmon.h:
	  * ext/esd/esdsink.h:
	  remove obsolete _factory_init protos

2006-11-24 07:46:54 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: remove dead code, tweak debugs statements, add comments, use _uint64_scale instead _uint64_sca...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_index_entry_for_time),
	  (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
	  (gst_avi_demux_peek_chunk), (gst_avi_demux_parse_subindex),
	  (gst_avi_demux_read_subindexes_push),
	  (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	  (gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
	  (gst_avi_demux_massage_index),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
	  (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
	  (gst_avi_demux_stream_data), (gst_avi_demux_loop):
	  remove dead code, tweak debugs statements, add comments, use
	  _uint64_scale instead _uint64_scale_int when using guint64 values,
	  small optimizations, reflow some error handling

2006-11-22 17:39:13 +0000  Edward Hervey <bilboed@bilboed.com>

	  po/.cvsignore: We never put .pot files in cvs. Let's ignore them all.
	  Original commit message from CVS:
	  * po/.cvsignore:
	  We never put .pot files in cvs. Let's ignore them all.

2006-11-21 12:57:50 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  enalbe LADSPA plugin in spec file
	  Original commit message from CVS:
	  enalbe LADSPA plugin in spec file

2006-11-19 18:46:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/POTFILES.in: ... but better exclude files that aren't disted.
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  ... but better exclude files that aren't disted.

2006-11-19 16:32:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/POTFILES.in: Add v4l2 source files to list of files with translations, so the strings are actually extracted (howe...
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  Add v4l2 source files to list of files with translations, so the
	  strings are actually extracted (however bad they still may be).

2006-11-19 16:30:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videobox/gstvideobox.c: Minor clean-ups: const-ify static array, remove trailing comma from use GST_DEBUG_FUNCPTR.
	  Original commit message from CVS:
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init):
	  Minor clean-ups: const-ify static array, remove trailing comma from
	  last enum (gcc-2.9x trips over that), use GST_DEBUG_FUNCPTR.

2006-11-19 13:41:53 +0000  René Stadler <mail@renestadler.de>

	  gst/id3demux/id3v2frames.c: Make sure that g_free always gets called on the same pointer that was returned by g_mallo...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
	  Make sure that g_free always gets called on the same pointer that was
	  returned by g_malloc.  Fixes #376594.
	  Do not leak memory if decompressed size is wrong.
	  Remove unneeded check of return value of g_malloc.
	  Patch by: René Stadler <mail@renestadler.de>

2006-11-18 18:14:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/v4l2src_calls.c: Add missing curly brackets.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_capture_deinit):
	  Add missing curly brackets.

2006-11-17 14:54:01 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* sys/v4l2/v4l2src_calls.c:
	  Fix capture_deinit.
	  Original commit message from CVS:
	  Fix capture_deinit.

2006-11-16 15:36:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-mux.c: Use GST_DEBUG_FUNCPTR; activate request pad before returning it.
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
	  (gst_matroska_mux_request_new_pad):
	  Use GST_DEBUG_FUNCPTR; activate request pad before returning it.
	  * tests/check/elements/matroskamux.c: (setup_src_pad),
	  (setup_sink_pad), (GST_START_TEST):
	  Activate pads before using them.

2006-11-16 15:04:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Initialise variable to get rid of bogus compiler warning.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
	  Initialise variable to get rid of bogus compiler warning.

2006-11-16 07:26:17 +0000  Ville Syrjala <ville.syrjala@movial.fi>

	  gst/rtp/: Specify H.263 variant and version in the caps (fixes #361637)
	  Original commit message from CVS:
	  Patch by: Ville Syrjala <ville.syrjala@movial.fi>
	  * gst/rtp/gstrtph263pay.c:
	  * gst/rtp/gstrtph263pdepay.c:
	  * gst/rtp/gstrtph263ppay.c:
	  Specify H.263 variant and version in the caps (fixes #361637)

2006-11-15 17:44:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspconnection.c: Don't set a data pointer to NULL and a size > 0 when we deal with empty packets.
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c: (read_body):
	  Don't set a data pointer to NULL and a size > 0 when we deal
	  with empty packets.
	  * gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
	  (rtsp_message_init_response), (rtsp_message_init_data),
	  (rtsp_message_unset), (rtsp_message_free),
	  (rtsp_message_take_body):
	  Check that we can't create invalid empty packets.

2006-11-15 12:35:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: Some small clean-ups: use enums instead of hard-coded numbers, const-ify element details, re-factor som...
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge  <slomo@circular-chaos.org>
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	  (gst_wavpack_dec_init), (gst_wavpack_dec_change_state):
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_base_init),
	  (gst_wavpack_enc_class_init), (gst_wavpack_enc_reset),
	  (gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config),
	  (gst_wavpack_enc_change_state):
	  * ext/wavpack/gstwavpackparse.c:
	  Some small clean-ups: use enums instead of hard-coded numbers,
	  const-ify element details, re-factor some code into _reset()
	  functions (#352605).

2006-11-15 12:08:20 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/matroska/matroska-mux.*: Add basic tag writing support; implement releasing pads (#374658).
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet be>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_add_interfaces),
	  (gst_matroska_mux_class_init), (gst_matroska_pad_free),
	  (gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event),
	  (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
	  (gst_matroska_mux_track_header), (gst_matroska_mux_start),
	  (gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish):
	  * gst/matroska/matroska-mux.h:
	  Add basic tag writing support; implement releasing pads (#374658).

2006-11-15 11:19:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: Handle opaque/unspecified A_AAC audio codec ID (fixes #374737).
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_audio_caps):
	  Handle opaque/unspecified A_AAC audio codec ID (fixes #374737).

2006-11-15 00:12:19 +0000  David Schleef <ds@schleef.org>

	  gst/matroska/matroska-mux.c: Add Dirac fourcc.
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: Add Dirac fourcc.

2006-11-14 20:07:22 +0000  Sergey Scobich <sergey.scobich@gmail.com>

	  win32/vs8/: Make end-of-line returns unixy, so that when the files are checked out on win32 the line returns will be ...
	  Original commit message from CVS:
	  Patch by: Sergey Scobich  <sergey.scobich at gmail com>
	  * win32/vs8/gst-plugins-good.sln:
	  * win32/vs8/libgst1394.vcproj:
	  * win32/vs8/libgstaasink.vcproj:
	  * win32/vs8/libgstalaw.vcproj:
	  * win32/vs8/libgstalpha.vcproj:
	  * win32/vs8/libgstalphacolor.vcproj:
	  * win32/vs8/libgstannodex.vcproj:
	  * win32/vs8/libgstapetag.vcproj:
	  * win32/vs8/libgstaudiofx.vcproj:
	  * win32/vs8/libgstauparse.vcproj:
	  * win32/vs8/libgstautodetect.vcproj:
	  * win32/vs8/libgstavi.vcproj:
	  * win32/vs8/libgstcacasink.vcproj:
	  * win32/vs8/libgstcdio.vcproj:
	  * win32/vs8/libgstcutter.vcproj:
	  * win32/vs8/libgstdv.vcproj:
	  * win32/vs8/libgsteffectv.vcproj:
	  * win32/vs8/libgstflac.vcproj:
	  * win32/vs8/libgstflxdec.vcproj:
	  * win32/vs8/libgstgoom.vcproj:
	  * win32/vs8/libgsticydemux.vcproj:
	  * win32/vs8/libgstid3demux.vcproj:
	  * win32/vs8/libgstjpeg.vcproj:
	  * win32/vs8/libgstladspa.vcproj:
	  * win32/vs8/libgstlevel.vcproj:
	  * win32/vs8/libgstmatroska.vcproj:
	  * win32/vs8/libgstmikmod.vcproj:
	  * win32/vs8/libgstmng.vcproj:
	  * win32/vs8/libgstmonoscope.vcproj:
	  * win32/vs8/libgstmulaw.vcproj:
	  * win32/vs8/libgstmultipart.vcproj:
	  * win32/vs8/libgstpng.vcproj:
	  * win32/vs8/libgstrtp.vcproj:
	  * win32/vs8/libgstrtsp.vcproj:
	  * win32/vs8/libgstshout2.vcproj:
	  * win32/vs8/libgstsmpte.vcproj:
	  * win32/vs8/libgstspeex.vcproj:
	  * win32/vs8/libgsttaglib.vcproj:
	  * win32/vs8/libgstudp.vcproj:
	  * win32/vs8/libgstvideobalance.vcproj:
	  * win32/vs8/libgstvideobox.vcproj:
	  * win32/vs8/libgstvideoflip.vcproj:
	  * win32/vs8/libgstvideomixer.vcproj:
	  * win32/vs8/libgstwavenc.vcproj:
	  * win32/vs8/libgstwavparse.vcproj:
	  Make end-of-line returns unixy, so that when the files are checked
	  out on win32 the line returns will be 0d 0a and not 0d 0d 0a.
	  Hopefully fixes #366492.

2006-11-14 15:55:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Disable init_frames delay timestamp adjustment, it does not seem to be needed at all. Fixes #3...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
	  Disable init_frames delay timestamp adjustment, it does not
	  seem to be needed at all. Fixes #369621.

2006-11-14 11:43:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Don't parse extra sample params for raw pcm. Fixes #374914.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
	  (gst_qtdemux_handle_src_query), (qtdemux_parse_trak):
	  Don't parse extra sample params for raw pcm. Fixes #374914.

2006-11-14 10:29:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/lame/gstlame.*: Make lame timestamp flushed eos buffer by some additional timestamp accounting. Fixes #374760.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain),
	  (gst_lame_change_state):
	  * ext/lame/gstlame.h:
	  Make lame timestamp flushed eos buffer by some additional timestamp
	  accounting. Fixes #374760.

2006-11-13 18:31:18 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/videomixer/videomixer.c: Fix memleak by unref'ing collectpads instance (when finalizing)
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet be>
	  * gst/videomixer/videomixer.c:
	  (gst_videomixer_set_master_geometry),
	  (gst_videomixer_pad_sink_setcaps), (gst_videomixer_class_init),
	  (gst_videomixer_collect_free), (gst_videomixer_reset),
	  (gst_videomixer_init), (gst_videomixer_finalize),
	  (gst_videomixer_request_new_pad), (gst_videomixer_release_pad),
	  (gst_videomixer_collected), (gst_videomixer_change_state):
	  Fix memleak by unref'ing collectpads instance (when finalizing)
	  Implement releasing a request pad. Fixes #374479.

2006-11-10 20:08:42 +0000  Sergey Scobich <sergey.scobich@gmail.com>

	  win32/vs8/: Add VS8 project files (note that many of the plugins in ext are disabled by default). Fixes #366492.
	  Original commit message from CVS:
	  Patch by: Sergey Scobich  <sergey.scobich at gmail com>
	  * win32/vs8/gst-plugins-good.sln:
	  * win32/vs8/libgst1394.vcproj:
	  * win32/vs8/libgstaasink.vcproj:
	  * win32/vs8/libgstalaw.vcproj:
	  * win32/vs8/libgstalpha.vcproj:
	  * win32/vs8/libgstalphacolor.vcproj:
	  * win32/vs8/libgstannodex.vcproj:
	  * win32/vs8/libgstapetag.vcproj:
	  * win32/vs8/libgstaudiofx.vcproj:
	  * win32/vs8/libgstauparse.vcproj:
	  * win32/vs8/libgstautodetect.vcproj:
	  * win32/vs8/libgstavi.vcproj:
	  * win32/vs8/libgstcacasink.vcproj:
	  * win32/vs8/libgstcdio.vcproj:
	  * win32/vs8/libgstcutter.vcproj:
	  * win32/vs8/libgstdv.vcproj:
	  * win32/vs8/libgsteffectv.vcproj:
	  * win32/vs8/libgstflac.vcproj:
	  * win32/vs8/libgstflxdec.vcproj:
	  * win32/vs8/libgstgoom.vcproj:
	  * win32/vs8/libgsticydemux.vcproj:
	  * win32/vs8/libgstid3demux.vcproj:
	  * win32/vs8/libgstjpeg.vcproj:
	  * win32/vs8/libgstladspa.vcproj:
	  * win32/vs8/libgstlevel.vcproj:
	  * win32/vs8/libgstmatroska.vcproj:
	  * win32/vs8/libgstmikmod.vcproj:
	  * win32/vs8/libgstmng.vcproj:
	  * win32/vs8/libgstmonoscope.vcproj:
	  * win32/vs8/libgstmulaw.vcproj:
	  * win32/vs8/libgstmultipart.vcproj:
	  * win32/vs8/libgstpng.vcproj:
	  * win32/vs8/libgstrtp.vcproj:
	  * win32/vs8/libgstrtsp.vcproj:
	  * win32/vs8/libgstshout2.vcproj:
	  * win32/vs8/libgstsmpte.vcproj:
	  * win32/vs8/libgstspeex.vcproj:
	  * win32/vs8/libgsttaglib.vcproj:
	  * win32/vs8/libgstudp.vcproj:
	  * win32/vs8/libgstvideobalance.vcproj:
	  * win32/vs8/libgstvideobox.vcproj:
	  * win32/vs8/libgstvideoflip.vcproj:
	  * win32/vs8/libgstvideomixer.vcproj:
	  * win32/vs8/libgstwavenc.vcproj:
	  * win32/vs8/libgstwavparse.vcproj:
	  Add VS8 project files (note that many of the plugins in ext are
	  disabled by default). Fixes #366492.

2006-11-10 19:18:33 +0000  David Schleef <ds@schleef.org>

	  gst/multifile/Makefile.am: Let's not depend on a file that doesn't exist.
	  Original commit message from CVS:
	  * gst/multifile/Makefile.am:
	  Let's not depend on a file that doesn't exist.

2006-11-10 18:51:10 +0000  David Schleef <ds@schleef.org>

	  Revive multifile[src|sink].
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/multifile/Makefile.am:
	  * gst/multifile/gstmultifile.c:
	  * gst/multifile/gstmultifilesink.c:
	  * gst/multifile/gstmultifilesrc.c:
	  * gst/multifile/multifile.vproj:
	  Revive multifile[src|sink].

2006-11-10 08:09:05 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/v4l2src_calls.c: we do not translate debug messages
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  we do not translate debug messages

2006-11-08 12:04:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/flx/gstflxdec.c: fix categorisation, make short desc more explicit, remove unused code
	  Original commit message from CVS:
	  * gst/flx/gstflxdec.c: (gst_flxdec_class_init):
	  fix categorisation, make short desc more explicit, remove unused code
	  Fixes #372021

2006-11-08 01:30:39 +0000  Christian Schaller <uraeus@gnome.org>

	  gst/rtp/: Fix element descriptions.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16depay.c:
	  * gst/rtp/gstrtpamrdepay.c:
	  * gst/rtp/gstrtpamrpay.c:
	  * gst/rtp/gstrtpgsmdepay.c:
	  * gst/rtp/gstrtph263pay.c:
	  * gst/rtp/gstrtph263pdepay.c:
	  * gst/rtp/gstrtph263ppay.c:
	  * gst/rtp/gstrtph264depay.c:
	  * gst/rtp/gstrtpmp2tdepay.c:
	  * gst/rtp/gstrtpmp4gdepay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmp4vdepay.c:
	  * gst/rtp/gstrtpmp4vpay.c:
	  * gst/rtp/gstrtpmpadepay.c:
	  * gst/rtp/gstrtpmpapay.c:
	  * gst/rtp/gstrtppcmadepay.c:
	  * gst/rtp/gstrtppcmapay.c:
	  * gst/rtp/gstrtppcmudepay.c:
	  * gst/rtp/gstrtppcmupay.c:
	  * gst/rtp/gstrtpspeexdepay.c:
	  * gst/rtp/gstrtpspeexpay.c:
	  * gst/rtp/gstrtpsv3vdepay.c:
	  Fix element descriptions.

2006-11-08 01:29:51 +0000  Christian Schaller <uraeus@gnome.org>

	  gst/rtp/: Fix description.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpvorbisdepay.c:
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_handle_buffer):
	  Fix description.
	  Small cleanup in the payloader.

2006-11-08 01:28:00 +0000  Christian Schaller <uraeus@gnome.org>

	  gst/rtp/: Add theora pay/depayloaders.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_base_init),
	  (gst_rtp_theora_depay_class_init), (gst_rtp_theora_depay_init),
	  (gst_rtp_theora_depay_finalize),
	  (gst_rtp_theora_depay_parse_configuration),
	  (gst_rtp_theora_depay_setcaps),
	  (gst_rtp_theora_depay_switch_codebook),
	  (gst_rtp_theora_depay_process),
	  (gst_rtp_theora_depay_set_property),
	  (gst_rtp_theora_depay_get_property),
	  (gst_rtp_theora_depay_change_state),
	  (gst_rtp_theora_depay_plugin_init):
	  * gst/rtp/gstrtptheoradepay.h:
	  * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_base_init),
	  (gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_init),
	  (gst_rtp_theora_pay_setcaps), (gst_rtp_theora_pay_reset_packet),
	  (gst_rtp_theora_pay_init_packet),
	  (gst_rtp_theora_pay_flush_packet),
	  (gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
	  (gst_rtp_theora_pay_handle_buffer),
	  (gst_rtp_theora_pay_plugin_init):
	  * gst/rtp/gstrtptheorapay.h:
	  Add theora pay/depayloaders.

2006-11-07 01:43:06 +0000  Christian Schaller <uraeus@gnome.org>

	  gst/rtp/Makefile.am: We depend on gsttag to generate the vorbis comments.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  We depend on gsttag to generate the vorbis comments.
	  * gst/rtp/gstrtpvorbisdepay.c:
	  (gst_rtp_vorbis_depay_parse_configuration),
	  (gst_rtp_vorbis_depay_setcaps),
	  (gst_rtp_vorbis_depay_switch_codebook),
	  (gst_rtp_vorbis_depay_process):
	  * gst/rtp/gstrtpvorbisdepay.h:
	  Parse configuration string in the depayloader.
	  Implement selecting and switching to a new codebook.
	  Receiving vorbis over RTP now works.
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_reset_packet),
	  (gst_rtp_vorbis_pay_init_packet),
	  (gst_rtp_vorbis_pay_finish_headers),
	  (gst_rtp_vorbis_pay_handle_buffer):
	  * gst/rtp/gstrtpvorbispay.h:
	  Set timestamps on outgoing buffers and RTP packets.
	  Fix configuration string, prepend number of Packet headers.
	  Fix encoding of ident string.
	  Add delivery-method to caps.
	  Streaming vorbis over RTP now works.

2006-11-06 20:52:10 +0000  Christian Schaller <uraeus@gnome.org>

	  gst/rtp/gstrtpvorbispay.*: Generate a valid configuration string in the caps based on the vorbis headers.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
	  (gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_parse_id),
	  (gst_rtp_vorbis_pay_handle_buffer):
	  * gst/rtp/gstrtpvorbispay.h:
	  Generate a valid configuration string in the caps based on the
	  vorbis headers.

2006-11-02 20:13:26 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Fix enum nicks; only emit no-more-pads once; add support for very fast encoding mode in upcoming 4.40.0 release (#369...
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo at circular-chaos.org>
	  * configure.ac:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type),
	  (gst_wavpack_enc_correction_mode_get_type),
	  (gst_wavpack_enc_joint_stereo_mode_get_type),
	  (gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config):
	  Fix enum nicks; only emit no-more-pads once; add support for very
	  fast encoding mode in upcoming 4.40.0 release (#369539).

2006-11-02 14:43:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cdio/: Move CD-TEXT utility function into common file so it can also be used by a future cdioparanoiasrc.
	  Original commit message from CVS:
	  * ext/cdio/gstcdio.c: (gst_cdio_get_cdtext):
	  * ext/cdio/gstcdio.h:
	  * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open):
	  Move CD-TEXT utility function into common file so it can also be
	  used by a future cdioparanoiasrc.

2006-11-01 19:48:26 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	  Improved comments in ELEMENT_ERROR/WARNING and added "#if 0" to xoverlay code that is still not implemented.
	  Original commit message from CVS:
	  Improved comments in ELEMENT_ERROR/WARNING and added "#if 0" to xoverlay code that is still not implemented.

2006-11-01 13:59:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: We require a -base more recent than 0.10.9, so it's safe to use
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  We require a -base more recent than 0.10.9, so it's safe to use
	  GST_TYPE_TAG_IMAGE_TYPE unconditionally now.
	  * ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
	  Use _newsegment_full() now that we depend on a recent enough core.
	  * gst/wavparse/gstwavparse.c:
	  Remove cruft that we don't need any longer now that we depend on
	  a recent enough -base.

2006-11-01 10:19:18 +0000  Sergey Scobich <sergey.scobich@gmail.com>

	  sys/: Wait until the window is created before using it; guard unistd.h includes with HAVE_UNISTD_H. (#366523)
	  Original commit message from CVS:
	  Patch by: Sergey Scobich  <sergey dot scobich at gmail com>
	  * sys/directdraw/gstdirectdrawsink.c:
	  (gst_directdrawsink_window_thread),
	  (gst_directdrawsink_create_default_window):
	  * sys/directdraw/gstdirectdrawsink.h:
	  * sys/directsound/gstdirectsoundsink.c:
	  Wait until the window is created before using it; guard unistd.h
	  includes with HAVE_UNISTD_H. (#366523)
	  * win32/vs8/libgstdirectdraw.vcproj:
	  * win32/vs8/libgstdirectsound.vcproj:
	  Update project files.

2006-10-31 10:52:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Fix and activate ILBC pay and depayloaders. Fixes #368162.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_init),
	  (gst_rtpilbcpay_setcaps):
	  Fix and activate ILBC pay and depayloaders. Fixes #368162.

2006-10-31 10:31:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Handle unbounded length streams a bit better. Fixes #367696.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
	  (gst_qtdemux_handle_src_query), (qtdemux_parse_tree),
	  (qtdemux_parse_trak):
	  Handle unbounded length streams a bit better. Fixes #367696.

2006-10-31 09:44:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/speex/gstspeexdec.c: Some small cleanups, use _scale.
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_convert),
	  (speex_dec_sink_event), (speex_dec_chain_parse_header):
	  Some small cleanups, use _scale.

2006-10-31 09:29:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Use higher precision scale function.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
	  Use higher precision scale function.

2006-10-30 16:18:18 +0000  Michal Benes <michal.benes@itonis.tv>

	  gst/matroska/matroska-demux.c: Fix several issues with encoded/compressed/encrypted/signed tracks; also, remove super...
	  Original commit message from CVS:
	  Patch by: Michal Benes  <michal dot benes at itonis tv>
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_encoding_cmp),
	  (gst_matroska_demux_read_track_encodings),
	  (gst_matroska_decode_buffer):
	  Fix several issues with encoded/compressed/encrypted/signed tracks;
	  also, remove superfluous newline characters from some debug
	  statements. (#366155)

2006-10-30 09:24:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/jpeg/: Various cleanups, capsnego and leak fixes.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps):
	  * ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init),
	  (gst_smokedec_init), (gst_smokedec_finalize), (gst_smokedec_chain),
	  (gst_smokedec_change_state):
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
	  (gst_smokeenc_init), (gst_smokeenc_finalize),
	  (gst_smokeenc_getcaps), (gst_smokeenc_setcaps),
	  (gst_smokeenc_resync), (gst_smokeenc_chain),
	  (gst_smokeenc_set_property), (gst_smokeenc_get_property),
	  (gst_smokeenc_change_state):
	  Various cleanups, capsnego and leak fixes.

2006-10-30 08:17:08 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/videomixer/videomixer.c: Fix videomixer so that it can handle any combination of framerates.
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet be>
	  * gst/videomixer/videomixer.c: (gst_videomixer_update_queues):
	  Fix videomixer so that it can handle any combination of framerates.
	  Fixes #367221.

2006-10-28 16:37:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Fix position query for audio. also fixes timestamps in streaming mode and bug #364958.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	  (gst_avi_demux_parse_file_header),
	  (gst_avi_demux_stream_init_push), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data),
	  (gst_avi_demux_chain):
	  Fix position query for audio. also fixes timestamps in streaming
	  mode and bug #364958.
	  Small cleanups.

2006-10-27 17:10:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libpng/gstpngenc.*: Fix strides. Fixes #364856.
	  Original commit message from CVS:
	  * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps), (gst_pngenc_chain):
	  * ext/libpng/gstpngenc.h:
	  Fix strides. Fixes #364856.
	  Cleanup capsnego.
	  Set caps on outgoing buffers.

2006-10-18 17:06:21 +0000  Ville Syrjala <ville.syrjala@movial.fi>

	  gst/rtp/: Add static payload numbers in addition to the dynamic ones.
	  Original commit message from CVS:
	  Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
	  * gst/rtp/gstrtpgsmpay.c:
	  * gst/rtp/gstrtph263pay.c:
	  * gst/rtp/gstrtpmpapay.c:
	  * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
	  (gst_rtp_pcma_pay_handle_buffer):
	  * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush):
	  Add static payload numbers in addition to the dynamic ones.
	  Fixes #361639.

2006-10-18 16:18:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Reuse already existing enum for lower transport.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
	  (gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
	  (gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri):
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create):
	  * gst/rtsp/rtspdefs.h:
	  * gst/rtsp/rtspurl.c: (rtsp_url_parse):
	  * gst/rtsp/rtspurl.h:
	  Reuse already existing enum for lower transport.
	  Add rtspt and rtspu protocols.
	  Send redirect to rtspt when udp times out.

2006-10-18 14:00:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Fix seeking some more, mostly for speed changes.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_data):
	  Fix seeking some more, mostly for speed changes.

2006-10-18 11:28:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery: fix Fredrik's e-mail address
	  Original commit message from CVS:
	  ChangeLog surgery: fix Fredrik's e-mail address

2006-10-18 11:04:09 +0000  Fredrik Persson <frepe@broadband.net>

	  sys/v4l2/gstv4l2tuner.*: Fix _set_channel(): remove useless g_object_notify() for "channel" property that doesn't exi...
	  Original commit message from CVS:
	  Patch by: Fredrik Persson  <frepe at broadband net>
	  * sys/v4l2/gstv4l2tuner.c:
	  * sys/v4l2/gstv4l2tuner.h:
	  Fix _set_channel(): remove useless g_object_notify() for "channel"
	  property that doesn't exist any longer and therefore now also
	  useless redirect (#338818).

2006-10-17 15:16:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Activate pads before adding them to running element.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_set_wp_config):
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_create_src_pad):
	  * gst/nuvdemux/gstnuvdemux.c: (gst_nuv_demux_create_pads):
	  * tests/check/elements/wavpackparse.c: (wavpackparse_found_pad):
	  Activate pads before adding them to running element.

2006-10-17 14:57:17 +0000  Josep Torra Valles <josep@fluendo.com>

	  gst/qtdemux/qtdemux.c: Make compile with Forte compiler, mostly don't do pointer arithmetic with void pointers (#3626...
	  Original commit message from CVS:
	  Patch by: Josep Torra Valles  <josep at fluendo com>
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
	  (next_entry_size), (qtdemux_inflate), (qtdemux_parse_moov),
	  (qtdemux_parse_tree), (qtdemux_parse_trak), (qtdemux_tag_add_str),
	  (qtdemux_tag_add_num), (qtdemux_tag_add_date),
	  (qtdemux_tag_add_gnre):
	  Make compile with Forte compiler, mostly don't do pointer arithmetic
	  with void pointers (#362626).

2006-10-17 14:37:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/oss/gstosssink.c: Some drivers do not support unsetting the non-blocking flag once the device is opened. In those...
	  Original commit message from CVS:
	  * sys/oss/gstosssink.c: (gst_oss_sink_prepare):
	  Some drivers do not support unsetting the non-blocking flag once the
	  device is opened. In those cases, close/open the device in
	  non-blocking mode. Fixes #362673.

2006-10-17 13:44:14 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/: dear stefan, framespersecond is not frameperiod, reverting but adding comment
	  Original commit message from CVS:
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
	  (gst_v4l2src_get_fps):
	  dear stefan, framespersecond is not frameperiod, reverting but adding
	  comment

2006-10-17 11:28:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/: Numerator is numerator and denominator is denominator. Say that aloud 5 times and retry after next beer.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
	  (gst_v4l2src_get_fps):
	  Numerator is numerator and denominator is denominator. Say that aloud
	  5 times and retry after next beer.

2006-10-17 10:59:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.*: Avoid void pointer usage, better use guint8 * instead.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_moov), (qtdemux_parse),
	  (qtdemux_node_dump_foreach), (qtdemux_dump_mvhd),
	  (qtdemux_dump_tkhd), (qtdemux_dump_elst), (qtdemux_dump_mdhd),
	  (qtdemux_dump_hdlr), (qtdemux_dump_vmhd), (qtdemux_dump_dref),
	  (qtdemux_dump_stsd), (qtdemux_dump_stts), (qtdemux_dump_stss),
	  (qtdemux_dump_stsc), (qtdemux_dump_stsz), (qtdemux_dump_stco),
	  (qtdemux_dump_co64), (qtdemux_dump_dcom), (qtdemux_dump_cmvd),
	  (qtdemux_dump_unknown), (qtdemux_tree_get_child_by_type),
	  (qtdemux_tree_get_sibling_by_type):
	  * gst/qtdemux/qtdemux.h:
	  Avoid void pointer usage, better use guint8 * instead.

2006-10-16 18:22:47 +0000  Josep Torra Valles <josep@fluendo.com>

	  Fix a bunch of problems discovered by the Forte compiler, mostly type mixups and pointer arithmetics with void pointe...
	  Original commit message from CVS:
	  Patch by: Josep Torra Valles  <josep at fluendo com>
	  * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	  * ext/esd/esdsink.c: (gst_esdsink_write):
	  * ext/flac/gstflacdec.c: (gst_flac_dec_length),
	  (gst_flac_dec_read_seekable), (gst_flac_dec_chain),
	  (gst_flac_dec_send_newsegment):
	  * ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback),
	  (gst_flac_enc_tell_callback):
	  * ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode),
	  (smokecodec_parse_header), (smokecodec_decode):
	  * gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index):
	  * gst/debug/efence.c: (gst_fenced_buffer_alloc):
	  * gst/goom/Makefile.am:
	  * gst/goom/gstgoom.c:
	  * gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward):
	  * gst/rtsp/gstrtspsrc.c:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	  * gst/udp/gstudpsink.c:
	  * gst/udp/gstudpsrc.c:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_change_state):
	  * sys/sunaudio/gstsunaudiomixertrack.h:
	  Fix a bunch of problems discovered by the Forte compiler, mostly type
	  mixups and pointer arithmetics with void pointers. Fixes #362603.

2006-10-13 14:45:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.c: Round up not allowed bitrates to the next higher allowed one (Closes: #361140).
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_set_property):
	  Round up not allowed bitrates to the next higher allowed one
	  (Closes: #361140).

2006-10-13 14:19:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add docs for lame and lame to docs. Specify allowed bitrates in the properties description (#361140). Canonicalise ob...
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-ugly-plugins-sections.txt:
	  * ext/lame/gstlame.c: (gst_lame_class_init):
	  * ext/lame/gstlame.h:
	  Add docs for lame and lame to docs. Specify allowed bitrates
	  in the properties description (#361140). Canonicalise object
	  property names (ie. use hyphen instead of underscore).
	  * docs/plugins/inspect/plugin-a52dec.xml:
	  * docs/plugins/inspect/plugin-amrnb.xml:
	  * docs/plugins/inspect/plugin-asf.xml:
	  * docs/plugins/inspect/plugin-dvdlpcmdec.xml:
	  * docs/plugins/inspect/plugin-dvdread.xml:
	  * docs/plugins/inspect/plugin-dvdsub.xml:
	  * docs/plugins/inspect/plugin-iec958.xml:
	  * docs/plugins/inspect/plugin-lame.xml:
	  * docs/plugins/inspect/plugin-mad.xml:
	  * docs/plugins/inspect/plugin-mpeg2dec.xml:
	  * docs/plugins/inspect/plugin-mpegaudioparse.xml:
	  * docs/plugins/inspect/plugin-mpegstream.xml:
	  * docs/plugins/inspect/plugin-siddec.xml:
	  Update version to CVS.

2006-10-13 10:00:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add i18n magic to lame plugin. Throw decent error message when we fail to setup the encoder (#361140, 361151); misc. ...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_setcaps),
	  (gst_lame_set_property), (gst_lame_get_property), (gst_lame_chain),
	  (plugin_init):
	  * po/POTFILES.in:
	  Add i18n magic to lame plugin. Throw decent error message when we
	  fail to setup the encoder (#361140, 361151); misc. minor clean-ups.

2006-10-12 19:02:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/speex/: Miscellaneous clean-ups, among other things: speexenc => enc to enhance code readability; change speexenc...
	  Original commit message from CVS:
	  * ext/speex/gstspeex.c: (plugin_init):
	  * ext/speex/gstspeexenc.c: (gst_speex_enc_get_formats),
	  (gst_speex_enc_setup_interfaces), (gst_speex_enc_base_init),
	  (gst_speex_enc_class_init), (gst_speex_enc_finalize),
	  (gst_speex_enc_sink_setcaps), (gst_speex_enc_convert_src),
	  (gst_speex_enc_convert_sink), (gst_speex_enc_get_query_types),
	  (gst_speex_enc_src_query), (gst_speex_enc_sink_query),
	  (gst_speex_enc_init), (gst_speex_enc_create_metadata_buffer),
	  (gst_speex_enc_set_last_msg), (gst_speex_enc_setup),
	  (gst_speex_enc_buffer_from_data), (gst_speex_enc_push_buffer),
	  (gst_speex_enc_set_header_on_caps), (gst_speex_enc_sinkevent),
	  (gst_speex_enc_chain), (gst_speex_enc_get_property),
	  (gst_speex_enc_set_property), (gst_speex_enc_change_state):
	  * ext/speex/gstspeexenc.h:
	  Miscellaneous clean-ups, among other things: speexenc => enc to
	  enhance code readability; change speexenc => speex_enc; in chain
	  function unref input buffer in case of error; take reference in
	  event function; use boilerplate macro; use gst_pad_query_peer_*
	  convenience functions.

2006-10-12 18:35:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/speex/gstspeexenc.c: Fix some mem leaks.
	  Original commit message from CVS:
	  * ext/speex/gstspeexenc.c: (gst_speexenc_finalize),
	  (gst_speexenc_set_last_msg), (gst_speexenc_setup),
	  (gst_speexenc_set_header_on_caps):
	  Fix some mem leaks.

2006-10-11 16:21:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/URLS: Added some other URL.
	  Original commit message from CVS:
	  * gst/rtsp/URLS:
	  Added some other URL.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp),
	  (gst_rtspsrc_handle_request), (gst_rtspsrc_send),
	  (gst_rtspsrc_open), (gst_rtspsrc_play),
	  (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Work on fallback to TCP connection when the UDP socket times out.
	  Handler server requests, just reply with OK for now.
	  * gst/rtsp/rtspdefs.c: (rtsp_strresult):
	  * gst/rtsp/rtspdefs.h:
	  Added some more Real extension headers.
	  * gst/rtsp/rtspurl.c: (rtsp_url_parse):
	  Fix parsing of urls with a ':' that is not part of the hostname:port
	  part of the url.

2006-10-11 13:49:26 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Add some fourcc for DV format.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add some fourcc for DV format.

2006-10-11 13:24:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Activate pad before adding it to the already-running element.
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
	  * gst/icydemux/gsticydemux.c: (gst_icydemux_add_srcpad):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
	  Activate pad before adding it to the already-running element.
	  * tests/check/elements/icydemux.c: (icydemux_found_pad):
	  Activate newly-created pad too.

2006-10-11 08:34:14 +0000  Sebastien Cote <sebas642@yahoo.ca>

	  gst/udp/gstudpsrc.c: Fix some leaks in caps and uris. Fixes #361252.
	  Original commit message from CVS:
	  Patch by: Sebastien Cote <sebas642 at yahoo dot ca>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	  (gst_udpsrc_finalize), (gst_udpsrc_create), (gst_udpsrc_set_uri),
	  (gst_udpsrc_start):
	  Fix some leaks in caps and uris. Fixes #361252.

2006-10-10 18:54:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Printf format fixes.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (extract_initial_length_and_fourcc),
	  (gst_qtdemux_loop_state_header):
	  Printf format fixes.
	  * sys/dvb/gstdvbsrc.c:
	  Use "_stdint.h".

2006-10-10 09:57:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Reorganise some stuff.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
	  (gst_qtdemux_push_event), (gst_qtdemux_do_seek),
	  (gst_qtdemux_change_state), (extract_initial_length_and_fourcc),
	  (gst_qtdemux_loop_state_header), (gst_qtdemux_activate_segment),
	  (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop),
	  (gst_qtdemux_post_buffering), (gst_qtdemux_chain),
	  (gst_qtdemux_add_stream), (qtdemux_process_redirects),
	  (qtdemux_parse_tree), (qtdemux_parse_trak):
	  Reorganise some stuff.
	  Parse RTSP redirection URLS.

2006-10-10 08:29:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/Makefile.am: Fix copy'n'paste-o (spotted by Mark Nauwelaerts, #341489).
	  Original commit message from CVS:
	  * gst/wavparse/Makefile.am:
	  Fix copy'n'paste-o (spotted by Mark Nauwelaerts, #341489).

2006-10-09 07:01:19 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/v4l2/gstv4l2xoverlay.*: Fix build as per the patch in #338818 comment 36.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2xoverlay.c:
	  * sys/v4l2/gstv4l2xoverlay.h:
	  Fix build as per the patch in #338818 comment 36.

2006-10-08 20:05:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	  inspect updates
	  Original commit message from CVS:
	  inspect updates

2006-10-07 21:15:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtsp/gstrtspsrc.c: Activate pads before adding them to the source.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
	  Activate pads before adding them to the source.

2006-10-07 11:37:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Add/update docs stuff.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.prerequisites:
	  * docs/plugins/inspect/plugin-dtsdec.xml:
	  * docs/plugins/inspect/plugin-mms.xml:
	  * docs/plugins/inspect/plugin-mpeg2enc.xml:
	  * docs/plugins/inspect/plugin-neon.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-spcdec.xml:
	  * docs/plugins/inspect/plugin-swfdec.xml:
	  * docs/plugins/inspect/plugin-videocrop.xml:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  Add/update docs stuff.

2006-10-06 17:00:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Activate pads before adding.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads), (gst_dvdemux_chain):
	  * gst/auparse/gstauparse.c: (gst_au_parse_add_srcpad):
	  Activate pads before adding.

2006-10-06 16:03:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/multipart/multipartdemux.c: Activate pads before adding.
	  Original commit message from CVS:
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
	  (gst_multipart_find_pad_by_mime):
	  Activate pads before adding.
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	  BOILERPLATE sets parent_class for us.

2006-10-06 15:56:01 +0000  René Stadler <mail@renestadler.de>

	  Add ReplayGain analysis element (#357069).
	  Original commit message from CVS:
	  Patch by: René Stadler  <mail at renestadler de>
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * gst/replaygain/Makefile.am:
	  * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_base_init),
	  (gst_rg_analysis_class_init), (gst_rg_analysis_init),
	  (gst_rg_analysis_set_property), (gst_rg_analysis_get_property),
	  (gst_rg_analysis_start), (gst_rg_analysis_set_caps),
	  (gst_rg_analysis_transform_ip), (gst_rg_analysis_event),
	  (gst_rg_analysis_stop), (gst_rg_analysis_handle_tags),
	  (gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result),
	  (gst_rg_analysis_album_result), (plugin_init):
	  * gst/replaygain/gstrganalysis.h:
	  * gst/replaygain/rganalysis.c: (yule_filter), (butter_filter),
	  (apply_filters), (reset_filters), (accumulator_add),
	  (accumulator_clear), (accumulator_result), (rg_analysis_new),
	  (rg_analysis_set_sample_rate), (rg_analysis_destroy),
	  (rg_analysis_analyze_mono_float),
	  (rg_analysis_analyze_stereo_float),
	  (rg_analysis_analyze_mono_int16),
	  (rg_analysis_analyze_stereo_int16), (rg_analysis_analyze),
	  (rg_analysis_track_result), (rg_analysis_album_result),
	  (rg_analysis_reset_album), (rg_analysis_reset):
	  * gst/replaygain/rganalysis.h:
	  Add ReplayGain analysis element (#357069).
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/rganalysis.c: (get_expected_gain),
	  (setup_rganalysis), (cleanup_rganalysis), (set_playing_state),
	  (send_eos_event), (send_tag_event), (poll_eos), (poll_tags),
	  (fail_unless_track_gain), (fail_unless_track_peak),
	  (fail_unless_album_gain), (fail_unless_album_peak),
	  (fail_if_track_tags), (fail_if_album_tags),
	  (fail_unless_num_tracks), (test_buffer_const_float_mono),
	  (test_buffer_const_float_stereo), (test_buffer_const_int16_mono),
	  (test_buffer_const_int16_stereo), (test_buffer_square_float_mono),
	  (test_buffer_square_float_stereo), (test_buffer_square_int16_mono),
	  (test_buffer_square_int16_stereo), (push_buffer), (GST_START_TEST),
	  (rganalysis_suite), (main):
	  Unit tests for the new replaygain element.

2006-10-06 15:49:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/faad/gstfaad.c: Some cleanups.
	  Original commit message from CVS:
	  * ext/faad/gstfaad.c: (gst_faad_setcaps), (gst_faad_chain),
	  (gst_faad_close_decoder):
	  Some cleanups.
	  Added some more debugging.
	  Don't ever ignore unlinked, we're not a demuxer.
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream):
	  Activate pad before adding it to the element.

2006-10-06 12:55:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try to share channels and udp ports.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
	  (gst_rtspsrc_class_init), (gst_rtspsrc_init),
	  (gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_alloc_udp_ports),
	  (gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
	  (gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_create_transports_string),
	  (gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
	  (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Rework how the transport string is constructed, try to share channels
	  and udp ports.
	  Make most of the stuff less dependant on RTP as we are also going to use
	  it for RDT.
	  Add support for transport specific session managers.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
	  Implement _flush().
	  * gst/rtsp/rtspdefs.c: (rtsp_strresult):
	  * gst/rtsp/rtspdefs.h:
	  Add generic error return code.
	  * gst/rtsp/rtspext.h:
	  Add support for pluggable tranport strings.
	  * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
	  (rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
	  (rtsp_ext_wms_get_context):
	  Detect WMServer and activate the extension.
	  * gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
	  (rtsp_transport_get_manager), (rtsp_transport_parse):
	  * gst/rtsp/rtsptransport.h:
	  Added methods to get mime/manager for certain transports.

2006-10-06 11:31:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/spectrum/gstspectrum.c: Fix mem leak, avoid unnecessary memcpy.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip):
	  Fix mem leak, avoid unnecessary memcpy.

2006-10-06 02:29:35 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.c: Removed cruft code that was just commented out. Removed some obsolete debug logs statements.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_init),
	  (gst_spectrum_transform_ip):
	  Removed cruft code that was just commented out. Removed some obsolete
	  debug logs statements.

2006-10-05 18:14:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Another batch of printf format fixes.
	  Original commit message from CVS:
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_chain):
	  * ext/musicbrainz/gsttrm.c: (gst_trm_setcaps):
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps):
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_chain), (qtdemux_parse),
	  (qtdemux_parse_trak):
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip):
	  Another batch of printf format fixes.

2006-10-05 16:37:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Printf format fixes.
	  Original commit message from CVS:
	  * ext/cairo/gsttimeoverlay.c:
	  (gst_cairo_time_overlay_update_font_height):
	  * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps):
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data):
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	  * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
	  * ext/libpng/gstpngdec.c: (user_endrow_callback):
	  * gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
	  (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_stream_data):
	  * gst/cutter/gstcutter.c: (gst_cutter_chain):
	  * gst/debug/efence.c: (gst_efence_buffer_alloc),
	  (gst_fenced_buffer_copy):
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
	  (gst_rtspsrc_handle_message):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  * sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
	  Printf format fixes.

2006-10-04 22:37:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videocrop/gstvideocrop.*: Handle packed YUV formats (UYVY, YUY2, YUYV) separately; also, fix passthrough mode; la...
	  Original commit message from CVS:
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init),
	  (gst_video_crop_init),
	  (gst_video_crop_get_image_details_from_caps),
	  (gst_video_crop_transform_packed_complex),
	  (gst_video_crop_transform_packed_simple),
	  (gst_video_crop_transform), (gst_video_crop_transform_caps),
	  (gst_video_crop_set_caps),
	  (gst_videocrop_clear_negotiated_caps_locked),
	  (gst_video_crop_set_property):
	  * gst/videocrop/gstvideocrop.h:
	  Handle packed YUV formats (UYVY, YUY2, YUYV) separately; also, fix
	  passthrough mode; lastly, clear negotiated basetransform caps when
	  the cropping changes in order to force renegotiation.

2006-10-04 20:05:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/: Visual test for videocrop, shows that packed yuv doesn't work right yet. --with-ffmpegcolorspace option...
	  Original commit message from CVS:
	  * tests/icles/.cvsignore:
	  * tests/icles/Makefile.am:
	  * tests/icles/videocrop-test.c: (quit_mainloop), (tick_cb),
	  (test_with_caps), (video_crop_get_test_caps), (main):
	  Visual test for videocrop, shows that packed yuv doesn't work right
	  yet. --with-ffmpegcolorspace option doesn't work yet for unknown
	  reasons (another basetransform issue?)

2006-10-04 17:53:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/Makefile.am: Dist new .h file too.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  Dist new .h file too.

2006-10-04 17:24:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Factor out extension in separate module.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
	  (gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	  (gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
	  (gst_rtspsrc_parse_rtpmap),
	  (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
	  (gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	  (gst_rtspsrc_play), (gst_rtspsrc_handle_message):
	  * gst/rtsp/gstrtspsrc.h:
	  * gst/rtsp/rtspdefs.c: (rtsp_strresult):
	  * gst/rtsp/rtspdefs.h:
	  * gst/rtsp/rtspext.h:
	  * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
	  (rtsp_ext_wms_get_context):
	  * gst/rtsp/rtspextwms.h:
	  * gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
	  (rtsp_transport_parse):
	  * gst/rtsp/rtsptransport.h:
	  Factor out extension in separate module.
	  Fix getcaps to filter against the padtemplate.
	  Use Content-Base if the server gives one.
	  Rework the transport parsing a bit for future extensions.
	  Added some Real Header field definitions.

2006-10-04 10:29:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  docs/plugins/: added v4l2 stubs
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  added v4l2 stubs
	  * gst-plugins-good.spec.in:
	  add v4l2

2006-10-04 10:24:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/apetag/gstapedemux.c: Extract disc/album/medium number and count and try harder to extract track number/count.
	  Original commit message from CVS:
	  * gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	  Extract disc/album/medium number and count and try harder
	  to extract track number/count.

2006-10-03 18:36:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/icles/.gitignore:
	  moap ignore
	  Original commit message from CVS:
	  moap ignore

2006-10-03 18:35:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/icles/Makefile.am:
	  add icle for v4l2
	  Original commit message from CVS:
	  add icle for v4l2

2006-10-03 18:15:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  add build stuff for v4l2, needs --enable-experimental until the last bits are resolved
	  Original commit message from CVS:
	  * configure.ac:
	  * sys/Makefile.am:
	  add build stuff for v4l2, needs --enable-experimental until
	  the last bits are resolved

2006-10-03 13:47:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* sys/v4l2/gstv4l2object.c:
	  comment out the notifies for removed properties
	  Original commit message from CVS:
	  comment out the notifies for removed properties

2006-10-03 13:30:48 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  sys/v4l2/gstv4l2object.c: comment out the properties that are already part of the tuner interface.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2object.c:
	  (gst_v4l2_object_install_properties_helper):
	  comment out the properties that are already part of the tuner
	  interface.

2006-10-03 13:18:59 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/v4l2/gstv4l2src.c: Improve docs.
	  Original commit message from CVS:
	  2006-10-03  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/v4l2/gstv4l2src.c:
	  Improve docs.

2006-10-02 16:14:06 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  stop removing gdkpixbuf plugin from package
	  Original commit message from CVS:
	  stop removing gdkpixbuf plugin from package

2006-09-29 15:39:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Disable autodetect test temporarily, so that the build bots update -bad and the ranks of unr...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Disable autodetect test temporarily, so that the build bots
	  update -bad and the ranks of unreliable video sinks in there.
	  * tests/check/elements/autodetect.c: (GST_START_TEST):
	  Skip test if no usable videosink is found.

2006-09-29 15:37:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/URLS: Add some more URLs.
	  Original commit message from CVS:
	  * gst/rtsp/URLS:
	  Add some more URLs.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	  (gst_rtspsrc_init), (gst_rtspsrc_finalize),
	  (gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	  (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved),
	  (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	  (gst_rtspsrc_loop), (gst_rtspsrc_send),
	  (gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	  (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	  (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Add timeout property to control UDP timeouts.
	  Fix error messages.
	  Also start a loop function when operating in UDP mode so that we can
	  do some more stuff async.
	  Handle element messages from udpsrc to detect timeouts. If a timeout
	  happens we currently generate an error.
	  API: rtspsrc::timeout property.
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	  (gst_udpsrc_create):
	  Really implement the timeout in microseconds and not milliseconds.

2006-09-29 11:09:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.*: Added property to post a message on timeout.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	  (gst_udpsrc_create), (gst_udpsrc_set_property),
	  (gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop):
	  * gst/udp/gstudpsrc.h:
	  Added property to post a message on timeout.
	  Updated docs.
	  When restarting the select, initialize the fdsets again.
	  Init control sockets so we don't accidentally close a random socket.
	  API: GstUDPSrc::timeout property

2006-09-29 08:15:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Fix flag registration.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
	  Fix flag registration.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	  Reading 0 also means 'no more commands'

2006-09-29 08:09:24 +0000  Antoine Tremblay <hexa00@gmail.com>

	  gst/udp/gstudpsrc.c: Fix possible infinite loop when shutting down, a read can also return 0 to indicate no more mess...
	  Original commit message from CVS:
	  Patch by: Antoine Tremblay <hexa00 at gmail dot com>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Fix possible infinite loop when shutting down, a read can also return
	  0 to indicate no more messages are available. Fixes #358156.

2006-09-28 17:08:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/: Framerate can be 0/1 too.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_all_caps),
	  (gst_v4l2src_get_caps):
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  Framerate can be 0/1 too.
	  Init framerate to 0/1 before querying it so that we can detect
	  devices that don't know about a framerate.
	  Add some more debugging info.

2006-09-28 14:31:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Add support for 'yv12' fourcc.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add support for 'yv12' fourcc.

2006-09-27 17:47:57 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* tests/icles/v4l2src-test.c:
	  Removed set-undef-fps.
	  Original commit message from CVS:
	  Removed set-undef-fps.

2006-09-27 17:04:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/: Renamed some properties to match the tuner interface naming.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2object.c:
	  (gst_v4l2_object_install_properties_helper), (gst_v4l2_object_new),
	  (gst_v4l2_object_set_property_helper),
	  (gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	  (gst_v4l2src_create):
	  * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_contains_channel),
	  (gst_v4l2_tuner_list_channels),
	  (gst_v4l2_tuner_set_channel_and_notify),
	  (gst_v4l2_tuner_get_channel), (gst_v4l2_tuner_contains_norm),
	  (gst_v4l2_tuner_list_norms), (gst_v4l2_tuner_set_norm_and_notify),
	  (gst_v4l2_tuner_get_norm):
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	  (gst_v4l2_fill_lists), (gst_v4l2_empty_lists):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_get_fps):
	  Renamed some properties to match the tuner interface naming.

2006-09-27 16:14:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Small cleanups.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_set_property_helper),
	  (gst_v4l2_set_defaults):
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read),
	  (gst_v4l2src_create):
	  * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_open):
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	  (gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_set_norm),
	  (gst_v4l2_get_frequency), (gst_v4l2_set_frequency),
	  (gst_v4l2_signal_strength), (gst_v4l2_get_attribute),
	  (gst_v4l2_set_attribute), (gst_v4l2_get_input),
	  (gst_v4l2_set_input):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_grab_frame), (gst_v4l2src_get_capture),
	  (gst_v4l2src_set_capture), (gst_v4l2src_capture_init),
	  (gst_v4l2src_capture_start), (gst_v4l2src_capture_stop),
	  (gst_v4l2src_buffer_new):
	  * tests/icles/v4l2src-test.c: (my_bus_callback), (main):
	  Small cleanups.
	  Fix error messages.
	  Use locks when getting timestamps.
	  Fix leaks in test.
	  Add licensing header to tests.

2006-09-27 15:14:07 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	* tests/icles/v4l2src-test.c:
	  Some cleanups and comments.
	  Original commit message from CVS:
	  Some cleanups and comments.

2006-09-27 13:41:35 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add audiofx plugin
	  Original commit message from CVS:
	  add audiofx plugin

2006-09-26 14:17:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Add v4l2 plugin to the docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  Add v4l2 plugin to the docs.
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read),
	  (gst_v4l2src_get_mmap), (gst_v4l2src_create):
	  * sys/v4l2/gstv4l2src.h:
	  * sys/v4l2/gstv4l2vidorient.c:
	  Fix docs.
	  Remove some more externs.

2006-09-26 13:18:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/Makefile.am: Fix makefile, list libs in stack order.
	  Original commit message from CVS:
	  * sys/v4l2/Makefile.am:
	  Fix makefile, list libs in stack order.
	  * sys/v4l2/gstv4l2colorbalance.c:
	  * sys/v4l2/gstv4l2colorbalance.h:
	  * sys/v4l2/gstv4l2object.c: (gst_v4l2_device_get_type),
	  (gst_v4l2_object_install_properties_helper):
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read),
	  (gst_v4l2src_get_mmap), (gst_v4l2src_create):
	  * sys/v4l2/gstv4l2src.h:
	  * sys/v4l2/gstv4l2tuner.h:
	  * sys/v4l2/gstv4l2vidorient.h:
	  * sys/v4l2/gstv4l2xoverlay.h:
	  * sys/v4l2/v4l2_calls.h:
	  * sys/v4l2/v4l2src_calls.h:
	  Fix coding style:
	  - Remove extern from functions.
	  - Fix header indentation.
	  Fix Flags, add defaults for properties.
	  Remove unused enums.
	  Fix TOO_LAZY in error messages.

2006-09-26 11:06:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l2/: Fix pass at code cleanups, move errors cases out of the normal flow for additional code clarity.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2object.c: (gst_v4l2_class_probe_devices),
	  (gst_v4l2_probe_needs_probe),
	  (gst_v4l2_object_install_properties_helper), (gst_v4l2_object_new),
	  (gst_v4l2_object_destroy), (gst_v4l2_object_set_property_helper),
	  (gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults),
	  (gst_v4l2_object_start), (gst_v4l2_object_stop):
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	  (gst_v4l2src_init), (gst_v4l2src_dispose),
	  (gst_v4l2src_set_property), (gst_v4l2src_get_property),
	  (gst_v4l2src_fixate), (gst_v4l2src_get_caps),
	  (gst_v4l2src_set_caps), (gst_v4l2src_get_read),
	  (gst_v4l2src_get_mmap), (gst_v4l2src_create):
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	  (gst_v4l2_open), (gst_v4l2_close), (gst_v4l2_get_norm),
	  (gst_v4l2_set_norm), (gst_v4l2_get_frequency),
	  (gst_v4l2_set_frequency), (gst_v4l2_signal_strength),
	  (gst_v4l2_get_attribute), (gst_v4l2_set_attribute),
	  (gst_v4l2_get_input), (gst_v4l2_set_input):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
	  (gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
	  (gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
	  (gst_v4l2src_capture_stop), (gst_v4l2src_capture_deinit),
	  (gst_v4l2src_get_size_limits), (gst_v4l2src_set_fps),
	  (gst_v4l2src_get_fps), (gst_v4l2src_buffer_finalize),
	  (gst_v4l2src_buffer_new):
	  Fix pass at code cleanups, move errors cases out of the normal
	  flow for additional code clarity.

2006-09-25 13:55:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/autodetect/: Small cleanups. don't try to set "sync" property when it is not available.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
	  (gst_auto_audio_sink_find_best):
	  * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
	  Small cleanups.
	  don't try to set "sync" property when it is not available.

2006-09-25 11:47:42 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/: Include stdlib.h in some more places, makes things compile with uClibc and -Werror (#357592).
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/alpha/gstalpha.c:
	  * gst/rtp/gstrtpamrdepay.c:
	  * gst/rtsp/gstrtspsrc.c:
	  * gst/udp/gstudpsrc.c:
	  * gst/videomixer/videomixer.c:
	  Include stdlib.h in some more places, makes things compile
	  with uClibc and -Werror (#357592).

2006-09-25 09:15:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/gstjpegdec.c: our code should handle that fine. Some of the buttons on the apple trailer site are apparently...
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c:
	  Set minimum height to 8 (from 16), our code should handle
	  that fine. Some of the buttons on the apple trailer site
	  are apparently only 15 pixels high (see #357470).

2006-09-23 15:31:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Improve error reporting.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop), (gst_rtspsrc_send),
	  (gst_rtspsrc_open):
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	  (rtsp_connection_connect), (rtsp_connection_read), (read_body),
	  (rtsp_connection_receive):
	  * gst/rtsp/rtspdefs.c: (rtsp_strresult):
	  * gst/rtsp/rtspdefs.h:
	  Improve error reporting.

2006-09-23 15:30:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Fix klass typos.
	  Original commit message from CVS:
	  * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
	  * gst/rtp/gstrtpdepay.c:
	  * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
	  * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
	  (gst_rtp_mp2t_depay_plugin_init):
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
	  * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
	  * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
	  * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
	  * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
	  * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
	  Fix klass typos.
	  Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.

2006-09-22 17:53:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Need  -base CVS for gst_base_rtp_depayload_push_ts().
	  Original commit message from CVS:
	  * configure.ac:
	  Need  -base CVS for gst_base_rtp_depayload_push_ts().

2006-09-22 17:22:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Don't check for a tag that is never there and check if we read the correct tag. Fixes seeking ...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
	  Don't check for a tag that is never there and check if we read the
	  correct tag. Fixes seeking again.
	  We must post an error when all pads are unlinked.

2006-09-22 15:15:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: More fixage, set endoder-params correctly in the payloader.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
	  (gst_rtp_vorbis_pay_reset_packet),
	  (gst_rtp_vorbis_pay_init_packet),
	  (gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
	  (gst_rtp_vorbis_pay_handle_buffer):
	  More fixage, set endoder-params correctly in the payloader.

2006-09-22 12:12:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/autodetect/: Make static pad templates static to appease valgrind's leak detector.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_base_init):
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_base_init):
	  Make static pad templates static to appease valgrind's leak
	  detector.
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/autodetect.c: (GST_START_TEST),
	  (autodetect_suite):
	  Add simple test for the ghostpad lockup on shutdown fixed in core
	  CVS (audio bit disabled because it would need dozens of alsa
	  suppressions and I'm too lazy to add those now).

2006-09-22 12:08:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Small cleanups.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
	  Small cleanups.
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
	  (gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
	  (gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
	  (gst_rtp_vorbis_depay_process),
	  (gst_rtp_vorbis_depay_set_property),
	  (gst_rtp_vorbis_depay_get_property),
	  (gst_rtp_vorbis_depay_change_state),
	  (gst_rtp_vorbis_depay_plugin_init):
	  * gst/rtp/gstrtpvorbisdepay.h:
	  * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
	  (gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
	  (gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
	  (gst_rtp_vorbis_pay_flush_packet),
	  (gst_rtp_vorbis_pay_append_buffer),
	  (gst_rtp_vorbis_pay_handle_buffer),
	  (gst_rtp_vorbis_pay_plugin_init):
	  * gst/rtp/gstrtpvorbispay.h:
	  Add experimental vorbis pay and depayloaders.

2006-09-21 13:33:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4gpay.c: Fix profile-level-id parsing and setup.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):
	  Fix profile-level-id parsing and setup.

2006-09-21 09:50:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/: Update README, simple cleanup.
	  Original commit message from CVS:
	  * gst/udp/README:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
	  Update README, simple cleanup.

2006-09-21 09:35:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/README: Update README with some examples.
	  Original commit message from CVS:
	  * gst/rtp/README:
	  Update README with some examples.
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
	  (gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
	  (gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
	  (gst_rtp_mp4g_pay_setcaps):
	  * gst/rtp/gstrtpmp4gpay.h:
	  Make optional RTP parameters of type STRING, as required by the
	  application/x-rtp caps specification.

2006-09-20 19:37:45 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst/rtp/: Correctly calculate size of each H263+ RTP buffer taking into account MTU and
	  Original commit message from CVS:
	  * gst/rtp/gstrtph263pdepay.c:
	  * gst/rtp/gstrtph263ppay.c:
	  Correctly calculate size of each H263+ RTP buffer taking into account MTU and
	  RTP header.

2006-09-20 16:41:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/Makefile.am: And makefile too.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  And makefile too.

2006-09-20 16:09:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added preliminary ASF depayloader.
	  Original commit message from CVS:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpasfdepay.c: (gst_rtp_asf_depay_base_init),
	  (gst_rtp_asf_depay_class_init), (gst_rtp_asf_depay_init),
	  (decode_base64), (gst_rtp_asf_depay_setcaps),
	  (gst_rtp_asf_depay_process), (gst_rtp_asf_depay_set_property),
	  (gst_rtp_asf_depay_get_property), (gst_rtp_asf_depay_change_state),
	  (gst_rtp_asf_depay_plugin_init):
	  * gst/rtp/gstrtpasfdepay.h:
	  Added preliminary ASF depayloader.
	  * gst/rtp/gstrtph264depay.c: (decode_base64):
	  Fix base64 decoding.

2006-09-20 16:06:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/URLS: Added some test URLS.
	  Original commit message from CVS:
	  * gst/rtsp/URLS:
	  Added some test URLS.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	  (gst_rtspsrc_loop), (gst_rtspsrc_open):
	  * gst/rtsp/gstrtspsrc.h:
	  When creating streams, give access to the complete SDP.
	  Fix some leaks.
	  Collect and merge global stream properties in stream caps.
	  Preliminary support for WMServer.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	  (rtsp_connection_connect), (rtsp_connection_read), (read_body),
	  (rtsp_connection_receive):
	  * gst/rtsp/rtspconnection.h:
	  Make connection interruptable.
	  Refactor to make it reconnectable.
	  Don't fail on short reads when reading data packets.
	  * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
	  (rtsp_url_get_port):
	  * gst/rtsp/rtspurl.h:
	  Add methods for getting/setting the port.
	  * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
	  (sdp_message_get_attribute_val), (sdp_media_get_attribute),
	  (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
	  (sdp_media_get_format), (sdp_parse_line),
	  (sdp_message_parse_buffer):
	  Fix headers.
	  Add methods for getting multiple attributes with the same name.
	  Increase buffer size when parsing.
	  Fix parsing of a=foo fields.
	  * gst/rtsp/test.c: (main):
	  Update to new connection API.
	  * gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
	  (rtsp_message_init_response), (rtsp_message_init_data),
	  (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
	  * gst/rtsp/rtspmessage.h:
	  * gst/rtsp/rtsptransport.c: (rtsp_transport_free):
	  * gst/rtsp/rtsptransport.h:
	  * gst/rtsp/sdp.h:
	  * gst/rtsp/sdpmessage.h:
	  * gst/rtsp/gstrtsp.c:
	  * gst/rtsp/gstrtsp.h:
	  * gst/rtsp/gstrtpdec.c:
	  * gst/rtsp/gstrtpdec.h:
	  * gst/rtsp/rtsp.h:
	  * gst/rtsp/rtspdefs.c:
	  * gst/rtsp/rtspdefs.h:
	  Dual licensed under MIT and LGPL now.

2006-09-19 17:25:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
	  (gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
	  (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	  (gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
	  (gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
	  (gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	  (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	  * gst/rtsp/gstrtspsrc.h:
	  Reorganize stream parsing and creation.
	  Detect container formats in interleaved mode.
	  Keep more state about the streams.
	  Assume a server also supports PLAY if it does not say.
	  Add unicast and interleaved properties to TCP transport requests to make
	  some servers happy (WMServer).
	  * gst/rtsp/sdpmessage.h:
	  Add some defines for the standard Bandwidth types.

2006-09-19 16:24:10 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* tests/icles/v4l2src-test.c:
	  Just a small fix to the app options.
	  Original commit message from CVS:
	  Just a small fix to the app options.

2006-09-19 13:08:35 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2vidorient.c:
	* sys/v4l2/gstv4l2vidorient.h:
	* tests/icles/v4l2src-test.c:
	  Add Video Orientation interface support to v4l2src.
	  Original commit message from CVS:
	  Add Video Orientation interface support to v4l2src.

2006-09-19 10:53:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/test.c: Fix build.
	  Original commit message from CVS:
	  * gst/rtsp/test.c: (main):
	  Fix build.

2006-09-19 10:14:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Add ms-gsm to the src template.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Add ms-gsm to the src template.

2006-09-18 17:37:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
	  (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
	  (gst_rtspsrc_pause), (gst_rtspsrc_change_state),
	  (gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
	  * gst/rtsp/gstrtspsrc.h:
	  Small cleanups, added documentation.
	  Try to clean up the requests and responses.
	  Refactor parsing the supported methods.
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_open),
	  (rtsp_connection_create), (rtsp_connection_send),
	  (parse_response_status), (parse_request_line),
	  (rtsp_connection_receive), (rtsp_connection_close),
	  (rtsp_connection_free):
	  * gst/rtsp/rtsptransport.c: (rtsp_transport_new),
	  (rtsp_transport_init), (rtsp_transport_parse),
	  (rtsp_transport_free):
	  * gst/rtsp/rtspurl.c: (rtsp_url_parse):
	  * gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
	  (sdp_message_clean), (sdp_message_free), (sdp_media_new),
	  (sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
	  Use g_return_val some more.
	  * gst/rtsp/rtspdefs.h:
	  Add more enum values to track initial states.
	  * gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
	  (rtsp_message_init_request), (rtsp_message_new_response),
	  (rtsp_message_init_response), (rtsp_message_init_data),
	  (rtsp_message_unset), (rtsp_message_free),
	  (rtsp_message_add_header), (rtsp_message_remove_header),
	  (rtsp_message_get_header), (rtsp_message_set_body),
	  (rtsp_message_take_body), (rtsp_message_get_body),
	  (rtsp_message_steal_body), (rtsp_message_dump):
	  * gst/rtsp/rtspmessage.h:
	  Reorder arguments, object goes as the first one.
	  Use g_return_val some more.

2006-09-18 15:36:14 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/v4l2src_calls.c:
	  Fix GST_BUFFER_DURATION.
	  Original commit message from CVS:
	  Fix GST_BUFFER_DURATION.

2006-09-18 14:00:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.*: Export sometimes source pad with correct caps on the template, create the ghostpad from the te...
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
	  (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
	  (gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
	  * gst/rtsp/gstrtspsrc.h:
	  Export sometimes source pad with correct caps on the template, create
	  the ghostpad from the template.
	  Remove RTCP template as we never expose RTCP.
	  Protect against invalid body size.
	  Avoid memcpy when creating the output buffer.
	  Properly post an error and send EOS when the loop function is shut down.

2006-09-18 11:29:12 +0000  Lutz Mueller <lutz@topfrose.de>

	  gst/rtsp/gstrtspsrc.*: Make sure we can never set an invalid location.
	  Original commit message from CVS:
	  Based on patch by: Lutz Mueller <lutz at topfrose dot de>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	  (gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
	  (gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
	  * gst/rtsp/gstrtspsrc.h:
	  Make sure we can never set an invalid location.
	  * gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
	  * gst/rtsp/rtspmessage.h:
	  Added _steal_body method for future use.
	  * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
	  Make freeing of NULL url return immediatly.

2006-09-18 10:42:52 +0000  Lutz Mueller <lutz@topfrose.de>

	  gst/rtsp/gstrtspsrc.*: Use boilerplate.
	  Original commit message from CVS:
	  Based on patch by: Lutz Mueller <lutz at topfrose dot de>
	  * gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
	  (gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
	  (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
	  (gst_rtspsrc_change_state):
	  * gst/rtsp/gstrtspsrc.h:
	  Use boilerplate.
	  Make rtspsrc subclass GstBin to make state changes easier.
	  Add Range header field on the PLAY request.

2006-09-18 08:59:17 +0000  Thijs Vermeir <thijs.vermeir@barco.com>

	  gst/rtsp/: Small cleanups. when multicast is selected as the transport, create UDP sources and connect to the multica...
	  Original commit message from CVS:
	  Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
	  (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	  (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
	  (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
	  * gst/rtsp/rtspconnection.c: (inet_aton):
	  Small cleanups.
	  when multicast is selected as the transport, create UDP sources and
	  connect to the multicast group.
	  Move parsing and setting of caps to a common place.
	  Fixes #349894.

2006-09-16 22:14:35 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  More G_OBJECT macro fixing.
	  Original commit message from CVS:
	  * ext/hermes/gsthermescolorspace.c:
	  * ext/ivorbis/vorbisfile.c:
	  * ext/lcs/gstcolorspace.c:
	  * ext/wavpack/gstwavpackenc.h:
	  * ext/xine/xineaudiodec.c:
	  * ext/xine/xineaudiosink.c:
	  * ext/xine/xineinput.c:
	  * gst/chart/gstchart.c:
	  * gst/equalizer/gstiirequalizer.c:
	  * gst/games/gstpuzzle.c:
	  * gst/librfb/gstrfbsrc.c:
	  * gst/mixmatrix/mixmatrix.c:
	  * gst/nsf/gstnsf.h:
	  * gst/vbidec/gstvbidec.c:
	  * gst/virtualdub/gstxsharpen.c:
	  More G_OBJECT macro fixing.

2006-09-16 21:57:29 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  More G_OBJECT macro fixing.
	  Original commit message from CVS:
	  * ext/flac/gstflactag.c:
	  * gst/alpha/gstalpha.c:
	  * gst/debug/breakmydata.c:
	  * gst/debug/negotiation.c:
	  * gst/debug/testplugin.c:
	  * gst/effectv/gstaging.c:
	  * gst/effectv/gstdice.c:
	  * gst/effectv/gstedge.c:
	  * gst/effectv/gstquark.c:
	  * gst/effectv/gstrev.c:
	  * gst/effectv/gstshagadelic.c:
	  * gst/effectv/gstvertigo.c:
	  * gst/effectv/gstwarp.c:
	  * gst/multipart/multipartdemux.c:
	  * gst/multipart/multipartmux.c:
	  * gst/videobox/gstvideobox.c:
	  * gst/videofilter/gstgamma.c:
	  * gst/videofilter/gstvideotemplate.c:
	  * gst/videomixer/videomixer.c:
	  * sys/sunaudio/gstsunaudiosrc.h:
	  More G_OBJECT macro fixing.

2006-09-16 14:30:59 +0000  Yves Lefebvre <ivanohe@abacom.com>

	  gst/avi/gstavimux.c: Correctly set the dwLength in strh.
	  Original commit message from CVS:
	  Patch by: Yves Lefebvre <ivanohe at abacom dot com>
	  * gst/avi/gstavimux.c: (gst_avi_mux_stop_file):
	  Correctly set the dwLength in strh.
	  With this patch, the file duration is now displayed correctly in window
	  media player and the AVI plays completely. Fixes #356147

2006-09-15 19:11:00 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	* tests/icles/v4l2src-test.c:
	  The test application and the plgind error messages has been improved.
	  Original commit message from CVS:
	  The test application and the plgind error messages has been improved.

2006-09-15 17:10:22 +0000  Darren Kenny <darren.kenny@sun.com>

	  sys/sunaudio/gstsunaudiomixerctrl.c: Set the output track as the MASTER so that the gnome-settings-daemon keybindings...
	  Original commit message from CVS:
	  Patch by: Darren Kenny <darren dot kenny at sun dot com>
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_build_list):
	  Set the output track as the MASTER so that the gnome-settings-daemon
	  keybindings for changing the volume using the keyboard works.
	  Fixes #356142.

2006-09-15 16:01:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/multipart/multipartdemux.c: Fix documentation, it is not possible to control the framerate of jpegdec using filte...
	  Original commit message from CVS:
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
	  Fix documentation, it is not possible to control the framerate of jpegdec
	  using filtered caps yet. Fixes #355210.
	  Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we
	  stop when there is an error.

2006-09-14 11:05:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Don't interpret a first buffer with an offset of NONE as 'from the middle of the stream', but only a first buff...
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Don't interpret a first buffer with an offset of NONE as
	  'from the middle of the stream', but only a first buffer
	  that has a valid buffer offset that's non-zero (see #345449).

2006-09-14 10:38:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/icydemux/gsticydemux.*: When we merge/collect multiple incoming buffers for typefinding purposes, keep an initial...
	  Original commit message from CVS:
	  * gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
	  (gst_icydemux_typefind_or_forward):
	  * gst/icydemux/gsticydemux.h:
	  When we merge/collect multiple incoming buffers for typefinding
	  purposes, keep an initial 0 offset on the first outgoing buffer
	  as well (otherwise id3demux won't work right). Fixes #345449.
	  Also Make buffer metadata writable before setting buffer caps.
	  * tests/check/elements/icydemux.c: (typefind_succeed),
	  (cleanup_icydemux), (push_data), (GST_START_TEST),
	  (icydemux_suite):
	  Small test case for the above.

2006-09-13 13:26:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: More code reuse and better logging in _peek_chunk(). Reintroduce check for chunk sizes before ...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk),
	  (gst_avi_demux_stream_index), (gst_avi_demux_sync),
	  (gst_avi_demux_stream_header_push),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	  (gst_avi_demux_loop):
	  More code reuse and better logging in _peek_chunk(). Reintroduce check
	  for chunk sizes before reading them (avoid oom). Better handling for
	  invalid chunksizes when streaming.

2006-09-12 20:18:55 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.c: Implements stop() to clear the adapter and event() to clear the adapter on FLUSH_STOP and...
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
	  (gst_spectrum_start), (gst_spectrum_stop), (gst_spectrum_event):
	  Implements stop() to clear the adapter and event() to clear the
	  adapter on FLUSH_STOP and EOS.

2006-09-11 20:38:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/level/gstlevel.*: Fix type mixup in level->interval (gdouble<->guint64). Spotted by
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_set_property):
	  * gst/level/gstlevel.h:
	  Fix type mixup in level->interval (gdouble<->guint64). Spotted by
	  René Stadler

2006-09-11 18:23:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.*: Fix type mixup in spectrum->interval (gdouble<->guint64). Spotted by
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_init),
	  (gst_spectrum_set_property):
	  * gst/spectrum/gstspectrum.h:
	  Fix type mixup in spectrum->interval (gdouble<->guint64). Spotted by
	  René Stadler

2006-09-11 18:02:39 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/demo-osssrc.c: Use more defines
	  Original commit message from CVS:
	  * gst/spectrum/demo-osssrc.c: (draw_spectrum), (main):
	  Use more defines
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_init),
	  (gst_spectrum_dispose), (gst_spectrum_set_caps),
	  (gst_spectrum_transform_ip):
	  * gst/spectrum/gstspectrum.h:
	  Apply some of the spectrum cleanup changes suggested in #348085.

2006-09-08 16:47:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump requirements of -base (videocrop test case needs this).
	  Original commit message from CVS:
	  * configure.ac:
	  Bump requirements of -base (videocrop test case needs this).
	  * gst/videocrop/gstvideocrop.c:
	  Document sloppy handling of subsampled chroma planes if
	  left/top cropping is an odd number.
	  * tests/check/elements/videocrop.c: (handoff_cb),
	  (videocrop_test_cropping_init_context),
	  (videocrop_test_cropping_deinit_context),
	  (videocrop_test_cropping), (check_1x1_buffer), (GST_START_TEST),
	  (videocrop_suite), (main):
	  Add another unit test that crops the input to 1x1 (and checks
	  that that pixel has the expected values in a number of formats).

2006-09-08 11:04:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videocrop/: Some quick tests indicate that it doesn't make a great deal of sense to use liboil here, at least not...
	  Original commit message from CVS:
	  * gst/videocrop/Makefile.am:
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init),
	  (gst_video_crop_transform_packed),
	  (gst_video_crop_transform_planar):
	  Some quick tests indicate that it doesn't make a great deal
	  of sense to use liboil here, at least not for the memcpy()s
	  we do, so remove liboil usage until there is clear evidence
	  it actually makes a positive difference somewhere.

2006-09-06 09:05:33 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Revert one change to fix streaming avi (adapter size != data size).
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
	  (gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
	  (gst_avi_demux_stream_data):
	  Revert one change to fix streaming avi (adapter size != data size).

2006-09-04 16:21:17 +0000  Frédéric Riss <frederic.riss@gmail.com>

	  gst/matroska/: Add support for VOBSUB subtitle tracks and zlib-compressed tracks. Make sure we start on a keyframe af...
	  Original commit message from CVS:
	  Patch by: Frédéric Riss  <frederic.riss at gmail dot com>
	  * gst/matroska/matroska-demux.c: (gst_matroska_track_free),
	  (gst_matroska_demux_reset),
	  (gst_matroska_demux_read_track_encodings),
	  (gst_matroska_demux_add_stream), (gst_matroska_decode_buffer),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_subtitle_caps):
	  * gst/matroska/matroska-ids.h:
	  Add support for VOBSUB subtitle tracks and zlib-compressed
	  tracks. Make sure we start on a keyframe after a seek. (#343348)

2006-09-04 15:06:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: not perfect yet though, needs some tweaking in flacdec; also, seeking could be better.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf),
	  (gst_matroska_demux_push_flac_codec_priv_data),
	  (gst_matroska_demux_push_xiph_codec_priv_data),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
	  * gst/matroska/matroska-ids.h:
	  Add basic FLAC support (#311586), not perfect yet though, needs some
	  tweaking in flacdec; also, seeking could be better.
	  Do better bounds checking when deserialising vorbis stream headers
	  to make sure we don't read beyond the end of the buffer on bad input.

2006-09-04 09:34:25 +0000  Alessandro Decina <alessandro@nnva.org>

	  ext/annodex/gstcmmldec.c: Seeking back in a file containing a CMML stream errors out if the seek goes back up to the ...
	  Original commit message from CVS:
	  Patch by: Alessandro Decina <alessandro at nnva dot org>
	  * ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain):
	  Seeking back in a file containing a CMML stream errors out if the seek
	  goes back up to the CMML headers. This is because after the seek the xml
	  processing instruction <?xml ...?> is submitted to the xml parser again,
	  which results in an error. The attached patch fixes the problem.
	  Fixes #353908.
	  * ext/annodex/gstcmmlenc.h:
	  Fix authors name.

2006-09-03 10:46:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/videocrop.c: More tests: check passthrough mode and caps transform in both directions with fixed...
	  Original commit message from CVS:
	  * tests/check/elements/videocrop.c: (handoff_cb),
	  (buffer_probe_cb), (test_caps_transform), (test_passthrough),
	  (notgst_value_list_get_nth_int), (videocrop_suite):
	  More tests: check passthrough mode and caps transform in
	  both directions with fixed values, ranges and lists.

2006-09-02 18:49:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Add videocrop to docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  Add videocrop to docs.
	  * gst/videocrop/Makefile.am:
	  * gst/videocrop/gstvideocrop.c:
	  * gst/videocrop/gstvideocrop.h:
	  Move boilerplate stuff and structures into a header file.
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/videocrop.c: (video_crop_get_test_caps),
	  (test_unit_sizes), (videocrop_test_cropping_init_context),
	  (videocrop_test_cropping_deinit_context),
	  (videocrop_test_cropping), (test_cropping), (videocrop_suite):
	  Add unit tests for videocrop.

2006-09-02 15:30:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Port/rewrite videocrop from scratch for GStreamer-0.10, and make it support all formats videoscale supports (#345653).
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/videocrop/Makefile.am:
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_base_init),
	  (gst_video_crop_class_init), (gst_video_crop_init),
	  (gst_video_crop_get_image_details_from_caps),
	  (gst_video_crop_get_unit_size), (gst_video_crop_transform_packed),
	  (gst_video_crop_transform_planar), (gst_video_crop_transform),
	  (gst_video_crop_transform_dimension),
	  (gst_video_crop_transform_dimension_value),
	  (gst_video_crop_transform_caps), (gst_video_crop_set_caps),
	  (gst_video_crop_set_property), (gst_video_crop_get_property),
	  (plugin_init):
	  Port/rewrite videocrop from scratch for GStreamer-0.10, and make
	  it support all formats videoscale supports (#345653).

2006-09-02 14:45:04 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/: Whitespace cleanups, dashify property-names.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2.c:
	  * sys/v4l2/gstv4l2colorbalance.c:
	  * sys/v4l2/gstv4l2object.c:
	  (gst_v4l2_object_install_properties_helper):
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init):
	  * sys/v4l2/gstv4l2src.h:
	  Whitespace cleanups, dashify property-names.

2006-09-02 14:28:55 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/: Cleanup error messages and unify header comments
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2.c:
	  * sys/v4l2/gstv4l2colorbalance.c:
	  * sys/v4l2/gstv4l2colorbalance.h:
	  * sys/v4l2/gstv4l2object.c:
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/gstv4l2src.c:
	  * sys/v4l2/gstv4l2src.h:
	  * sys/v4l2/gstv4l2tuner.c:
	  * sys/v4l2/gstv4l2tuner.h:
	  * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_open):
	  * sys/v4l2/gstv4l2xoverlay.h:
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	  (gst_v4l2_open):
	  * sys/v4l2/v4l2_calls.h:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_queue_frame),
	  (gst_v4l2src_capture_init):
	  * sys/v4l2/v4l2src_calls.h:
	  Cleanup error messages and unify header comments

2006-08-31 13:04:31 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Add missing GST_LIBS to the link flags
	  Original commit message from CVS:
	  * ext/lame/Makefile.am:
	  * ext/mpeg2dec/Makefile.am:
	  * gst/dvdlpcmdec/Makefile.am:
	  * gst/dvdsub/Makefile.am:
	  * gst/mpegaudioparse/Makefile.am:
	  Add missing GST_LIBS to the link flags

2006-08-30 18:01:52 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	  Another small fix to set_caps function.
	  Original commit message from CVS:
	  Another small fix to set_caps function.

2006-08-30 13:30:13 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	  Send new_segment in GST_FORMAT_TIME instead of in GST_FORMAT_BYTES.
	  Original commit message from CVS:
	  Send new_segment in GST_FORMAT_TIME instead of in GST_FORMAT_BYTES.

2006-08-30 11:36:06 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	  A small fix to set_caps function.
	  Original commit message from CVS:
	  A small fix to set_caps function.

2006-08-30 11:27:40 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Reset each streams last_flow to GST_FLOW_OK.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c:
	  (gst_qtdemux_do_seek):
	  Reset each streams last_flow to GST_FLOW_OK.
	  (gst_qtdemux_activate_segment):
	  Removing mystic modifications for good.

2006-08-30 11:07:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/qtdemux.c: put back 'segment start<=stop' change that was mystically reverted by the last commit
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	  (qtdemux_parse_tree):
	  put back 'segment start<=stop' change that was mystically reverted by
	  the last commit

2006-08-30 10:43:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/qtdemux/qtdemux.c: Fix the build for disabled debug
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	  (qtdemux_parse_tree):
	  Fix the build for disabled debug

2006-08-29 20:59:47 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  Fixed framerate negotiation.
	  Original commit message from CVS:
	  Fixed framerate negotiation.

2006-08-28 17:47:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Make sure segment start<=stop in weird quicktime files.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	  (gst_qtdemux_add_stream), (qtdemux_parse_trak),
	  (qtdemux_video_caps):
	  Make sure segment start<=stop in weird quicktime files.

2006-08-28 16:59:13 +0000  Andy Wingo <wingo@pobox.com>

	  ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle): New helper function to lessen the ifdefs.
	  Original commit message from CVS:
	  2006-08-28  Andy Wingo  <wingo@pobox.com>
	  * ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle):
	  New helper function to lessen the ifdefs.
	  (GST_INFO_OBJECT):
	  (gst_dv1394src_iso_receive): Use it.
	  (gst_dv1394src_create): Also use the control sockets in iec61883
	  mode.
	  (gst_dv1394src_start, gst_dv1394src_stop): Always use a separate
	  handle for AVC operations; fixes #348233.

2006-08-28 14:59:05 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/v4l2/v4l2_calls.c: add comments and more debug logging
	  Original commit message from CVS:
	  * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	  add comments and more debug logging

2006-08-27 17:14:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Rename again (audiofxgood -> audiofx).
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/inspect/plugin-audiofx.xml:
	  * docs/plugins/inspect/plugin-audiofxgood.xml:
	  * gst/audiofx/Makefile.am:
	  * gst/audiofx/audiofx.c:
	  * gst/audiofxgood/.cvsignore:
	  * gst/audiofxgood/Makefile.am:
	  * gst/audiofxgood/audiofx.c:
	  * gst/audiofxgood/audiopanorama.c:
	  * gst/audiofxgood/audiopanorama.h:
	  Rename again (audiofxgood -> audiofx).

2006-08-27 13:12:52 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Initialze variables.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_next_data_buffer),
	  (gst_avi_demux_stream_scan):
	  Initialze variables.

2006-08-25 16:21:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.*: More attempts to turn this into readable code.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	  (gst_avi_demux_init), (gst_avi_demux_finalize),
	  (gst_avi_demux_reset), (gst_avi_demux_index_last),
	  (gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time),
	  (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index),
	  (gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
	  (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
	  (gst_avi_demux_massage_index),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
	  (gst_avi_demux_chain), (gst_avi_demux_sink_activate),
	  (gst_avi_demux_change_state):
	  * gst/avi/gstavidemux.h:
	  More attempts to turn this into readable code.
	  Don't leak adapters.
	  Calculate duration according to index more efficiently.
	  Don't try to act like we drive the pipeline in chain mode.

2006-08-25 09:53:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/annodex/gstcmmlutils.c: Fix build.
	  Original commit message from CVS:
	  * ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt):
	  Fix build.

2006-08-25 09:42:43 +0000  Alessandro Decina <alessandro@nnva.org>

	  ext/annodex/gstannodex.c: Do some extra sanity checks.
	  Original commit message from CVS:
	  Patch by: Alessandro Decina <alessandro at nnva dot org>
	  * ext/annodex/gstannodex.c: (gst_annodex_granule_to_time):
	  Do some extra sanity checks.
	  Fixes #350340.
	  * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state),
	  (gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip),
	  (gst_cmml_enc_push_clip), (gst_cmml_enc_push):
	  Check if clip->start_time is valid before adding the clip to the
	  track list.
	  Reset enc->preamble going from PAUSED to READY.
	  Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is
	  only used for EOS.
	  Only post an error message if we were the one that created the fatal
	  GstFlowReturn value.
	  * ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt),
	  (gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip):
	  Parse the seconds field of the npt-sec time format using %llu rather than
	  %d and check that the value scaled by GST_SECOND doesn't overflow.
	  Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos.
	  Lookup a clip's track with clip->track rather than clip->id which
	  makes no sense.
	  Identify a clip by its track and start time and not its xml id.
	  do some more input checking and make sure we don't do undefined shifts.
	  * tests/check/elements/cmmldec.c: (setup_cmmldec),
	  (teardown_cmmldec), (check_output_buffer_is_equal), (push_data),
	  (cmml_tag_message_pop), (check_headers), (push_clip_full),
	  (push_clip), (push_empty_clip), (check_output_clip),
	  (GST_START_TEST), (cmmldec_suite):
	  * tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	  (teardown_cmmlenc), (check_output_buffer_is_equal), (push_data),
	  (check_headers), (push_clip), (check_clip_times), (check_clip),
	  (check_empty_clip), (GST_START_TEST), (cmmlenc_suite):
	  Added some more checks.

2006-08-24 19:00:22 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Make also the pan-property float (saves scaling and yields better resolution)
	  Original commit message from CVS:
	  * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_class_init),
	  (gst_audio_panorama_set_property),
	  (gst_audio_panorama_get_property),
	  (gst_audio_panorama_transform_m2s_int),
	  (gst_audio_panorama_transform_s2s_int),
	  (gst_audio_panorama_transform_m2s_float),
	  (gst_audio_panorama_transform_s2s_float):
	  * gst/audiofxgood/audiopanorama.h:
	  * tests/check/elements/audiopanorama.c: (GST_START_TEST):
	  Make also the pan-property float (saves scaling and yields better
	  resolution)

2006-08-24 18:23:14 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiofxgood/audiopanorama.c: ChangeLog surgery to add cymax's real name
	  Original commit message from CVS:
	  * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
	  (gst_audio_panorama_transform_m2s_float),
	  (gst_audio_panorama_transform_s2s_float):
	  ChangeLog surgery to add cymax's real name

2006-08-24 18:17:20 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiofxgood/audiopanorama.*: Added float support (thanks cymax)
	  Original commit message from CVS:
	  * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
	  (gst_audio_panorama_transform_m2s_int),
	  (gst_audio_panorama_transform_s2s_int),
	  (gst_audio_panorama_transform_m2s_float),
	  (gst_audio_panorama_transform_s2s_float),
	  (gst_audio_panorama_transform):
	  * gst/audiofxgood/audiopanorama.h:
	  Added float support (thanks cymax)

2006-08-24 14:16:55 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiofxgood/audiopanorama.c: Fix docs & debug category. Add Fixme for volume pan levels.
	  Original commit message from CVS:
	  * gst/audiofxgood/audiopanorama.c:
	  (gst_audio_panorama_transform_m2s):
	  Fix docs & debug category. Add Fixme for volume pan levels.

2006-08-24 13:51:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: unbreak AVI index handling, some more debug, remove an obsolete adapter_flush that caused stre...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
	  (gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
	  (gst_avi_demux_stream_header_pull),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	  (gst_avi_demux_chain):
	  unbreak AVI index handling, some more debug, remove an obsolete
	  adapter_flush that caused streaming to wander off in the wild

2006-08-24 11:21:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.*: Some more cleanups.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	  (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex),
	  (gst_avi_demux_parse_stream), (gst_avi_demux_parse_odml),
	  (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_stream_header_push),
	  (gst_avi_demux_stream_header_pull):
	  * gst/avi/gstavidemux.h:
	  Some more cleanups.
	  Fix totalFrames parsing in ODML.
	  Disable use of index for length calculation in case of ODML as this is
	  broken now.

2006-08-24 10:03:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.c: Use libgsttag helper function here too.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_update_metadata):
	  Use libgsttag helper function here too.

2006-08-24 09:24:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackdec.c: Post audio codec and average bitrate tags on bus (#344472).
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo at circular-chaos.org>
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_post_tags),
	  (gst_wavpack_dec_chain):
	  Post audio codec and average bitrate tags on bus (#344472).
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init),
	  (gst_wavpack_parse_src_query):
	  Forward queries in other formats (BYTE format in particular)
	  upstream; add Sebastian to authors.

2006-08-24 00:40:07 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  Fix set_caps to set width and height to the values the driver is really working with.
	  Original commit message from CVS:
	  Fix set_caps to set width and height to the values the driver is really working with.

2006-08-23 15:33:47 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.*: Initial streaming support for avidemux (fixes #336465)
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	  (gst_avi_demux_init), (gst_avi_demux_dispose),
	  (gst_avi_demux_reset), (gst_avi_demux_index_next),
	  (gst_avi_demux_index_entry_for_time), (gst_avi_demux_src_convert),
	  (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
	  (gst_avi_demux_peek_chunk_info), (gst_avi_demux_peek_chunk),
	  (gst_avi_demux_stream_init_push), (gst_avi_demux_stream_init_pull),
	  (gst_avi_demux_parse_subindex),
	  (gst_avi_demux_read_subindexes_push),
	  (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
	  (sort), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	  (gst_avi_demux_sync), (gst_avi_demux_peek_tag),
	  (gst_avi_demux_massage_index), (gst_avi_demux_stream_header_push),
	  (gst_avi_demux_stream_header_pull),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	  (push_tag_lists), (gst_avi_demux_loop), (gst_avi_demux_chain),
	  (gst_avi_demux_sink_activate), (gst_avi_demux_activate_push),
	  (gst_avi_demux_change_state):
	  * gst/avi/gstavidemux.h:
	  Initial streaming support for avidemux (fixes #336465)

2006-08-23 10:30:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/gstwavpackenc.c: Fix mem leak, send newsegment event on correction pad as well (#352476).
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
	  Fix mem leak, send newsegment event on correction pad
	  as well (#352476).
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
	  Restore original author (on Sebastian's request).
	  * tests/check/Makefile.am:
	  * tests/check/gst-plugins-bad.supp:
	  Add (so far empty) suppression file for -bad. Remove
	  wavpackenc test from VALGRIND_TO_FIX now that the leak
	  is fixed.

2006-08-23 09:22:07 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/: Add unit tests for wavpack elements (#352476).
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo at circular-chaos.org>
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/wavpackdec.c: (setup_wavpackdec),
	  (cleanup_wavpackdec), (GST_START_TEST), (wavpackdec_suite), (main):
	  * tests/check/elements/wavpackenc.c: (setup_wavpackenc),
	  (cleanup_wavpackenc), (GST_START_TEST), (wavpackenc_suite), (main):
	  * tests/check/elements/wavpackparse.c: (wavpackparse_found_pad),
	  (setup_wavpackparse), (cleanup_wavpackparse), (GST_START_TEST),
	  (wavpackparse_suite), (main):
	  Add unit tests for wavpack elements (#352476).

2006-08-23 08:52:50 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add docs for wavpack elements (#352476).
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo at circular-chaos.org>
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-wavpack.xml:
	  * ext/wavpack/gstwavpackdec.c:
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackenc.c:
	  * ext/wavpack/gstwavpackenc.h:
	  * ext/wavpack/gstwavpackparse.c:
	  * ext/wavpack/gstwavpackparse.h:
	  Add docs for wavpack elements (#352476).

2006-08-22 20:39:26 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  Fixed query size to work with drivers that uses intermediate step like "width * height" to find closest size.
	  Original commit message from CVS:
	  Fixed query size to work with drivers that uses intermediate step like "width * height" to find closest size.

2006-08-22 17:20:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/gst-plugins-good-plugins-docs.sgml: There is no taglibmux element ...
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  There is no taglibmux element ...
	  * gst/rtsp/gstrtspsrc.c:
	  Use '%' rather than '&perc;' in gtk-doc blurb, docs build
	  was complaining about unknown entity here.

2006-08-22 17:02:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.*: Mark DISCONT.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_do_seek), (gst_avi_demux_handle_seek),
	  (gst_avi_demux_process_next_entry):
	  * gst/avi/gstavidemux.h:
	  Mark DISCONT.
	  Remove old unused fields and reorder the struct a bit.

2006-08-22 16:45:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Small documentation updates.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play),
	  (gst_rtspsrc_pause):
	  * gst/rtsp/gstrtspsrc.h:
	  * sys/oss/gstosssink.c: (gst_oss_sink_open),
	  (gst_oss_sink_prepare), (gst_oss_sink_unprepare):
	  Small documentation updates.

2006-08-22 16:42:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.*: Precalc most of the duration query for each stream.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	  (gst_avi_demux_index_entry_for_time),
	  (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
	  (gst_avi_demux_stream_init), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
	  (gst_avi_demux_next_data_buffer),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_stream_header), (gst_avi_demux_do_seek),
	  (gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
	  (gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state):
	  * gst/avi/gstavidemux.h:
	  Precalc most of the duration query for each stream.
	  Make seeking more correct.
	  Use GstSegment to track position and duration.
	  Code cleanups and leak fixes.
	  Calculate correct total duration based on index length.

2006-08-22 13:53:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3v2frames.c: If strings in text fields are marked ISO8859-1, but contain valid UTF-8 already, then han...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
	  (parse_insert_string_field):
	  If strings in text fields are marked ISO8859-1, but contain
	  valid UTF-8 already, then handle them as UTF-8 and ignore
	  the encoding. (#351794)

2006-08-22 12:28:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.*: Make flac-in-ogg work (#352100).
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame),
	  (gst_flac_dec_write), (gst_flac_dec_loop),
	  (gst_flac_dec_sink_event), (gst_flac_dec_chain),
	  (gst_flac_dec_src_query):
	  * ext/flac/gstflacdec.h:
	  Make flac-in-ogg work (#352100).

2006-08-22 12:10:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/monoscope/gstmonoscope.c: Don't unref buffers of which we've already given away ownership to the adapter.
	  Original commit message from CVS:
	  * gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
	  Don't unref buffers of which we've already given away
	  ownership to the adapter.

2006-08-22 10:32:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/speex/gstspeexdec.c: Make metadata extraction actually work.
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_comments):
	  Make metadata extraction actually work.
	  * ext/speex/gstspeexenc.c: (gst_speexenc_base_init),
	  (gst_speexenc_init), (gst_speexenc_create_metadata_buffer),
	  (gst_speexenc_chain):
	  Fix metadata writing: replace old code which wrote completely
	  broken tags with libgsttag-based code. Plus miscellaneous
	  code cleanups (use static pad templates etc.) and a bunch
	  of leak fixes.

2006-08-21 19:34:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiopanorama/: die! die! die! you should never have been there
	  Original commit message from CVS:
	  * gst/audiopanorama/.cvsignore:
	  * gst/audiopanorama/Makefile.am:
	  * gst/audiopanorama/audiofx.c:
	  * gst/audiopanorama/audiopanorama.c:
	  * gst/audiopanorama/audiopanorama.h:
	  die! die! die! you should never have been there

2006-08-21 16:24:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Some more constification.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse),
	  (qtdemux_node_dump_foreach), (qtdemux_parse_trak),
	  (qtdemux_video_caps), (qtdemux_audio_caps):
	  Some more constification.
	  Fix some paletted data formats again.
	  Fix ulaw/alaw in qt.
	  Set correct caps for raw RGB.
	  Add support for yuv2, which is like Yuv2.
	  Add support for raw audio with the NONE fourcc, which is like raw.

2006-08-21 13:59:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/: More clean-ups: use shorter variable names to make code easier to read; prefix structures we define wit...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_init),
	  (gst_wavpack_enc_finalize), (gst_wavpack_enc_sink_set_caps),
	  (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_format_samples),
	  (gst_wavpack_enc_push_block), (gst_wavpack_enc_chain),
	  (gst_wavpack_enc_rewrite_first_block),
	  (gst_wavpack_enc_sink_event), (gst_wavpack_enc_change_state),
	  (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
	  * ext/wavpack/gstwavpackenc.h:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
	  (gst_wavpack_parse_src_query), (gst_wavpack_parse_src_event),
	  (gst_wavpack_parse_init), (gst_wavpack_parse_get_upstream_length),
	  (gst_wavpack_parse_loop):
	  More clean-ups: use shorter variable names to make code easier to
	  read; prefix structures we define with 'Gst' to make it clearer
	  where they come from.

2006-08-21 13:26:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/gstwavpackenc.c: Fix caps set on buffers and template caps (output is framed) and make them match (#35166...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_init),
	  (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_push_block),
	  (gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block),
	  (gst_wavpack_enc_sink_event):
	  Fix caps set on buffers and template caps (output is framed)
	  and make them match (#351663); use GST_WARNING_OBJECT instead of
	  GST_ELEMENT_WARNING; simplify push_block(); do some small
	  clean-ups here and there; fix memleak (#351663).

2006-08-21 13:12:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/elements/audiopanorama.c: Fix invalid memory access in audiopanorama test suite.
	  Original commit message from CVS:
	  * tests/check/elements/audiopanorama.c: (GST_START_TEST):
	  Fix invalid memory access in audiopanorama test suite.

2006-08-21 11:34:41 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/check/elements/.cvsignore: ignore built file
	  Original commit message from CVS:
	  * tests/check/elements/.cvsignore:
	  ignore built file

2006-08-21 10:46:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/Makefile.am: Fix the build again.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  Fix the build again.

2006-08-21 09:21:27 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiofxgood/: resubmit with the desired name *again*
	  Original commit message from CVS:
	  * gst/audiofxgood/.cvsignore:
	  * gst/audiofxgood/Makefile.am:
	  * gst/audiofxgood/audiofx.c: (plugin_init):
	  * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
	  (gst_audio_panorama_class_init), (gst_audio_panorama_init),
	  (gst_audio_panorama_set_property),
	  (gst_audio_panorama_get_property),
	  (gst_audio_panorama_get_unit_size),
	  (gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
	  (gst_audio_panorama_transform_m2s),
	  (gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
	  * gst/audiofxgood/audiopanorama.h:
	  resubmit with the desired name *again*

2006-08-20 13:09:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  use g_assert in _get_unit_size
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size):
	  * gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size):
	  use g_assert in _get_unit_size

2006-08-20 13:06:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: cleanup -unused.txt to make it useful, add previously missing docs
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-audiofxgood.xml:
	  cleanup -unused.txt to make it useful, add previously missing docs
	  * ext/Makefile.am:
	  * ext/esd/esdmon.c:
	  * ext/esd/esdsink.c:
	  * ext/esd/gstesd.c: (plugin_init):
	  reflow to get rid of two external symbols
	  * gst/audiofxgood/audiofx.c: (plugin_init):
	  re-add

2006-08-20 12:09:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiofxgood/audiofx.c
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/audiofxgood/.cvsignore:
	  * gst/audiofxgood/Makefile.am:
	  * gst/audiofxgood/audiofx.c
	  * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
	  (gst_audio_panorama_class_init), (gst_audio_panorama_init),
	  (gst_audio_panorama_set_property),
	  (gst_audio_panorama_get_property),
	  (gst_audio_panorama_get_unit_size),
	  (gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
	  (gst_audio_panorama_transform_m2s),
	  (gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
	  * gst/audiofxgood/audiopanorama.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/audiopanorama.c: (setup_panorama_m),
	  (setup_panorama_s), (cleanup_panorama), (GST_START_TEST),
	  (panorama_suite), (main):
	  Add audiofxgood plugin with audiopanorama element

2006-08-18 21:39:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.c: Fix resyncing in push mode not stopping re-syncing at embedded zeroes; skip garbage be...
	  Original commit message from CVS:
	  Based on patch by: Sebastian Dröge <slomo at circular-chaos.org>
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_sink_event),
	  (gst_wavpack_parse_get_upstream_length),
	  (gst_wavpack_parse_find_marker), (gst_wavpack_parse_resync_loop),
	  (gst_wavpack_parse_loop), (gst_wavpack_parse_resync_adapter):
	  Fix resyncing in push mode not stopping re-syncing at embedded
	  zeroes; skip garbage between frames in pull mode as well if
	  necessary; use gst_pad_query_peer_duration(); push EOS and
	  NEWSEGMENT event in right direction (#351659).

2006-08-18 17:00:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/Makefile.am: More Oss docs fixage.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  More Oss docs fixage.

2006-08-18 16:52:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added experimental SVQ3 depayloader.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_base_init),
	  (gst_rtp_sv3v_depay_class_init), (gst_rtp_sv3v_depay_init),
	  (gst_rtp_sv3v_depay_finalize), (gst_rtp_sv3v_depay_setcaps),
	  (gst_rtp_sv3v_depay_process), (gst_rtp_sv3v_depay_set_property),
	  (gst_rtp_sv3v_depay_get_property),
	  (gst_rtp_sv3v_depay_change_state),
	  (gst_rtp_sv3v_depay_plugin_init):
	  * gst/rtp/gstrtpsv3vdepay.h:
	  Added experimental SVQ3 depayloader.

2006-08-18 13:25:06 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/dv/gstdvdemux.*: When handling seek requests, don't send the newsegment event from the calling thread. Instead sa...
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek),
	  (gst_dvdemux_loop), (gst_dvdemux_change_state):
	  * ext/dv/gstdvdemux.h:
	  When handling seek requests, don't send the newsegment event from the
	  calling thread. Instead save it so it can be sent from the streaming
	  thread.

2006-08-17 15:51:50 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/multipart/multipartdemux.c: Accept leading whitespace before the boundary
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * gst/multipart/multipartdemux.c: (multipart_parse_header):
	  Accept leading whitespace before the boundary
	  This patch makes the demuxer allow some whitespace before the actual
	  boundary. This makes the demuxer work with the ``old'' gstreamer
	  multipartmuxer again (which placed an extra \n before the start
	  of the stream) Fixes #349068.

2006-08-17 15:47:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph264depay.c: Error out on non-implemented stuff.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
	  Error out on non-implemented stuff.

2006-08-16 16:50:00 +0000  Andy Wingo <wingo@pobox.com>

	  ext/ladspa/gstsignalprocessor.c: Make ladspa elements reusable. Fixes #350006.
	  Original commit message from CVS:
	  Patch by: Andy Wingo <wingo at pobox dot com>
	  * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setup),
	  (gst_signal_processor_start), (gst_signal_processor_stop),
	  (gst_signal_processor_cleanup), (gst_signal_processor_setcaps),
	  (gst_signal_processor_pen_buffer), (gst_signal_processor_flush),
	  (gst_signal_processor_do_pulls), (gst_signal_processor_do_pushes),
	  (gst_signal_processor_change_state):
	  Make ladspa elements reusable. Fixes #350006.

2006-08-16 15:33:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ladspa/gstladspa.c: Convert ' ' into '_'. Try to keep as many characters in the padtemplate names as possible.
	  Original commit message from CVS:
	  * ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
	  Convert ' ' into '_'. Try to keep as many characters in the padtemplate
	  names as possible.

2006-08-16 14:47:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ladspa/gstsignalprocessor.c: A push() gives away our refcount so we should not use the buffer on the pen anymore.
	  Original commit message from CVS:
	  * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_flush),
	  (gst_signal_processor_do_pushes):
	  A push() gives away our refcount so we should not use the buffer on the
	  pen anymore.

2006-08-16 13:48:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/gstossmixerelement.c: Don't leak device string.
	  Original commit message from CVS:
	  * sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
	  (gst_oss_mixer_element_finalize):
	  Don't leak device string.

2006-08-16 13:01:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Require CVS of GStreamer core and -base (for
	  Original commit message from CVS:
	  * configure.ac:
	  Require CVS of GStreamer core and -base (for
	  GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).
	  * ext/taglib/gstid3v2mux.cc:
	  Write extended comment tags properly (#348762).
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	  (parse_comment_frame):
	  Extract COMM frames into extended comments, which makes it
	  easier to properly retain the description bit of the tag
	  and maintain this information when re-tagging (#348762).

2006-08-16 12:02:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Don't try to run annodex unit tests if the annodex plugin has not been built (Fixes #351116).
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Don't try to run annodex unit tests if the annodex
	  plugin has not been built (Fixes #351116).

2006-08-16 10:53:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/autodetect/gstautoaudiosink.c: When we can't find a usable audiosink, don't error out, but use a fake sink instea...
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_find_best):
	  When we can't find a usable audiosink, don't error out,
	  but use a fake sink instead and post a warning message
	  on the bus (#341278).

2006-08-16 10:40:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/: In push mode, re-sync to next wavpack header if sync is lost (#351557). Also use hyphens instead of und...
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo at circular-chaos.org>
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init):
	  * ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_resync_adapter), (gst_wavpack_parse_chain):
	  In push mode, re-sync to next wavpack header if sync is lost
	  (#351557). Also use hyphens instead of underscores in
	  GObject property names.

2006-08-16 10:22:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/: Document OSS elements; add gtk-doc blurb with 'Since 0.10.5' for ossmixer's new device property.
	  Original commit message from CVS:
	  * sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init):
	  * sys/oss/gstosssink.c:
	  * sys/oss/gstosssrc.c:
	  Document OSS elements; add gtk-doc blurb with 'Since 0.10.5' for
	  ossmixer's new device property.
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Add docs for OSS elements.
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-shout2send.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update to CVS version.

2006-08-16 10:05:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Caps extra properties must be defined as strings for depayloaders because they are generated from an SDP.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrdepay.c:
	  * gst/rtp/gstrtpmp4gdepay.c:
	  Caps extra properties must be defined as strings for
	  depayloaders because they are generated from an SDP.
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init),
	  (gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init),
	  (gst_rtp_h264_depay_finalize), (decode_base64),
	  (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
	  (gst_rtp_h264_depay_set_property),
	  (gst_rtp_h264_depay_get_property),
	  (gst_rtp_h264_depay_change_state),
	  (gst_rtp_h264_depay_plugin_init):
	  * gst/rtp/gstrtph264depay.h:
	  Added basic, not completely functional RFC 3984 H264 depayloader.

2006-08-16 09:48:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtpdec.c: Add pads after setting them up.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
	  Add pads after setting them up.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	  (gst_rtspsrc_init), (gst_rtspsrc_finalize),
	  (gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_stream_setup_rtp),
	  (gst_rtspsrc_stream_configure_transport),
	  (gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
	  (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
	  (gst_rtspsrc_pause):
	  * gst/rtsp/gstrtspsrc.h:
	  Fix interleaved mode.
	  - Protect streaming with lock.
	  - Combine flows
	  - set caps on outgoing buffers.
	  - strip trailing \0 from data packets.
	  - Configure RTP/RTCP in stream.
	  Use DEBUG_OBJECT more.

2006-08-16 09:29:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.c: Turn a g_print into a DEBUG line.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
	  Turn a g_print into a DEBUG line.

2006-08-16 09:25:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/oss/: Small cleanups. Better error reporting.
	  Original commit message from CVS:
	  * sys/oss/gstossmixer.c: (gst_ossmixer_open), (gst_ossmixer_new):
	  * sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
	  (gst_oss_mixer_element_init), (gst_oss_mixer_element_set_property),
	  (gst_oss_mixer_element_get_property),
	  (gst_oss_mixer_element_change_state):
	  * sys/oss/gstossmixerelement.h:
	  Small cleanups. Better error reporting.
	  Add device property for the mixer instead of the hardcoded
	  /dev/mixer. Fixes #350785.
	  API: GstOssMixerElement::device property

2006-08-15 22:44:27 +0000  Jens Granseuer <jensgr@gmx.net>

	  gconf/Makefile.am: Make --disable-schemas work right (they still need to be copied to the installation directory, jus...
	  Original commit message from CVS:
	  Patch by: Jens Granseuer <jensgr at gmx net>
	  * gconf/Makefile.am:
	  Make --disable-schemas work right (they still need
	  to be copied to the installation directory, just not
	  applied). Fixes #351347 (also #344100).

2006-08-15 20:29:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackparse.*: Make wavpackparse also work in push-mode (not seekable yet though); some small clean-u...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_class_init),
	  (gst_wavpack_parse_reset), (gst_wavpack_parse_get_src_query_types),
	  (gst_wavpack_parse_src_query),
	  (gst_wavpack_parse_handle_seek_event),
	  (gst_wavpack_parse_sink_event), (gst_wavpack_parse_init),
	  (gst_wavpack_parse_create_src_pad),
	  (gst_wavpack_parse_push_buffer), (gst_wavpack_parse_loop),
	  (gst_wavpack_parse_chain), (gst_wavpack_parse_sink_activate),
	  (gst_wavpack_parse_sink_activate_pull):
	  * ext/wavpack/gstwavpackparse.h:
	  Patch by: Sebastian Dröge <slomo at circular-chaos.org>
	  Make wavpackparse also work in push-mode (not seekable yet though);
	  some small clean-ups along the way; add support for SEEKING query
	  and query types function. (#351495).

2006-08-14 11:37:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	* win32/common/config.h:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

2006-08-14 11:14:43 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* win32/common/config.h:
	  releasing 0.10.4
	  Original commit message from CVS:
	  releasing 0.10.4

2006-08-14 10:06:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Extract all references/redirections if there is more than one and sort them; also extract mini...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_redirects_sort_func),
	  (qtdemux_process_redirects), (qtdemux_parse_tree):
	  Extract all references/redirections if there is more
	  than one and sort them; also extract minimum required
	  bitrate information if available. (#350399)

2006-08-10 14:10:28 +0000  Edward Hervey <edward@fluendo.com>

	  Send the newsegment event in the streaming thread.
	  Original commit message from CVS:
	  Patch by: Edward Hervey <edward@fluendo.com>
	  * configure.ac:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_data):
	  Send the newsegment event in the streaming thread.
	  Fixes #347529

2006-08-10 14:02:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* win32/common/config.h:
	  bumped for prerel
	  Original commit message from CVS:
	  bumped for prerel

2006-08-10 13:10:38 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  update translations
	  Original commit message from CVS:
	  update translations

2006-08-08 14:55:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Fix silly typo.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_tree):
	  Fix silly typo.

2006-08-08 14:46:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery: mention bug number
	  Original commit message from CVS:
	  ChangeLog surgery: mention bug number

2006-08-08 14:40:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/: Refuse sink caps in the encoder if width or height is not a multiple of 16, the encoder does not support t...
	  Original commit message from CVS:
	  * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps),
	  (gst_smokeenc_resync), (gst_smokeenc_chain):
	  Refuse sink caps in the encoder if width or height is not a
	  multiple of 16, the encoder does not support that yet; along the
	  same lines, check the return value of the encoder setup function;
	  also remove some debug log clutter.

2006-08-04 11:38:54 +0000  Andy Wingo <wingo@pobox.com>

	  ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing whether a processor can work in place or not, and for...
	  Original commit message from CVS:
	  2006-08-04  Andy Wingo  <wingo@pobox.com>
	  * ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing
	  whether a processor can work in place or not, and for keeping
	  track of its state. Change the FlowReturn instance variable from
	  "state" to "flow_state", all callers changed.
	  * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setup)
	  (gst_signal_processor_start, gst_signal_processor_stop)
	  (gst_signal_processor_cleanup): New functions to manage the
	  processor's state.
	  (gst_signal_processor_setcaps): start() as well as setup() here.
	  (gst_signal_processor_prepare): Respect CAN_PROCESS_IN_PLACE.
	  (gst_signal_processor_change_state): Stop and cleanup the
	  processor as we go to NULL.
	  * ext/ladspa/gstladspa.c (gst_ladspa_base_init): Reuse buffers if
	  INPLACE_BROKEN is not set.
	  * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_prepare):
	  Do the alloc_buffer in bytes, not frames.

2006-08-04 10:21:26 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximage/ximageutil.c: Fix rgb masks when recording in < 24bpp.
	  Original commit message from CVS:
	  2006-08-04  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
	  Fix rgb masks when recording in < 24bpp.

2006-08-04 09:20:26 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/ladspa/gstsignalprocessor.c:
	  BPB
	  Original commit message from CVS:
	  (gst_signal_processor_src_activate_pull): BPB

2006-08-04 09:05:53 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/ladspa/gstsignalprocessor.c:
	  ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps) (gst_signal_processor_prepare) (gst_signal_processor_u...
	  Original commit message from CVS:
	  2006-08-04  Andy Wingo  <wingo@pobox.com>
	  * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps)
	  (gst_signal_processor_prepare)
	  (gst_signal_processor_update_inputs)
	  (gst_signal_processor_process, gst_signal_processor_pen_buffer)
	  (gst_signal_processor_flush)
	  (gst_signal_processor_sink_activate_push)
	  (gst_signal_processor_src_activate_pull)
	  (gst_signal_processor_change_state): Remove the last of the code
	  that assumes that we process whole buffers at a time. Fix some
	  debugging. Seems to work now in some cases.

2006-07-31 22:27:22 +0000  Andy Wingo <wingo@pobox.com>

	  ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process): Fix nframes-choosing.
	  Original commit message from CVS:
	  2006-08-01  Andy Wingo  <wingo@pobox.com>
	  * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process):
	  Fix nframes-choosing.
	  (gst_signal_processor_init): Init pending_in and pending_out.

2006-07-31 22:03:09 +0000  Andy Wingo <wingo@pobox.com>

	  ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No more default sample rate, although we never check tha...
	  Original commit message from CVS:
	  2006-08-01  Andy Wingo  <wingo@pobox.com>
	  * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No
	  more default sample rate, although we never check that the sample
	  rate actually gets set. Something for the future.
	  (gst_signal_processor_setcaps): Some refcount fixes, flow fixes.
	  (gst_signal_processor_event): Refcount fixen.
	  (gst_signal_processor_process): Pull the number of frames to
	  process from the sizes of the buffers in the input pens.
	  (gst_signal_processor_pen_buffer): Remove an incorrect FIXME :)
	  (gst_signal_processor_do_pulls): Add an nframes argument, and use
	  it instead of buffer_frames.
	  (gst_signal_processor_getrange): Refcount fixen, pass nframes on
	  to do_pulls.
	  (gst_signal_processor_chain)
	  (gst_signal_processor_sink_activate_push)
	  (gst_signal_processor_src_activate_pull):  Refcount fixen.
	  * ext/ladspa/gstsignalprocessor.h: No more buffer_frames, yay.

2006-07-31 19:44:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/ladspa/gstsignalprocessor.c: don't query buffer-frames from caps, add lots of debug-log, try fix for assert (#349...
	  Original commit message from CVS:
	  * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
	  (gst_signal_processor_process):
	  don't query buffer-frames from caps, add lots of debug-log,
	  try fix for assert (#349189)

2006-07-31 15:58:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.c: Fix docs.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c:
	  Fix docs.

2006-07-29 16:32:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/ladspa/gstsignalprocessor.c: Add debugs logs here and there, add more error handling, add some
	  Original commit message from CVS:
	  * ext/ladspa/gstsignalprocessor.c:
	  (gst_signal_processor_add_pad_from_template),
	  (gst_signal_processor_init), (gst_signal_processor_setcaps),
	  (gst_signal_processor_process), (gst_signal_processor_pen_buffer),
	  (gst_signal_processor_do_pulls), (gst_signal_processor_getrange),
	  (gst_signal_processor_sink_activate_push),
	  (gst_signal_processor_src_activate_pull),
	  (gst_signal_processor_change_state):
	  Add debugs logs here and there, add more error handling, add some
	  FIXME comments, filed #349189

2006-07-29 11:22:47 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/jpeg/gstsmokeenc.c: Set caps on buffer correctly.  Fixes bug #349155.
	  Original commit message from CVS:
	  2006-07-29  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
	  (gst_smokeenc_setcaps), (gst_smokeenc_chain):
	  Set caps on buffer correctly.  Fixes bug #349155.

2006-07-28 16:17:17 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/multipart/multipartdemux.c: Uses GstAdapter instead of own buffering.
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
	  (gst_multipart_demux_class_init), (gst_multipart_demux_init),
	  (gst_multipart_demux_finalize), (get_line_end),
	  (multipart_parse_header), (multipart_find_boundary),
	  (gst_multipart_demux_chain), (gst_multipart_demux_change_state),
	  (gst_multipart_set_property), (gst_multipart_get_property):
	  Uses GstAdapter instead of own buffering.
	  Actually parses the mime-type correctly (In tests the mime-type was
	  always "" with the old version).
	  Uses the Content-length header if available to speed up things.
	  Reliably autoscans the boundary name by default.
	  Fixes #349068.
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	  Don't start the stream with a \n.

2006-07-28 08:32:47 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/gstsunaudiosrc.c: Open source with O_NONBLOCK (#349015).
	  Original commit message from CVS:
	  Patch by: Brian Cameron <brian dot cameron at sun com>
	  * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	  Open source with O_NONBLOCK (#349015).

2006-07-28 08:21:27 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.*: Whitespace fixes and more debug
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
	  (gst_avi_demux_massage_index):
	  * gst/avi/gstavidemux.h:
	  Whitespace fixes and more debug

2006-07-27 11:21:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/autodetect/gstautoaudiosink.c: Get rid of old and unused magic sound-server properties stuff.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_create_element_with_pretty_name),
	  (gst_auto_audio_sink_find_best),
	  (gst_auto_audio_sink_change_state):
	  Get rid of old and unused magic sound-server properties stuff.
	  Add suffix to child sink's name that makes it easy to see from
	  the name alone which type it actually is (alsa, oss, esd, etc.).

2006-07-27 10:05:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstudpsrc.*: Rename "buffer" to "buffer-size" to make clear it is a size we set and not some sort of feature ...
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	  (gst_udpsrc_set_property), (gst_udpsrc_get_property),
	  (gst_udpsrc_start):
	  * gst/udp/gstudpsrc.h:
	  Rename "buffer" to "buffer-size" to make clear it is a size we set and
	  not some sort of feature we enable.

2006-07-27 10:01:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/udp/gstudpsrc.c: Use CLOSE_SOCKET() here instead of close() to maintain win32 workiness.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  Use CLOSE_SOCKET() here instead of close() to maintain
	  win32 workiness.

2006-07-27 09:04:51 +0000  Thijs Vermeir <thijs.vermeir@barco.com>

	  gst/udp/gstudpsrc.*: Added "buffer" property to control the kernel receive buffer size.
	  Original commit message from CVS:
	  Patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	  (gst_udpsrc_create), (gst_udpsrc_set_property),
	  (gst_udpsrc_get_property), (gst_udpsrc_start):
	  * gst/udp/gstudpsrc.h:
	  Added "buffer" property to control the kernel receive buffer size.
	  Update documentation.
	  Small cleanups. Fixes #348752.
	  API: buffer property

2006-07-26 17:09:04 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.c: Fix lame putting lots of 0's at start of mp3.  Fixes bug #348786.
	  Original commit message from CVS:
	  2006-07-26  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_setup):
	  Fix lame putting lots of 0's at start of mp3.  Fixes bug #348786.

2006-07-26 16:36:59 +0000  Kai Vehmanen <kv2004@eca.cx>

	  gst/rtp/: Fix timestamp calculation on outgoing RTP packets.
	  Original commit message from CVS:
	  Patch by: Kai Vehmanen <kv2004 at eca dot cx>
	  * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
	  (gst_rtp_pcma_pay_handle_buffer):
	  * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush),
	  (gst_rtp_pcmu_pay_handle_buffer):
	  Fix timestamp calculation on outgoing RTP packets.
	  Fixes #348675.

2006-07-26 10:07:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gstid3v2mux.cc: is still sub-optimal though, since we don't retain or extract the comment descriptions pro...
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc:
	  Fix writing of comment frames (should be COMM not TCOM),
	  is still sub-optimal though, since we don't retain or
	  extract the comment descriptions properly (#334375,
	  also see #334375).

2006-07-26 09:02:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: #define 'fact' RIFF chunk if we are not compiling against
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  #define 'fact' RIFF chunk if we are not compiling against
	  -base CVS (we don't want to depend on -base CVS for this
	  one define only, and also not for release order reasons).

2006-07-26 08:17:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gstid3v2mux.cc: Handle multiple tags of the same type properly. Re-inject unparsed ID3v2 frames that we ge...
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc:
	  Handle multiple tags of the same type properly. Re-inject
	  unparsed ID3v2 frames that we get as binary blobs from
	  id3demux into the tag again so we don't lose information
	  when retagging (#334375).

2006-07-25 17:54:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/ximage/gstximagesrc.c: Document newly-added properties properly, so that there is a 'Since: 0.10.4' in the plugin...
	  Original commit message from CVS:
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_class_init):
	  Document newly-added properties properly, so that there is a
	  'Since: 0.10.4' in the plugin docs. Convert some property
	  names into canonical GObject style (GObject will do that
	  internally anyway).

2006-07-25 16:47:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3tags.c: Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as well, and add the version to...
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c:
	  (id3demux_add_id3v2_frame_blob_to_taglist):
	  Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as
	  well, and add the version to the blob's buffer caps, since that
	  information will be needed for deserialisation later on (#348644).

2006-07-25 13:14:05 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavidemux.c: Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed indentation and spacing.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes),
	  (gst_avi_demux_parse_stream):
	  Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed
	  indentation and spacing.

2006-07-24 21:43:06 +0000  Sébastien Moutte <sebastien@moutte.net>

	  sys/directsound/gstdirectsoundsink.*: Add an attenuation property that will directly attenuate the directsound buffer.
	  Original commit message from CVS:
	  * sys/directsound/gstdirectsoundsink.h:
	  * sys/directsound/gstdirectsoundsink.c:
	  Add an attenuation property that will directly attenuate the
	  directsound buffer.
	  Change the size of the directsound secondary buffer to a half second.
	  Add more debug logs.
	  Add a lock to protect dsound buffer write access.
	  Fix a bad implementation of reset.
	  * sys/directsound/gstdirectdrawsink.c:
	  * sys/directsound/gstdirectdrawsink.h:
	  Add a keep_aspect_ratio property.
	  Do not use overlay if not supported.
	  Add more debug logs.
	  Remove overwrite of WM_ERASEBKGND message handling. It was not
	  redrawing border when keep_aspect_ratio was enabled.
	  * win32/common/config.h:
	  update version waiting an auto-generated config.h

2006-07-24 15:25:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Update files to CVS/Prerelease version, add esdsink docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-1394.xml:
	  * docs/plugins/inspect/plugin-aasink.xml:
	  * docs/plugins/inspect/plugin-alaw.xml:
	  * docs/plugins/inspect/plugin-alpha.xml:
	  * docs/plugins/inspect/plugin-alphacolor.xml:
	  * docs/plugins/inspect/plugin-annodex.xml:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * docs/plugins/inspect/plugin-auparse.xml:
	  * docs/plugins/inspect/plugin-autodetect.xml:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * docs/plugins/inspect/plugin-cacasink.xml:
	  * docs/plugins/inspect/plugin-cairo.xml:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  * docs/plugins/inspect/plugin-cutter.xml:
	  * docs/plugins/inspect/plugin-debug.xml:
	  * docs/plugins/inspect/plugin-dv.xml:
	  * docs/plugins/inspect/plugin-efence.xml:
	  * docs/plugins/inspect/plugin-effectv.xml:
	  * docs/plugins/inspect/plugin-esdsink.xml:
	  * docs/plugins/inspect/plugin-flac.xml:
	  * docs/plugins/inspect/plugin-flxdec.xml:
	  * docs/plugins/inspect/plugin-gconfelements.xml:
	  * docs/plugins/inspect/plugin-gdkpixbuf.xml:
	  * docs/plugins/inspect/plugin-goom.xml:
	  * docs/plugins/inspect/plugin-halelements.xml:
	  * docs/plugins/inspect/plugin-icydemux.xml:
	  * docs/plugins/inspect/plugin-id3demux.xml:
	  * docs/plugins/inspect/plugin-jpeg.xml:
	  * docs/plugins/inspect/plugin-level.xml:
	  * docs/plugins/inspect/plugin-matroska.xml:
	  * docs/plugins/inspect/plugin-mulaw.xml:
	  * docs/plugins/inspect/plugin-multipart.xml:
	  * docs/plugins/inspect/plugin-navigationtest.xml:
	  * docs/plugins/inspect/plugin-ossaudio.xml:
	  * docs/plugins/inspect/plugin-png.xml:
	  * docs/plugins/inspect/plugin-rtp.xml:
	  * docs/plugins/inspect/plugin-rtsp.xml:
	  * docs/plugins/inspect/plugin-shout2send.xml:
	  * docs/plugins/inspect/plugin-smpte.xml:
	  * docs/plugins/inspect/plugin-speex.xml:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * docs/plugins/inspect/plugin-videobalance.xml:
	  * docs/plugins/inspect/plugin-videobox.xml:
	  * docs/plugins/inspect/plugin-videoflip.xml:
	  * docs/plugins/inspect/plugin-videomixer.xml:
	  * docs/plugins/inspect/plugin-wavenc.xml:
	  * docs/plugins/inspect/plugin-wavparse.xml:
	  * docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update files to CVS/Prerelease version, add esdsink docs.
	  * ext/esd/esdsink.c:
	  Add gtk-doc blurb.
	  * gst/rtp/gstrtpmp4vpay.c:
	  Fix typo in element description.

2006-07-24 14:54:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery: fix Stefan's e-mail address
	  Original commit message from CVS:
	  ChangeLog surgery: fix Stefan's e-mail address

2006-07-24 14:49:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/esd/esdsink.c: Prevent libesd from auto-spawning a sound daemon if it is not already running. Now that we don't d...
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_open),
	  (gst_esdsink_factory_init):
	  Prevent libesd from auto-spawning a sound daemon if it
	  is not already running. Now that we don't do evil stuff
	  like that any longer we can give esdsink a rank so that
	  autoaudiosink will try it as well if all other audio
	  sinks fail (#343051).

2006-07-24 14:42:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/esd/Makefile.am: Oops, need to remove README from EXTRA_DIST as well.
	  Original commit message from CVS:
	  * ext/esd/Makefile.am:
	  Oops, need to remove README from EXTRA_DIST as well.

2006-07-24 14:37:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/esd/README: Remove, it contains nothing useful anyway.
	  Original commit message from CVS:
	  * ext/esd/README:
	  Remove, it contains nothing useful anyway.
	  * ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_prepare),
	  (gst_esdsink_delay):
	  Some small clean-ups; use GST_BOILERPLATE etc.

2006-07-24 14:16:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/law/: Fix negotiation to deal with ANY/EMPTY caps instead of leaking.
	  Original commit message from CVS:
	  * gst/law/alaw-decode.c: (alawdec_getcaps):
	  * gst/law/alaw-encode.c: (alawenc_getcaps), (gst_alawenc_chain):
	  * gst/law/mulaw-decode.c: (mulawdec_getcaps):
	  * gst/law/mulaw-encode.c: (mulawenc_getcaps):
	  Fix negotiation to deal with ANY/EMPTY caps instead of leaking.

2006-07-24 13:40:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.*: Use information from 'fact' chunk for length calculation of compressed samples. Calculate...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
	  (gst_wavparse_other), (gst_wavparse_perform_seek),
	  (gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers),
	  (gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
	  (gst_wavparse_pad_query):
	  * gst/wavparse/gstwavparse.h:
	  Use information from 'fact' chunk for length calculation of compressed
	  samples. Calculate bps if bogus value is found in wav header (embeded
	  mp2/mp3).

2006-07-24 11:48:03 +0000  Joni Valtanen <joni.valtanen@movial.fi>

	  Port udp plugin to win32 (#345288).
	  Original commit message from CVS:
	  Based on patch by: Joni Valtanen  <joni dot valtanen at movial fi>
	  * configure.ac:
	  * gst/udp/Makefile.am:
	  * gst/udp/gstdynudpsink.c: (gst_dynudpsink_init),
	  (gst_dynudpsink_finalize), (gst_dynudpsink_close):
	  * gst/udp/gstdynudpsink.h:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init),
	  (gst_multiudpsink_finalize), (gst_multiudpsink_close):
	  * gst/udp/gstmultiudpsink.h:
	  * gst/udp/gstudp.c: (plugin_init):
	  * gst/udp/gstudpsink.h:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create),
	  (gst_udpsrc_start), (gst_udpsrc_stop):
	  * gst/udp/gstudpsrc.h:
	  * gst/udp/gstudpnetutils.c: (gst_udp_net_utils_win32_inet_aton),
	  (gst_udp_net_utils_win32_wsa_startup):
	  * gst/udp/gstudpnetutils.h:
	  Port udp plugin to win32 (#345288).

2006-07-24 11:00:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspconnection.c: Remove unwanted DEBUG line.
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_send):
	  Remove unwanted DEBUG line.

2006-07-23 11:33:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/: On second thought, it might be wiser and more efficient not to do tag registration from a streaming th...
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (plugin_init):
	  * gst/id3demux/id3tags.c:
	  (id3demux_add_id3v2_frame_blob_to_taglist):
	  * gst/id3demux/id3tags.h:
	  On second thought, it might be wiser and more efficient
	  not to do tag registration from a streaming thread.

2006-07-23 10:56:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3tags.c: Put ID3v2 frames we can't parse as binary blobs into private tags, so that they are not lost ...
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c:
	  (id3demux_add_id3v2_frame_blob_to_taglist),
	  (id3demux_id3v2_frames_to_tag_list):
	  Put ID3v2 frames we can't parse as binary blobs into private
	  tags, so that they are not lost when retagging, at least once
	  id3v2mux has been taught to re-inject those frames again.
	  See bug #334375.

2006-07-21 10:57:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Fix some leaks.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_process_next_entry):
	  Fix some leaks.
	  * gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
	  Don't use \n in debug lines.

2006-07-20 18:48:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: Add annodex and icydemux, cleanup the sections a bit
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Add annodex and icydemux, cleanup the sections a bit

2006-07-19 14:36:00 +0000  Martin Szulecki <compiz@sukimashita.com>

	  sys/v4l2/gstv4l2object.c: If "device-name" is requested and the device is not open, try to temporarily open it to obt...
	  Original commit message from CVS:
	  Patch by: Martin Szulecki
	  * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_get_property_helper):
	  If "device-name" is requested and the device is not
	  open, try to temporarily open it to obtain this
	  information (#342494).

2006-07-19 11:52:53 +0000  Alex Lancaster <alexl@users.sourceforge.net>

	  ext/taglib/gstid3v2mux.cc: Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as
	  Original commit message from CVS:
	  Patch by: Alex Lancaster <alexl at users sourceforge net>
	  * ext/taglib/gstid3v2mux.cc:
	  Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as
	  ID3v2 TSSE frames (#347898).

2006-07-19 07:40:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery: mention fixed bug
	  Original commit message from CVS:
	  ChangeLog surgery: mention fixed bug

2006-07-18 19:59:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/avi/gstavimux.c: Respect mpegversion for "video/mpeg" and give message in case of unhandled versions.
	  Original commit message from CVS:
	  * gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
	  Respect mpegversion for "video/mpeg" and give message in case of
	  unhandled versions.

2006-07-18 18:05:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/gstwavpackdec.c: Fix caps after previous change to byte order endianness.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  Fix caps after previous change to byte order endianness.
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
	  (gst_wavpack_parse_sink_event), (gst_wavpack_parse_init),
	  (gst_wavpack_parse_loop):
	  * ext/wavpack/gstwavpackparse.h:
	  Queue incoming events if there's no source pad yet and
	  send them downstream later when the pad is there.

2006-07-18 16:47:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/gstwavpackdec.*: Output audio in native byte order (which is also how we get samples from wavpack); outpu...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
	  (gst_wavpack_dec_format_samples),
	  (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_chain),
	  (gst_wavpack_dec_change_state):
	  * ext/wavpack/gstwavpackdec.h:
	  Output audio in native byte order (which is also how we get
	  samples from wavpack); output samples with 21-24 bit depth
	  with 32 bit width (makes things easier for us).

2006-07-18 15:53:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/gstwavpackdec.*: More clean-ups: remove most of the disfunctional correction pad stuff for now, if it eve...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init),
	  (gst_wavpack_dec_class_init), (gst_wavpack_dec_init),
	  (gst_wavpack_dec_finalize), (gst_wavpack_dec_format_samples),
	  (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_chain),
	  (gst_wavpack_dec_sink_event), (gst_wavpack_dec_change_state):
	  * ext/wavpack/gstwavpackdec.h:
	  More clean-ups: remove most of the disfunctional correction
	  pad stuff for now, if it ever gets implemented a lot of stuff
	  will have to be rewritten anyway; redo chain function, move
	  errors to end, error out instead of g_assert()ing. Also rename
	  overly long variable 'wavpackdec' to just 'dec'; miscellaneous
	  other small stuff.

2006-07-18 14:08:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Check for wavpack version and define WAVPACK_OLD_API if necessary.
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge <slomo at circular-chaos.org>
	  * configure.ac:
	  Check for wavpack version and define WAVPACK_OLD_API if
	  necessary.
	  * ext/wavpack/Makefile.am:
	  * ext/wavpack/gstwavpackcommon.c: (gst_wavpack_read_header),
	  (gst_wavpack_read_metadata):
	  * ext/wavpack/gstwavpackcommon.h:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init),
	  (gst_wavpack_dec_class_init), (gst_wavpack_dec_init),
	  (gst_wavpack_dec_finalize), (gst_wavpack_dec_format_samples),
	  (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_chain),
	  (gst_wavpack_dec_sink_event), (gst_wavpack_dec_change_state),
	  (gst_wavpack_dec_request_new_pad), (gst_wavpack_dec_plugin_init):
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
	  (gst_wavpack_enc_init), (gst_wavpack_enc_finalize),
	  (gst_wavpack_enc_set_wp_config):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init),
	  (gst_wavpack_parse_finalize), (gst_wavpack_parse_class_init),
	  (gst_wavpack_parse_index_get_entry_from_sample),
	  (gst_wavpack_parse_scan_to_find_sample),
	  (gst_wavpack_parse_handle_seek_event),
	  (gst_wavpack_parse_create_src_pad):
	  * ext/wavpack/gstwavpackstreamreader.c:
	  * ext/wavpack/gstwavpackstreamreader.h:
	  Port to new/official wavpack API, don't use API that was exported
	  in wavpack header files and in the lib but meant to be private, at
	  least not for recent wavpack versions; misc. 'cleanups' (#347443).

2006-07-17 10:25:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Store duration in uint64 too instead of clipping.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
	  (gst_qtdemux_prepare_current_sample),
	  (gst_qtdemux_loop_state_movie):
	  Store duration in uint64 too instead of clipping.
	  When we do a keyframe seek and the requested time is at the
	  keyframe, don't seek back to the beginning of the keyframe.
	  Fixes #347439.

2006-07-17 10:22:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libpng/gstpngdec.*: Use statically allocated segment instead of leaking.
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_init), (buffer_clip),
	  (gst_pngdec_caps_create_and_set), (gst_pngdec_task),
	  (gst_pngdec_chain), (gst_pngdec_sink_event),
	  (gst_pngdec_libpng_init), (gst_pngdec_change_state),
	  (gst_pngdec_sink_activate_push):
	  * ext/libpng/gstpngdec.h:
	  Use statically allocated segment instead of leaking.
	  Various cleanups.
	  Fix flush and seek handling.

2006-07-16 14:31:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added simple generic mpeg4 depayloader.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_base_init),
	  (gst_rtp_mp4g_depay_class_init), (gst_rtp_mp4g_depay_init),
	  (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process),
	  (gst_rtp_mp4g_depay_set_property),
	  (gst_rtp_mp4g_depay_get_property),
	  (gst_rtp_mp4g_depay_change_state),
	  (gst_rtp_mp4g_depay_plugin_init):
	  * gst/rtp/gstrtpmp4gdepay.h:
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
	  (gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_setcaps),
	  (gst_rtp_mp4g_pay_flush):
	  Added simple generic mpeg4 depayloader.
	  Fix generic mpeg4 payloader.

2006-07-15 15:25:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtsp/gstrtspsrc.c: Don't try doing state changes on a NULL pointer.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state):
	  Don't try doing state changes on a NULL pointer.

2006-07-15 11:50:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/: Do not use deprecated gtk functions.
	  Original commit message from CVS:
	  * gst/spectrum/demo-audiotest.c: (main):
	  * gst/spectrum/demo-osssrc.c: (main):
	  Do not use deprecated gtk functions.

2006-07-14 13:33:54 +0000  Sebastien Cote <sebas642@yahoo.ca>

	  gst/rtp/gstrtpamrdepay.*: rtpamrdec isn't a subclass of GstBaseRtpDepayload.
	  Original commit message from CVS:
	  Patch by: Sebastien Cote <sebas642 at yahoo dot ca>
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_base_init),
	  (gst_rtp_amr_depay_class_init), (gst_rtp_amr_depay_init),
	  (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
	  * gst/rtp/gstrtpamrdepay.h:
	  rtpamrdec isn't a subclass of GstBaseRtpDepayload.
	  Fixes #321191

2006-07-14 12:01:05 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximage/gstximagesrc.c: Fix segfault when moving mouse pointer to the bottom right corner.
	  Original commit message from CVS:
	  2006-07-14  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	  (gst_ximage_src_get_caps), (gst_ximage_src_class_init):
	  Fix segfault when moving mouse pointer to the bottom right corner.

2006-07-13 15:22:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* docs/plugins/inspect/plugin-qtdemux.xml:
	  remove sdlvideosink plugin and update the rest
	  Original commit message from CVS:
	  remove sdlvideosink plugin and update the rest

2006-07-12 09:34:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added mpeg2 TS depayloader. Closing #347234.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_base_init),
	  (gst_rtp_mp2t_depay_class_init), (gst_rtp_mp2t_depay_init),
	  (gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process),
	  (gst_rtp_mp2t_depay_set_property),
	  (gst_rtp_mp2t_depay_get_property),
	  (gst_rtp_mp2t_depay_change_state),
	  (gst_rtp_mp2t_depay_plugin_init):
	  * gst/rtp/gstrtpmp2tdepay.h:
	  Added mpeg2 TS depayloader. Closing #347234.

2006-07-12 09:28:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/spectrum/gstspectrum.c: Fix typo in property nick.
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
	  Fix typo in property nick.

2006-07-11 22:46:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cdio/gstcdiocddasrc.c: Remove g_assert that shouldn't be there.
	  Original commit message from CVS:
	  * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_close):
	  Remove g_assert that shouldn't be there.

2006-07-10 20:11:34 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.*: Don't push tag events found by gst_riff_parse_info() before outputting
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	  (gst_avi_demux_stream_header), (push_tag_lists):
	  * gst/avi/gstavidemux.h:
	  Don't push tag events found by gst_riff_parse_info() before outputting
	  GST_EVENT_NEWSEGMENT.

2006-07-10 16:41:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: replaced closesocket and close in code with one CLOSE_SOCKET.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/rtspconnection.c: (rtsp_connection_send),
	  (rtsp_connection_close):
	  * gst/rtsp/rtspdefs.h:
	  replaced closesocket and close in code with one CLOSE_SOCKET.
	  Some more cleanups. Fixes #345301.

2006-07-10 15:26:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/autodetect/gstautoaudiosink.c: Fix example pipeline in docs.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c:
	  Fix example pipeline in docs.

2006-07-10 14:49:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/filter/: Don't forget new files.
	  Original commit message from CVS:
	  * gst/filter/gstbpwsinc.h:
	  * gst/filter/gstiir.h:
	  * gst/filter/gstlpwsinc.h:
	  Don't forget new files.

2006-07-10 14:42:15 +0000  Mathis Hofer <mathis.hofer@dreamlab.net>

	  Ported the gstfilter plugin to GStreamer 0.10.
	  Original commit message from CVS:
	  Patch by: Mathis Hofer <mathis dot hofer at dreamlab dot net>
	  * configure.ac:
	  * gst/filter/Makefile.am:
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
	  (gst_bpwsinc_base_init), (gst_bpwsinc_class_init),
	  (gst_bpwsinc_init), (bpwsinc_set_caps), (bpwsinc_transform_ip),
	  (bpwsinc_set_property), (bpwsinc_get_property):
	  * gst/filter/gstfilter.c: (plugin_init):
	  * gst/filter/gstfilter.h:
	  * gst/filter/gstiir.c: (gst_iir_dispose), (gst_iir_base_init),
	  (gst_iir_class_init), (gst_iir_init), (iir_set_caps),
	  (iir_transform_ip), (iir_set_property), (iir_get_property):
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
	  (gst_lpwsinc_base_init), (gst_lpwsinc_class_init),
	  (gst_lpwsinc_init), (lpwsinc_set_caps), (lpwsinc_transform_ip),
	  (lpwsinc_set_property), (lpwsinc_get_property):
	  Ported the gstfilter plugin to GStreamer 0.10.

2006-07-10 10:21:57 +0000  Rob Taylor <robtaylor@floopily.org>

	  gst/udp/gstmultiudpsink.c: If a destination is added before the stream is set to PAUSED, the multicast group is not j...
	  Original commit message from CVS:
	  Patch by: Rob Taylor <robtaylor at floopily dot org>
	  * gst/udp/gstmultiudpsink.c: (join_multicast),
	  (gst_multiudpsink_init_send), (gst_multiudpsink_add):
	  If a destination is added before the stream is set to PAUSED, the
	  multicast group is not joined as the socket is not created yet.
	  Also TTL and LOOP should also be set. Fixes #346921.

2006-07-10 09:57:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Extract comment information!!
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
	  Extract comment information!!

2006-07-10 09:46:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Extract year/date information (fixes #347079).
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta),
	  (qtdemux_tag_add_date):
	  Extract year/date information (fixes #347079).

2006-07-08 22:41:25 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximage/gstximagesrc.*: Fix use-damage property to actually work :)
	  Original commit message from CVS:
	  2006-07-09  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	  (gst_ximage_src_set_property), (gst_ximage_src_get_property),
	  (gst_ximage_src_get_caps), (gst_ximage_src_class_init),
	  (gst_ximage_src_init):
	  * sys/ximage/gstximagesrc.h:
	  Fix use-damage property to actually work :)
	  Add startx, starty, endx, endy properties so screencasts other than full
	  screen ones can work.

2006-07-08 19:03:54 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximage/gstximagesrc.*: Add use_damage property to offer ability to choose whether to use
	  Original commit message from CVS:
	  2006-07-08  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	  (gst_ximage_src_set_property), (gst_ximage_src_get_property),
	  (gst_ximage_src_class_init), (gst_ximage_src_init):
	  * sys/ximage/gstximagesrc.h:
	  Add use_damage property to offer ability to choose whether to use
	  XDamage or not.

2006-07-07 15:04:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/goom/filters.c: Avoid goom coredumping by clearing memory.
	  Original commit message from CVS:
	  * gst/goom/filters.c: (zoomFilterSetResolution):
	  Avoid goom coredumping by clearing memory.
	  Fixes 345679.

2006-07-07 14:30:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Don't crash on twos/sowt/raw audio. #345830.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  Don't crash on twos/sowt/raw audio. #345830.

2006-07-05 20:21:02 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/vs6/libgstid3demux.dsp: Add a link to libgsttag-0.10.lib.
	  Original commit message from CVS:
	  * win32/vs6/libgstid3demux.dsp:
	  Add a link to libgsttag-0.10.lib.

2006-07-05 14:52:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Don't return FLOW_UNEXPECTED when a buffer is before the start of the stream (which might happen with large ID3...
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
	  (gst_tag_demux_read_range):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
	  (gst_id3demux_read_range):
	  Don't return FLOW_UNEXPECTED when a buffer is before
	  the start of the stream (which might happen with
	  large ID3v2 tags if the tag reading was done pullrange
	  based and we then switched to push mode later on).
	  Fixes regression introduced by commit from June 29th.

2006-07-05 10:14:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gstid3v2mux.cc: Make UTF-8 the default encoding when writing string tags (before, our UTF-8 strings would ...
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc:
	  Make UTF-8 the default encoding when writing string
	  tags (before, our UTF-8 strings would automatically
	  be converted to ISO-8859-1 by taglib and written as
	  ISO-8859-1 fields if that was possible).
	  * tests/check/elements/id3v2mux.c: (utf8_string_in_buf),
	  (test_taglib_id3mux_check_tag_buffer), (identity_cb),
	  (test_taglib_id3mux_with_tags):
	  Add test case that makes sure our UTF-8 strings have
	  actually been written into the tag as UTF-8.

2006-07-04 16:00:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Let's try that again.
	  Original commit message from CVS:
	  * configure.ac:
	  Let's try that again.

2006-07-04 15:40:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Disable monoscope plugin for now until it fulfills all the requirements.
	  Original commit message from CVS:
	  * configure.ac:
	  Disable monoscope plugin for now until it fulfills
	  all the requirements.

2006-07-03 20:35:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Port monoscope visualisation to 0.10.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/monoscope/Makefile.am:
	  * gst/monoscope/gstmonoscope.c: (gst_monoscope_base_init),
	  (gst_monoscope_class_init), (gst_monoscope_init),
	  (gst_monoscope_finalize), (gst_monoscope_reset),
	  (gst_monoscope_sink_setcaps), (gst_monoscope_src_setcaps),
	  (gst_monoscope_src_negotiate), (get_buffer), (gst_monoscope_chain),
	  (gst_monoscope_sink_event), (gst_monoscope_src_event),
	  (gst_monoscope_change_state), (plugin_init):
	  * gst/monoscope/gstmonoscope.h:
	  Port monoscope visualisation to 0.10.

2006-07-03 20:02:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Fix silly crasher in state change function; add
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
	  (gst_qtdemux_loop_state_header), (qtdemux_video_caps):
	  Fix silly crasher in state change function; add
	  IV41 fourcc (see bug #171111); don't output confusing
	  debug message when skipping atoms.

2006-07-03 16:43:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Return FLOW_UNEXPECTED when at the end of the file, not
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Return FLOW_UNEXPECTED when at the end of the file, not
	  FLOW_ERROR. Fixes 'internal stream error' errors that
	  would sometimes occur in totem when scrubbing to the
	  end of an ID3v1 tagged mp3 file.

2006-07-03 15:31:22 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/libpng/gstpngdec.*: Implement buffer clipping/dropping using GstSegment.
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_init), (user_info_callback),
	  (buffer_clip), (user_end_callback), (gst_pngdec_chain),
	  (gst_pngdec_sink_event), (gst_pngdec_change_state):
	  * ext/libpng/gstpngdec.h:
	  Implement buffer clipping/dropping using GstSegment.
	  This provides accurate seeking.

2006-07-03 15:28:48 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.*: Proper aggregation of each stream's GstFlowReturn in order to figure out whether the task shou...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	  (gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
	  (gst_avi_demux_process_next_entry), (push_tag_lists),
	  (gst_avi_demux_stream_data), (gst_avi_demux_loop):
	  * gst/avi/gstavidemux.h:
	  Proper aggregation of each stream's GstFlowReturn in order to figure out
	  whether the task should stop or not.
	  Don't send inline events before pushing out a NEW_SEGMENT, more
	  specifically for GST_TAG_EVENT.
	  Change a GST_ERROR to a GST_WARNING for a non-fatal situation in reading
	  sub-indexes.

2006-06-30 07:11:24 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/gstsunaudiomixerctrl.c: Move "Monitor" slider to input tab so it works more like sdtaudiocontrol, which ...
	  Original commit message from CVS:
	  Patch by: Brian Cameron  <brian dot cameron at sun dot com>
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_build_list):
	  Move "Monitor" slider to input tab so it works more like
	  sdtaudiocontrol, which is what people on Solaris are used
	  to using for their mixer program (#346259).

2006-06-29 14:50:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/check/elements/level.c: fix a leak, clean up at the end
	  Original commit message from CVS:
	  * tests/check/elements/level.c: (GST_START_TEST):
	  fix a leak, clean up at the end

2006-06-29 11:41:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Send tag event after newsegment event.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_send_event),
	  (gst_matroska_demux_loop_stream_parse_id):
	  * gst/matroska/matroska-ids.h:
	  Send tag event after newsegment event.

2006-06-29 11:11:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/gstid3demux.c: Make sure we don't return GST_FLOW_OK with a NULL buffer in certain cases where a read be...
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
	  (gst_id3demux_read_range):
	  Make sure we don't return GST_FLOW_OK with a NULL buffer in
	  certain cases where a read beyond the end of the file is
	  requested. Fixes #345930.
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
	  (gst_tag_demux_read_range):
	  Fix same issue here as well.

2006-06-29 11:05:14 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximage/gstximagesrc.c: Fix hypothetical crash.
	  Original commit message from CVS:
	  2006-06-29  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
	  Fix hypothetical crash.

2006-06-28 08:36:30 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/gstsunaudiosink.c: Do not modify the ports value. If the user has turned off the built-in speakers, then...
	  Original commit message from CVS:
	  Patch by: Brian Cameron  <brian dot cameron at sun dot com>
	  * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
	  Do not modify the ports value. If the user has turned off the
	  built-in speakers, then we should not reset it in the prepare
	  function, since this causes the built-in speakers to turn
	  back on anytime the user changes a track in totem, rhythmbox,
	  etc. (#346066).

2006-06-23 09:35:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/goom/gstgoom.c: Fix double caps unref when negotiation fails.
	  Original commit message from CVS:
	  * gst/goom/gstgoom.c: (gst_goom_src_negotiate):
	  Fix double caps unref when negotiation fails.

2006-06-22 19:31:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) plus two minor macro fixes.
	  Original commit message from CVS:
	  * ext/annodex/gstcmmldec.c:
	  * ext/annodex/gstcmmlenc.c:
	  * ext/annodex/gstcmmlparser.c:
	  * ext/dv/gstdvdec.c:
	  * ext/dv/gstdvdemux.c:
	  * ext/gdk_pixbuf/pixbufscale.c:
	  * ext/jpeg/gstjpegenc.c:
	  * ext/jpeg/gstsmokedec.c:
	  * ext/jpeg/gstsmokeenc.c:
	  * ext/libpng/gstpngdec.c:
	  * ext/libpng/gstpngenc.c:
	  * ext/speex/gstspeexenc.c:
	  * gst/alpha/gstalphacolor.c:
	  * gst/cutter/gstcutter.c:
	  * gst/debug/gstnavigationtest.c:
	  * gst/icydemux/gsticydemux.c:
	  * gst/level/gstlevel.c:
	  * gst/multipart/multipart.c:
	  * gst/rtp/gstrtpamrpay.c:
	  * gst/rtp/gstrtpdepay.c:
	  * gst/rtp/gstrtpilbcpay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmp4vpay.c:
	  * gst/rtsp/gstrtpdec.c:
	  * gst/rtsp/gstrtspsrc.c:
	  * gst/udp/gstdynudpsink.c:
	  * gst/udp/gstmultiudpsink.c:
	  * gst/udp/gstudpsrc.c:
	  * gst/videobox/gstvideobox.c:
	  * gst/videofilter/gstvideoflip.c:
	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
	  plus two minor macro fixes.

2006-06-22 16:27:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Try to fix up broken matroska files containing subtitle streams with non-UTF8 character encodings (cou...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_check_subtitle_buffer),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_subtitle_caps):
	  * gst/matroska/matroska-ids.c:
	  (gst_matroska_track_init_subtitle_context):
	  * gst/matroska/matroska-ids.h:
	  Try to fix up broken matroska files containing subtitle
	  streams with non-UTF8 character encodings (courtesy of
	  mkvmerge) using either the encoding specified in the
	  GST_SUBTITLE_ENCODING environment variable or the
	  current locale's character set if it is non-UTF8.
	  Fixes #337076.

2006-06-22 12:17:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: Set image type from APIC frame as "image-type" field of GST_TAG_IMAGE buffer caps (#344605).
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Set image type from APIC frame as "image-type" field
	  of GST_TAG_IMAGE buffer caps (#344605).

2006-06-20 19:40:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/: Support chain-based operation, should make flac-over-DAAP work (#340492).
	  Original commit message from CVS:
	  * ext/flac/Makefile.am:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_init),
	  (gst_flac_dec_reset_decoders),
	  (gst_flac_dec_setup_seekable_decoder),
	  (gst_flac_dec_setup_stream_decoder), (gst_flac_dec_finalize),
	  (gst_flac_dec_metadata_callback),
	  (gst_flac_dec_metadata_callback_seekable),
	  (gst_flac_dec_metadata_callback_stream),
	  (gst_flac_dec_error_callback),
	  (gst_flac_dec_error_callback_seekable),
	  (gst_flac_dec_error_callback_stream), (gst_flac_dec_read_seekable),
	  (gst_flac_dec_read_stream), (gst_flac_dec_write),
	  (gst_flac_dec_write_seekable), (gst_flac_dec_write_stream),
	  (gst_flac_dec_loop), (gst_flac_dec_sink_event),
	  (gst_flac_dec_chain), (gst_flac_dec_convert_sink),
	  (gst_flac_dec_get_sink_query_types), (gst_flac_dec_sink_query),
	  (gst_flac_dec_get_src_query_types), (gst_flac_dec_src_query),
	  (gst_flac_dec_handle_seek_event), (gst_flac_dec_sink_activate),
	  (gst_flac_dec_sink_activate_push),
	  (gst_flac_dec_sink_activate_pull), (gst_flac_dec_change_state):
	  * ext/flac/gstflacdec.h:
	  Support chain-based operation, should make flac-over-DAAP
	  work (#340492).

2006-06-20 15:35:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/gst-plugins-good-plugins-sections.txt: Doc updates, merge some unused symbols.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Doc updates, merge some unused symbols.

2006-06-20 14:57:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Added documentation for the rtsp plugin. Fixes #345393.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
	  * gst/rtsp/gstrtspsrc.c:
	  * gst/rtsp/gstrtspsrc.h:
	  Added documentation for the rtsp plugin. Fixes #345393.

2006-06-20 12:10:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspconnection.c: Use better G_OS_* macros. Fixes #345301 some more.
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
	  (rtsp_connection_close), (rtsp_connection_free):
	  Use better G_OS_* macros. Fixes #345301 some more.

2006-06-20 10:35:48 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/: Add a SunAudio source plugin.
	  Original commit message from CVS:
	  Patch by: Brian Cameron <brian dot cameron at sun dot com>
	  * sys/sunaudio/Makefile.am:
	  * sys/sunaudio/gstsunaudio.c: (plugin_init):
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new),
	  (gst_sunaudiomixer_ctrl_list_tracks),
	  (gst_sunaudiomixer_ctrl_get_volume),
	  (gst_sunaudiomixer_ctrl_set_volume),
	  (gst_sunaudiomixer_ctrl_set_mute),
	  (gst_sunaudiomixer_ctrl_set_record):
	  * sys/sunaudio/gstsunaudiomixerctrl.h:
	  * sys/sunaudio/gstsunaudiomixertrack.c:
	  (gst_sunaudiomixer_track_init), (gst_sunaudiomixer_track_new):
	  * sys/sunaudio/gstsunaudiomixertrack.h:
	  * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose),
	  (gst_sunaudiosrc_base_init), (gst_sunaudiosrc_class_init),
	  (gst_sunaudiosrc_init), (gst_sunaudiosrc_set_property),
	  (gst_sunaudiosrc_get_property), (gst_sunaudiosrc_getcaps),
	  (gst_sunaudiosrc_open), (gst_sunaudiosrc_close),
	  (gst_sunaudiosrc_prepare), (gst_sunaudiosrc_unprepare),
	  (gst_sunaudiosrc_read), (gst_sunaudiosrc_delay),
	  (gst_sunaudiosrc_reset):
	  * sys/sunaudio/gstsunaudiosrc.h:
	  Add a SunAudio source plugin.
	  Support stereo and right/left channel gain in the mixer plugin.
	  Support the RECORD flag so that you can switch between line-input and
	  microphone in gnome-volume-control.
	  Code cleanups like using an enumerator for track number instead of an
	  integer. Fixes #344923.

2006-06-20 10:31:41 +0000  Joni Valtanen <joni.valtanen@movial.fi>

	  gst/rtsp/rtspconnection.c: Make RTSP plugin compile on windows. Fixes #345301.
	  Original commit message from CVS:
	  Patch by: Joni Valtanen <joni dot valtanen at movial dot fi>
	  * gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
	  (rtsp_connection_close):
	  Make RTSP plugin compile on windows. Fixes #345301.
	  Some changes to original patch to catch errors better.
	  use ifdef WIN32 instead of ifndef.

2006-06-19 10:00:18 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  configure.ac: If we have libraw1394 >= 1.2.1, then we need libiec61883.
	  Original commit message from CVS:
	  2006-06-19  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * configure.ac:
	  If we have libraw1394 >= 1.2.1, then we need libiec61883.

2006-06-18 14:00:19 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/jpeg/gstjpegdec.c: After a failed buffer alloc, we need to abort the jpeg decoding (it started when parsing heade...
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
	  After a failed buffer alloc, we need to abort the jpeg decoding (it
	  started when parsing headers to figure out how many bytes we need
	  to request downstream).

2006-06-18 12:37:12 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/wavparse/gstwavparse.c: Make sure we don't read beyond the end of the file (#345232).
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet be>
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek):
	  Make sure we don't read beyond the end of the file (#345232).

2006-06-17 14:35:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Fix --disable-external (can't set conditionals conditionally, #343602).
	  Original commit message from CVS:
	  * configure.ac:
	  Fix --disable-external (can't set conditionals conditionally,
	  #343602).

2006-06-16 12:35:08 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  gst/spectrum/Makefile.am: Fix build.
	  Original commit message from CVS:
	  2006-06-16  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * gst/spectrum/Makefile.am:
	  Fix build.

2006-06-16 10:56:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use GST_PLUGIN_DOCS, --enable-plugin-docs etc.
	  Original commit message from CVS:
	  * autogen.sh:
	  * configure.ac:
	  * docs/Makefile.am:
	  Use GST_PLUGIN_DOCS, --enable-plugin-docs etc.
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/inspect/plugin-taglib.xml:
	  Add/fix apev2mux docs.

2006-06-16 09:49:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/: port to use message to get results, cleanly exit when closing the window
	  Original commit message from CVS:
	  * gst/spectrum/demo-audiotest.c: (on_window_destroy),
	  (draw_spectrum), (message_handler), (main):
	  * gst/spectrum/demo-osssrc.c: (on_window_destroy), (draw_spectrum),
	  (message_handler), (main):
	  port to use message to get results, cleanly exit when closing the window
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
	  (gst_spectrum_init), (gst_spectrum_dispose),
	  (gst_spectrum_set_property), (gst_spectrum_get_property),
	  (gst_spectrum_set_caps), (gst_spectrum_start),
	  (gst_spectrum_message_new), (gst_spectrum_transform_ip):
	  * gst/spectrum/gstspectrum.h:
	  port to derive from basetransform and send results via messages
	  (like level element)

2006-06-15 15:58:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Combine return values from src pad pushes.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
	  (gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie),
	  (gst_qtdemux_loop), (gst_qtdemux_chain), (qtdemux_parse_trak):
	  Combine return values from src pad pushes.

2006-06-15 08:50:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Don't crash on files with 0 samples, EOS immediatly instead.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header),
	  (gst_qtdemux_prepare_current_sample), (gst_qtdemux_advance_sample),
	  (gst_qtdemux_add_stream):
	  Don't crash on files with 0 samples, EOS immediatly instead.
	  Fixes #344944.

2006-06-14 15:59:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/dv/gstdvdec.c: Reset segment info on flush.
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c: (gst_dvdec_class_init), (gst_dvdec_init),
	  (gst_dvdec_finalize), (gst_dvdec_sink_event),
	  (gst_dvdec_change_state):
	  Reset segment info on flush.
	  Alloc segment in _init, free in _finalize.
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek):
	  Don't send segments twice.

2006-06-14 15:07:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/dv/gstdvdemux.c: Respect segment.stop. Fixes #342592.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame):
	  Respect segment.stop. Fixes #342592.

2006-06-14 11:28:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: No language specified means the implied language is English according to the matroska ...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
	  No language specified means the implied language is English
	  according to the matroska spec (partially fixes #344708);
	  add some more debug output.

2006-06-14 09:32:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/wavpack/gstwavpackenc.*: Use bitrate property solely for bitrates and add new bits-per-sample property for the ot...
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge  <slomo at circular-chaos org>
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_base_init),
	  (gst_wavpack_enc_class_init), (gst_wavpack_enc_set_wp_config),
	  (gst_wavpack_enc_chain), (gst_wavpack_enc_sink_event),
	  (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
	  * ext/wavpack/gstwavpackenc.h:
	  Use bitrate property solely for bitrates and add new
	  bits-per-sample property for the other stuff. Set duration
	  to 'unknown' in initial header and resend header with proper
	  duration on EOS; update Sebastian's e-mail address.

2006-06-14 08:06:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: When operating chain-based, don't make any assumptions about the chunking of the incoming...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_peek_chunk_info),
	  (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
	  (gst_wavparse_chain):
	  When operating chain-based, don't make any assumptions about the
	  chunking of the incoming data and make streaming work on days other
	  than the second Thursday after a full moon. Also fix up debug
	  messages here and there and make use of the most excellent new
	  gst_pad_query_peer_duration() utility function.
	  Skip any 'bext' chunks in front of the 'fmt ' chunk. Fixes #343837.
	  * gst/wavparse/gstwavparse.h:
	  Remove trailing comma after last enum value, some compilers don't
	  like that.

2006-06-13 17:05:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Handle premature EOS gracefully.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_data):
	  Handle premature EOS gracefully.

2006-06-13 09:54:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Prevent out of bounds array access when scrubbing towards the end of the file between the last...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
	  Prevent out of bounds array access when scrubbing towards
	  the end of the file between the last index entry and the
	  end. Fixes occasional 'start <= stop' newsegment event
	  assertions when scrubbing in MJPEG files.

2006-06-12 11:13:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/.cvsignore: And another one.
	  Original commit message from CVS:
	  * tests/check/elements/.cvsignore:
	  And another one.

2006-06-12 11:04:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/spectrum/.cvsignore: Ignore more.
	  Original commit message from CVS:
	  * gst/spectrum/.cvsignore:
	  Ignore more.

2006-06-12 10:53:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/libmms/gstmms.c: Set caps on outgoing buffers.
	  Original commit message from CVS:
	  * ext/libmms/gstmms.c: (gst_mms_create):
	  Set caps on outgoing buffers.
	  * sys/directdraw/gstdirectdrawsink.c: (gst_directdrawsink_init):
	  Comment out unused global instance variable.

2006-06-11 19:31:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: Extract images from ID3v2 tags (APIC frames). Fixes #339704.
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	  (scan_encoded_string), (parse_picture_frame):
	  Extract images from ID3v2 tags (APIC frames). Fixes #339704.
	  * configure.ac:
	  Require core >= 0.10.8 (for GST_TAG_IMAGE and
	  GST_TAG_PPEVIEW_IMAGE used in the patch above).

2006-06-11 18:56:24 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/raw1394/.gitignore:
	* ext/taglib/.gitignore:
	* tests/check/elements/.gitignore:
	* tests/examples/level/.gitignore:
	  moap ignore
	  Original commit message from CVS:
	  moap ignore

2006-06-11 18:52:19 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/raw1394/gstdv1394src.c: gratuitous comment changes
	  Original commit message from CVS:
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_discover_avc_node):
	  gratuitous comment changes
	  * tests/check/elements/level.c: (GST_START_TEST):
	  fix level test leaks

2006-06-11 18:44:54 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* .gitignore:
	  ignore more
	  Original commit message from CVS:
	  ignore more

2006-06-11 18:20:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Use gst_pad_query_peer_duration() utility function here.
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size):
	  * gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size):
	  Use gst_pad_query_peer_duration() utility function here.

2006-06-11 17:08:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  update build files
	  Original commit message from CVS:
	  * autogen.sh:
	  * configure.ac:
	  * ext/a52dec/Makefile.am:
	  * ext/dvdnav/Makefile.am:
	  * ext/dvdread/Makefile.am:
	  * ext/lame/Makefile.am:
	  * ext/mad/Makefile.am:
	  * ext/mpeg2dec/Makefile.am:
	  * ext/sidplay/Makefile.am:
	  update build files

2006-06-11 13:57:19 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  autogen.sh: require am17
	  Original commit message from CVS:
	  * autogen.sh:
	  require am17
	  * configure.ac:
	  * ext/annodex/Makefile.am:
	  * ext/cdio/Makefile.am:
	  * ext/dv/Makefile.am:
	  * ext/esd/Makefile.am:
	  * ext/flac/Makefile.am:
	  * ext/gdk_pixbuf/Makefile.am:
	  * ext/ladspa/Makefile.am:
	  * ext/libcaca/Makefile.am:
	  * ext/speex/Makefile.am:
	  * ext/taglib/Makefile.am:
	  * sys/oss/Makefile.am:
	  * sys/sunaudio/Makefile.am:
	  * sys/ximage/Makefile.am:
	  clean up build further

2006-06-11 13:55:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* win32/common/config.h:
	  update
	  Original commit message from CVS:
	  update

2006-06-10 15:33:18 +0000  Sebastian Dröge <mail@slomosnail.de>

	  ext/wavpack/: Add wavpack encoder element (#343131).
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge  <mail at slomosnail de>
	  * ext/wavpack/Makefile.am:
	  * ext/wavpack/gstwavpack.c: (plugin_init):
	  * ext/wavpack/gstwavpackcommon.h:
	  * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type),
	  (gst_wavpack_enc_correction_mode_get_type),
	  (gst_wavpack_enc_joint_stereo_mode_get_type),
	  (gst_wavpack_enc_base_init), (gst_wavpack_enc_class_init),
	  (gst_wavpack_enc_init), (gst_wavpack_enc_dispose),
	  (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
	  (gst_wavpack_enc_format_samples), (gst_wavpack_enc_push_block),
	  (gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block),
	  (gst_wavpack_enc_sink_event), (gst_wavpack_enc_change_state),
	  (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property),
	  (gst_wavpack_enc_plugin_init):
	  * ext/wavpack/gstwavpackenc.h:
	  * ext/wavpack/md5.c:
	  * ext/wavpack/md5.h:
	  Add wavpack encoder element (#343131).

2006-06-09 20:36:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gconf/Makefile.am: Honour --disable-schemas-install configure option. Fixes #344100.
	  Original commit message from CVS:
	  * gconf/Makefile.am:
	  Honour --disable-schemas-install configure option. Fixes #344100.

2006-06-09 18:33:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/examples/level/Makefile.am: Add -lm to LIBS for pow() function, don't assume one of our dependencies (such as l...
	  Original commit message from CVS:
	  * tests/examples/level/Makefile.am:
	  Add -lm to LIBS for pow() function, don't assume one of our
	  dependencies (such as libxml-2.0) drags it in automatically
	  (#343603).

2006-06-09 18:17:23 +0000  Peter Kjellerstedt <pkj@axis.com>

	  configure.ac: We should use $SED and not $(SED) in configure.ac (#343678).
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis dot com>
	  * configure.ac:
	  We should use $SED and not $(SED) in configure.ac (#343678).

2006-06-09 17:38:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Check for X before using X_CFLAGS in the check for opengl (#343866).
	  Original commit message from CVS:
	  * configure.ac:
	  Check for X before using X_CFLAGS in the check for opengl (#343866).
	  * ext/musepack/Makefile.am:
	  * ext/wavpack/Makefile.am:
	  * gst/speed/Makefile.am:
	  Add missing GST_LIBS, fixes build on cygwin (#343866).

2006-06-09 17:29:08 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/: Attached find a patch that fixes a number of bugs with the SunAudio mixer plugin and fixes #344101: 1....
	  Original commit message from CVS:
	  Patch by: Brian Cameron <brian dot cameron at sun dot com>
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_open), (gst_sunaudiomixer_ctrl_build_list),
	  (gst_sunaudiomixer_ctrl_new), (gst_sunaudiomixer_ctrl_set_volume),
	  (gst_sunaudiomixer_ctrl_set_mute):
	  * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init),
	  (gst_sunaudiosink_init), (gst_sunaudiosink_prepare),
	  (gst_sunaudiosink_write):
	  Attached find a patch that fixes a number of bugs with the SunAudio mixer
	  plugin and fixes #344101:
	  1. The gst_sunaudiomixer_ctrl_build_list kept appending the same 3 tracks onto
	  the tracklist causing gnome-volume-control's preferences dialog to be messed
	  up and would core dump if you checked/unchecked any item.
	  2. We weren't previously setting the MUTE flag properly.  Fixing this makes
	  gnome-volume-control work better.
	  3. Now we properly define the input track to be GST_MIXER_TRACK_INPUT and
	  the monitor to be GST_MIXER_TRACK_OUTPUT, so that makes gnome-volume-control
	  look better.
	  Also some minor cleanup in gstsunaudiosink.c.

2006-06-09 17:12:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/jpeg/gstjpegdec.*: API: Added IDCT method property
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_idct_method_get_type),
	  (gst_jpeg_dec_class_init), (gst_jpeg_dec_init),
	  (gst_jpeg_dec_decode_indirect), (gst_jpeg_dec_decode_direct),
	  (gst_jpeg_dec_chain), (gst_jpeg_dec_sink_event),
	  (gst_jpeg_dec_set_property), (gst_jpeg_dec_get_property):
	  * ext/jpeg/gstjpegdec.h:
	  API: Added IDCT method property
	  Small cleanups.
	  Avoid dynamic allocation of trivial fixed structure.
	  Allocate enough space for temp 4:4:4 YUV buffers. Fixes #343661.

2006-06-07 09:25:16 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  configure.ac: We now require libraw1394 >= 1.1.0 and that version onwards all have .pc files.
	  Original commit message from CVS:
	  2006-06-07  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * configure.ac:
	  We now require libraw1394 >= 1.1.0 and that version onwards all
	  have .pc files.

2006-06-02 15:02:54 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/law/alaw-decode.c: Trying to get items from an ANY or EMPTY caps is ... stupid.
	  Original commit message from CVS:
	  * gst/law/alaw-decode.c: (alawdec_getcaps):
	  Trying to get items from an ANY or EMPTY caps is ... stupid.

2006-06-02 11:33:18 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/dv/gstdvdec.*: Added GstSegment handling, now implements dropping/clipping.
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_event),
	  (gst_dvdec_chain), (gst_dvdec_change_state):
	  * ext/dv/gstdvdec.h:
	  Added GstSegment handling, now implements dropping/clipping.

2006-06-01 22:00:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
	  Original commit message from CVS:
	  * ext/alsaspdif/alsaspdifsink.h:
	  * ext/amrwb/gstamrwbdec.h:
	  * ext/amrwb/gstamrwbenc.h:
	  * ext/amrwb/gstamrwbparse.h:
	  * ext/arts/gst_arts.h:
	  * ext/artsd/gstartsdsink.h:
	  * ext/audiofile/gstafparse.h:
	  * ext/audiofile/gstafsink.h:
	  * ext/audiofile/gstafsrc.h:
	  * ext/audioresample/gstaudioresample.h:
	  * ext/bz2/gstbz2dec.h:
	  * ext/bz2/gstbz2enc.h:
	  * ext/dirac/gstdiracdec.h:
	  * ext/directfb/dfbvideosink.h:
	  * ext/divx/gstdivxdec.h:
	  * ext/divx/gstdivxenc.h:
	  * ext/dts/gstdtsdec.h:
	  * ext/faac/gstfaac.h:
	  * ext/gsm/gstgsmdec.h:
	  * ext/gsm/gstgsmenc.h:
	  * ext/ivorbis/vorbisenc.h:
	  * ext/libfame/gstlibfame.h:
	  * ext/nas/nassink.h:
	  * ext/neon/gstneonhttpsrc.h:
	  * ext/polyp/polypsink.h:
	  * ext/sdl/sdlaudiosink.h:
	  * ext/sdl/sdlvideosink.h:
	  * ext/shout/gstshout.h:
	  * ext/snapshot/gstsnapshot.h:
	  * ext/sndfile/gstsf.h:
	  * ext/swfdec/gstswfdec.h:
	  * ext/tarkin/gsttarkindec.h:
	  * ext/tarkin/gsttarkinenc.h:
	  * ext/theora/theoradec.h:
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackparse.h:
	  * ext/xine/gstxine.h:
	  * ext/xvid/gstxviddec.h:
	  * ext/xvid/gstxvidenc.h:
	  * gst/cdxaparse/gstcdxaparse.h:
	  * gst/cdxaparse/gstcdxastrip.h:
	  * gst/colorspace/gstcolorspace.h:
	  * gst/festival/gstfestival.h:
	  * gst/freeze/gstfreeze.h:
	  * gst/gdp/gstgdpdepay.h:
	  * gst/gdp/gstgdppay.h:
	  * gst/modplug/gstmodplug.h:
	  * gst/mpeg1sys/gstmpeg1systemencode.h:
	  * gst/mpeg1videoparse/gstmp1videoparse.h:
	  * gst/mpeg2sub/gstmpeg2subt.h:
	  * gst/mpegaudioparse/gstmpegaudioparse.h:
	  * gst/multifilesink/gstmultifilesink.h:
	  * gst/overlay/gstoverlay.h:
	  * gst/playondemand/gstplayondemand.h:
	  * gst/qtdemux/qtdemux.h:
	  * gst/rtjpeg/gstrtjpegdec.h:
	  * gst/rtjpeg/gstrtjpegenc.h:
	  * gst/smooth/gstsmooth.h:
	  * gst/smoothwave/gstsmoothwave.h:
	  * gst/spectrum/gstspectrum.h:
	  * gst/speed/gstspeed.h:
	  * gst/stereo/gststereo.h:
	  * gst/switch/gstswitch.h:
	  * gst/tta/gstttadec.h:
	  * gst/tta/gstttaparse.h:
	  * gst/videodrop/gstvideodrop.h:
	  * gst/xingheader/gstxingmux.h:
	  * sys/directdraw/gstdirectdrawsink.h:
	  * sys/directsound/gstdirectsoundsink.h:
	  * sys/dxr3/dxr3audiosink.h:
	  * sys/dxr3/dxr3spusink.h:
	  * sys/dxr3/dxr3videosink.h:
	  * sys/qcam/gstqcamsrc.h:
	  * sys/vcd/vcdsrc.h:
	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass

2006-06-01 22:00:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
	  Original commit message from CVS:
	  * ext/alsaspdif/alsaspdifsink.h:
	  * ext/amrwb/gstamrwbdec.h:
	  * ext/amrwb/gstamrwbenc.h:
	  * ext/amrwb/gstamrwbparse.h:
	  * ext/arts/gst_arts.h:
	  * ext/artsd/gstartsdsink.h:
	  * ext/audiofile/gstafparse.h:
	  * ext/audiofile/gstafsink.h:
	  * ext/audiofile/gstafsrc.h:
	  * ext/audioresample/gstaudioresample.h:
	  * ext/bz2/gstbz2dec.h:
	  * ext/bz2/gstbz2enc.h:
	  * ext/dirac/gstdiracdec.h:
	  * ext/directfb/dfbvideosink.h:
	  * ext/divx/gstdivxdec.h:
	  * ext/divx/gstdivxenc.h:
	  * ext/dts/gstdtsdec.h:
	  * ext/faac/gstfaac.h:
	  * ext/gsm/gstgsmdec.h:
	  * ext/gsm/gstgsmenc.h:
	  * ext/ivorbis/vorbisenc.h:
	  * ext/libfame/gstlibfame.h:
	  * ext/nas/nassink.h:
	  * ext/neon/gstneonhttpsrc.h:
	  * ext/polyp/polypsink.h:
	  * ext/sdl/sdlaudiosink.h:
	  * ext/sdl/sdlvideosink.h:
	  * ext/shout/gstshout.h:
	  * ext/snapshot/gstsnapshot.h:
	  * ext/sndfile/gstsf.h:
	  * ext/swfdec/gstswfdec.h:
	  * ext/tarkin/gsttarkindec.h:
	  * ext/tarkin/gsttarkinenc.h:
	  * ext/theora/theoradec.h:
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackparse.h:
	  * ext/xine/gstxine.h:
	  * ext/xvid/gstxviddec.h:
	  * ext/xvid/gstxvidenc.h:
	  * gst/cdxaparse/gstcdxaparse.h:
	  * gst/cdxaparse/gstcdxastrip.h:
	  * gst/colorspace/gstcolorspace.h:
	  * gst/festival/gstfestival.h:
	  * gst/freeze/gstfreeze.h:
	  * gst/gdp/gstgdpdepay.h:
	  * gst/gdp/gstgdppay.h:
	  * gst/modplug/gstmodplug.h:
	  * gst/mpeg1sys/gstmpeg1systemencode.h:
	  * gst/mpeg1videoparse/gstmp1videoparse.h:
	  * gst/mpeg2sub/gstmpeg2subt.h:
	  * gst/mpegaudioparse/gstmpegaudioparse.h:
	  * gst/multifilesink/gstmultifilesink.h:
	  * gst/overlay/gstoverlay.h:
	  * gst/playondemand/gstplayondemand.h:
	  * gst/qtdemux/qtdemux.h:
	  * gst/rtjpeg/gstrtjpegdec.h:
	  * gst/rtjpeg/gstrtjpegenc.h:
	  * gst/smooth/gstsmooth.h:
	  * gst/smoothwave/gstsmoothwave.h:
	  * gst/spectrum/gstspectrum.h:
	  * gst/speed/gstspeed.h:
	  * gst/stereo/gststereo.h:
	  * gst/switch/gstswitch.h:
	  * gst/tta/gstttadec.h:
	  * gst/tta/gstttaparse.h:
	  * gst/videodrop/gstvideodrop.h:
	  * gst/xingheader/gstxingmux.h:
	  * sys/directdraw/gstdirectdrawsink.h:
	  * sys/directsound/gstdirectsoundsink.h:
	  * sys/dxr3/dxr3audiosink.h:
	  * sys/dxr3/dxr3spusink.h:
	  * sys/dxr3/dxr3videosink.h:
	  * sys/qcam/gstqcamsrc.h:
	  * sys/vcd/vcdsrc.h:
	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass

2006-06-01 21:07:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
	  Original commit message from CVS:
	  * ext/aalib/gstaasink.h:
	  * ext/annodex/gstcmmldec.h:
	  * ext/cairo/gsttimeoverlay.h:
	  * ext/dv/gstdvdec.h:
	  * ext/dv/gstdvdemux.h:
	  * ext/esd/esdmon.h:
	  * ext/esd/esdsink.h:
	  * ext/flac/gstflacenc.h:
	  * ext/gconf/gstgconfaudiosink.h:
	  * ext/gconf/gstgconfaudiosrc.h:
	  * ext/gconf/gstgconfvideosink.h:
	  * ext/gconf/gstgconfvideosrc.h:
	  * ext/gdk_pixbuf/gstgdkanimation.h:
	  * ext/gdk_pixbuf/pixbufscale.h:
	  * ext/hal/gsthalaudiosink.h:
	  * ext/hal/gsthalaudiosrc.h:
	  * ext/jpeg/gstjpegenc.h:
	  * ext/jpeg/gstsmokedec.h:
	  * ext/jpeg/gstsmokeenc.h:
	  * ext/libcaca/gstcacasink.h:
	  * ext/libmng/gstmngdec.h:
	  * ext/libmng/gstmngenc.h:
	  * ext/libpng/gstpngdec.h:
	  * ext/libpng/gstpngenc.h:
	  * ext/raw1394/gstdv1394src.h:
	  * ext/speex/gstspeexenc.h:
	  * gst/autodetect/gstautoaudiosink.h:
	  * gst/autodetect/gstautovideosink.h:
	  * gst/avi/gstavidemux.h:
	  * gst/cutter/gstcutter.h:
	  * gst/debug/efence.h:
	  * gst/debug/gstnavigationtest.h:
	  * gst/debug/gstnavseek.h:
	  * gst/flx/gstflxdec.h:
	  * gst/goom/gstgoom.h:
	  * gst/icydemux/gsticydemux.h:
	  * gst/id3demux/gstid3demux.h:
	  * gst/law/alaw-decode.h:
	  * gst/law/alaw-encode.h:
	  * gst/law/mulaw-decode.h:
	  * gst/law/mulaw-encode.h:
	  * gst/matroska/matroska-mux.h:
	  * gst/median/gstmedian.h:
	  * gst/oldcore/gstaggregator.h:
	  * gst/oldcore/gstfdsink.h:
	  * gst/oldcore/gstmd5sink.h:
	  * gst/oldcore/gstmultifilesrc.h:
	  * gst/oldcore/gstpipefilter.h:
	  * gst/oldcore/gstshaper.h:
	  * gst/oldcore/gststatistics.h:
	  * gst/rtp/gstasteriskh263.h:
	  * gst/rtp/gstrtpL16depay.h:
	  * gst/rtp/gstrtpL16pay.h:
	  * gst/rtp/gstrtpamrdepay.h:
	  * gst/rtp/gstrtpamrpay.h:
	  * gst/rtp/gstrtpdepay.h:
	  * gst/rtp/gstrtpgsmdepay.h:
	  * gst/rtp/gstrtpgsmpay.h:
	  * gst/rtp/gstrtph263pay.h:
	  * gst/rtp/gstrtph263pdepay.h:
	  * gst/rtp/gstrtph263ppay.h:
	  * gst/rtp/gstrtpmp4gpay.h:
	  * gst/rtp/gstrtpmp4vdepay.h:
	  * gst/rtp/gstrtpmp4vpay.h:
	  * gst/rtp/gstrtpmpadepay.h:
	  * gst/rtp/gstrtpmpapay.h:
	  * gst/rtp/gstrtppcmadepay.h:
	  * gst/rtp/gstrtppcmapay.h:
	  * gst/rtp/gstrtppcmudepay.h:
	  * gst/rtp/gstrtppcmupay.h:
	  * gst/rtp/gstrtpspeexdepay.h:
	  * gst/rtp/gstrtpspeexpay.h:
	  * gst/rtsp/gstrtpdec.h:
	  * gst/rtsp/gstrtspsrc.h:
	  * gst/smpte/gstsmpte.h:
	  * gst/udp/gstdynudpsink.h:
	  * gst/udp/gstmultiudpsink.h:
	  * gst/udp/gstudpsink.h:
	  * gst/udp/gstudpsrc.h:
	  * gst/videofilter/gstvideobalance.h:
	  * gst/videofilter/gstvideoflip.h:
	  * sys/oss/gstossdmabuffer.h:
	  * sys/oss/gstossmixerelement.h:
	  * sys/oss/gstosssink.h:
	  * sys/oss/gstosssrc.h:
	  * sys/osxvideo/osxvideosink.h:
	  * sys/sunaudio/gstsunaudiomixer.h:
	  * sys/sunaudio/gstsunaudiosink.h:
	  * sys/ximage/gstximagesrc.h:
	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass

2006-05-31 16:23:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/goom/gstgoom.*: Handle QoS.
	  Original commit message from CVS:
	  * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
	  (gst_goom_finalize), (gst_goom_reset), (gst_goom_sink_setcaps),
	  (gst_goom_src_setcaps), (gst_goom_src_event),
	  (gst_goom_sink_event), (get_buffer), (gst_goom_chain),
	  (gst_goom_change_state):
	  * gst/goom/gstgoom.h:
	  Handle QoS.
	  Handle flushing, discont and events.
	  Fix timestamps and various other cleanups.

2006-05-31 15:37:16 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/raw1394/gstdv1394src.c: Fix bus reset when using libiec61883
	  Original commit message from CVS:
	  2006-05-31  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_bus_reset):
	  Fix bus reset when using libiec61883

2006-05-31 10:31:23 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  configure.ac: Detect libiec61883 and set necessary CFLAGS and LIBS for dv1394.
	  Original commit message from CVS:
	  2006-05-31  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * configure.ac:
	  Detect libiec61883 and set necessary CFLAGS and LIBS for dv1394.
	  * ext/raw1394/Makefile.am:
	  Add CFLAGS.
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_iec61883_receive),
	  New method, to receive using libiec61883.
	  (gst_dv1394src_iso_receive),
	  #ifdef'd out if libiec61883 is present.
	  (gst_dv1394src_bus_reset),
	  Get userdata correctly if using libiec61883.
	  (gst_dv1394src_create),
	  When using libiec61883, only poll one fd and no need to read.
	  (gst_dv1394src_discover_avc_node),
	  Replace g_warnings.
	  (gst_dv1394src_start),
	  Create new handle when we know which dv port.  More reliable
	  than setting port on an existing handle.  Initialise libiec61883.
	  (gst_dv1394src_stop):
	  If using libiec61883, then cleanup its handle properly.
	  * ext/raw1394/gstdv1394src.h:
	  Add libiec61883 handle.

2006-05-30 21:07:38 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/avi/gstavidemux.c: add an explicit dll imported declaration for GST_CAT_EVENT+WIN32
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c:
	  add an explicit dll imported declaration for GST_CAT_EVENT+WIN32
	  * win32/MANIFEST:
	  sort file listing
	  * win32/vs6/libgstavi.dsp:
	  add gstavimux.c to the project
	  * win32/vs6/libgstid3demux.dsp:
	  add link to zlib library
	  * win32/vs6/libgstmatroska.dsp:
	  add matroska-ids.c to the project

2006-05-30 14:35:18 +0000  Sebastian Dröge <mail@slomosnail.de>

	  Add apev2mux element (#343122).
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge  <mail at slomosnail de >
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * ext/taglib/Makefile.am:
	  * ext/taglib/gstapev2mux.cc:
	  * ext/taglib/gstapev2mux.h:
	  * ext/taglib/gstid3v2mux.cc:
	  * ext/taglib/gsttaglibmux.c: (plugin_init):
	  * ext/taglib/gsttaglibmux.h:
	  Add apev2mux element (#343122).
	  * tests/check/Makefile.am:
	  * tests/check/elements/apev2mux.c:
	  (test_taglib_apev2mux_create_tags),
	  (test_taglib_apev2mux_check_tags), (fill_mp3_buffer), (got_buffer),
	  (demux_pad_added), (test_taglib_apev2mux_check_output_buffer),
	  (test_taglib_apev2mux_with_tags), (GST_START_TEST),
	  (apev2mux_suite), (main):
	  Add unit test for apev2mux element.

2006-05-28 17:33:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: GST_PTR_FORMAT should be used to print caps in debug statements.
	  Original commit message from CVS:
	  * gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps):
	  * gst/debug/negotiation.c: (gst_negotiation_update_caps):
	  * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
	  GST_PTR_FORMAT should be used to print caps in debug statements.

2006-05-28 14:38:11 +0000  Sebastian Dröge <slomo@ubuntu.com>

	  gst/apetag/gstapedemux.c: Some clean-ups and additions: map APE 'file' tag to
	  Original commit message from CVS:
	  Patch by: Sebastian Dröge  <slomo at ubuntu dot com>
	  * gst/apetag/gstapedemux.c: (ape_demux_get_gst_tag_from_tag),
	  (ape_demux_parse_tags):
	  Some clean-ups and additions: map APE 'file' tag to
	  GST_TAG_LOCATION (#343123); add support for extracting
	  the track count and clean up parsing a bit (#343127).

2006-05-28 13:49:12 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/jpeg/gstjpegdec.c: Initialize segment to GST_FORMAT_UNDEFINED in READY->PAUSED.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_change_state):
	  Initialize segment to GST_FORMAT_UNDEFINED in READY->PAUSED.

2006-05-28 13:30:13 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/jpeg/gstjpegdec.*: Clip outgoing buffers according to currently configured segment.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_finalize),
	  (gst_jpeg_dec_init), (gst_jpeg_dec_chain),
	  (gst_jpeg_dec_sink_event), (gst_jpeg_dec_change_state):
	  * ext/jpeg/gstjpegdec.h:
	  Clip outgoing buffers according to currently configured segment.

2006-05-28 10:39:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gstid3v2mux.cc: Handle  writing of track-count or album-volume-count without track-number or albume-volume...
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc:
	  Handle  writing of track-count or album-volume-count without
	  track-number or albume-volume-number (in this case the number
	  will just be set to 0).
	  * tests/check/elements/id3v2mux.c: (test_taglib_id3mux_check_tags):
	  It would be nice if we actually checked the values received for
	  track/album-volume number/count in  _check_tags(), rather than
	  setting them again ...

2006-05-28 10:05:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: A track/volume number or count of 0 does not make sense, just ignore it along with negati...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
	  A track/volume number or count of 0 does not make sense,
	  just ignore it along with negative numbers (a tag might
	  only contain a track count without a track number).

2006-05-27 13:11:37 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/jpeg/gstjpegdec.c: Abort decompression when receiving FLUSH_STOP. This should avoid issues when interrupting deco...
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
	  (gst_jpeg_dec_sink_event):
	  Abort decompression when receiving FLUSH_STOP. This should avoid
	  issues when interrupting decoding with flushes.

2006-05-27 12:10:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflac.c: Don't #include file we don't dist any longer.
	  Original commit message from CVS:
	  * ext/flac/gstflac.c:
	  Don't #include file we don't dist any longer.

2006-05-27 11:27:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
	  Original commit message from CVS:
	  * README:
	  Replace current README (containing the release notes from
	  some 0.9.x version) with a proper README taken from the core.

2006-05-26 22:35:00 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/: added another example
	  Original commit message from CVS:
	  * gst/spectrum/Makefile.am:
	  * gst/spectrum/demo-audiotest.c: (on_frequency_changed),
	  (spectrum_chain), (main):
	  * gst/spectrum/demo-osssrc.c:
	  added another example
	  * sys/v4l2/gstv4l2src.c:
	  fix typo

2006-05-26 13:16:54 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Clip the outputed NEWSEGMENT stop time to the configured segment stop time.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment):
	  Clip the outputed NEWSEGMENT stop time to the configured segment stop
	  time.

2006-05-26 11:48:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Don't clear the running variable in the seek code.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_do_seek):
	  Don't clear the running variable in the seek code.

2006-05-24 16:03:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/dv/gstdvdemux.c: Implement EOS correctly by either posting
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
	  Implement EOS correctly by either posting
	  SEGMENT_DONE or pushing an EOS message depending
	  on the seek type. Fixes #342592

2006-05-24 11:56:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Detect QCELP in mp4a descriptors.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_esds):
	  Detect QCELP in mp4a descriptors.

2006-05-24 10:00:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/law/: Some cleanups in the chain functions.
	  Original commit message from CVS:
	  * gst/law/alaw-decode.c: (gst_alawdec_chain):
	  * gst/law/alaw-decode.h:
	  * gst/law/alaw-encode.c: (gst_alawenc_chain):
	  * gst/law/alaw-encode.h:
	  * gst/law/mulaw-decode.c: (gst_mulawdec_chain):
	  * gst/law/mulaw-decode.h:
	  * gst/law/mulaw-encode.c: (gst_mulawenc_chain):
	  * gst/law/mulaw-encode.h:
	  Some cleanups in the chain functions.
	  Remove some GStreamer 0.0.2 bits.

2006-05-23 20:15:04 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/matroska/matroska-mux.c: gst_collect_pads_stop() needs to be called before chaining up to the parent class (#3427...
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet be>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_change_state):
	  gst_collect_pads_stop() needs to be called before chaining up
	  to the parent class (#342734).

2006-05-23 16:45:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/: Remove backwards compatibility cruft for dealing with FLAC API changes in the 1.0.x series - we require 1....
	  Original commit message from CVS:
	  * ext/flac/Makefile.am:
	  * ext/flac/flac_compat.h:
	  * ext/flac/gstflac.c:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_init):
	  * ext/flac/gstflacenc.c:
	  Remove backwards compatibility cruft for dealing with FLAC API
	  changes in the 1.0.x series - we require 1.1.1 or newer these days.

2006-05-23 13:44:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Add support for muxing/demuxing theora video (#342448; too bad none of the usual linux players can act...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_push_xiph_codec_priv_data),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init),
	  (gst_matroska_mux_video_pad_setcaps),
	  (xiph3_streamheader_to_codecdata),
	  (vorbis_streamheader_to_codecdata),
	  (theora_streamheader_to_codecdata),
	  (gst_matroska_mux_audio_pad_setcaps),
	  (gst_matroska_mux_write_data):
	  Add support for muxing/demuxing theora video (#342448; too bad
	  none of the usual linux players can actually play this). Playback
	  in GStreamer will require additional changes to theoradec in -base.
	  Refactor streamheaders <=> CodecPrivateData code a bit; some small
	  cleanups.

2006-05-22 18:00:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: po/POTFILES.in:
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak), (plugin_init):
	  po/POTFILES.in:
	  Throw an error when the file is encrypted. Move plugin_init stuff
	  to the end of the file, add stuff for i18n, make debug category
	  static.

2006-05-22 15:23:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/gstjpegdec.c: Fix crashes when the horizontal subsampling is 1.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (hresamplecpy1),
	  (gst_jpeg_dec_decode_indirect), (gst_jpeg_dec_chain):
	  Fix crashes when the horizontal subsampling is 1.
	  Fixes #342097.

2006-05-22 14:56:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/rtp/gstasteriskh263.h:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtpmp4gpay.h:
	* gst/rtp/gstrtpmp4vdepay.h:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadepay.h:
	* gst/rtp/gstrtpmpapay.h:
	  cover up the dirty truth
	  Original commit message from CVS:
	  cover up the dirty truth

2006-05-22 13:53:18 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/avi/gstavimux.*: - add odml (large file) index support
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet be>
	  * gst/avi/gstavimux.c: (gst_avi_mux_finalize), (gst_avi_mux_init),
	  (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
	  (gst_avi_mux_write_tag), (gst_avi_mux_riff_get_avi_header),
	  (gst_avi_mux_riff_get_avix_header), (gst_avi_mux_write_avix_index),
	  (gst_avi_mux_add_index), (gst_avi_mux_bigfile),
	  (gst_avi_mux_start_file), (gst_avi_mux_stop_file),
	  (gst_avi_mux_handle_event), (gst_avi_mux_do_audio_buffer),
	  (gst_avi_mux_do_video_buffer), (gst_avi_mux_do_one_buffer),
	  (gst_avi_mux_change_state):
	  * gst/avi/gstavimux.h:
	  Some enhancements for avimux (#342526):
	  - add odml (large file) index support
	  - store codec init data (e.g. huffyuv)
	  - miscellaneous other fixes/cleanups

2006-05-22 13:51:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	  fix descriptions and license blocks cut and paste anyone ?
	  Original commit message from CVS:
	  fix descriptions and license blocks
	  cut and paste anyone ?

2006-05-21 16:41:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/gstspectrum.c: Use boilerplate macro, fix strings to match plugin-moval-requirements
	  Original commit message from CVS:
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
	  (gst_spectrum_init), (gst_spectrum_set_sink_caps),
	  (gst_spectrum_get_sink_caps), (gst_spectrum_chain):
	  Use boilerplate macro, fix strings to match plugin-moval-requirements

2006-05-21 16:23:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/spectrum/Makefile.am: Link to base libraries
	  Original commit message from CVS:
	  * gst/spectrum/Makefile.am:
	  Link to base libraries
	  * gst/spectrum/demo-osssrc.c: (main):
	  use new threshhold property
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
	  (gst_spectrum_init), (gst_spectrum_dispose),
	  (gst_spectrum_set_property), (gst_spectrum_set_sink_caps),
	  (gst_spectrum_get_sink_caps), (gst_spectrum_chain),
	  (gst_spectrum_change_state):
	  * gst/spectrum/gstspectrum.h:
	  Use gst_adapter, support multiple-channels, add threshold property for
	  result, add docs, fix resulting spectrum range (was including mirrored
	  results)

2006-05-20 22:42:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Initial port of the spectrum element
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/spectrum/demo-osssrc.c: (spectrum_chain), (main):
	  * gst/spectrum/fix_fft.c: (gst_spectrum_fix_dot):
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_get_type),
	  (gst_spectrum_base_init), (gst_spectrum_class_init),
	  (gst_spectrum_init), (gst_spectrum_dispose),
	  (gst_spectrum_set_property), (gst_spectrum_chain):
	  * gst/spectrum/gstspectrum.h:
	  Initial port of the spectrum element

2006-05-19 18:58:05 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2xoverlay.c:
	  I forget to add sys/v4l2/gstv4l2xoverlay.c in las commit
	  Original commit message from CVS:
	  I forget to add sys/v4l2/gstv4l2xoverlay.c in las commit

2006-05-19 18:31:25 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	  Some clean-ups requested by wingo in bug #338818.
	  Original commit message from CVS:
	  Some clean-ups requested by wingo in bug #338818.

2006-05-19 14:05:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3v2frames.c: Don't output any tag when we encounter a negative track number - the tag type is uint, so...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
	  Don't output any tag when we encounter a negative track number - the
	  tag type is uint, so we end up outputting huge positive numbers
	  instead. (Fixes: #342029)

2006-05-18 23:04:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: update for new GSTPB_PLUGINS_DIR
	  Original commit message from CVS:
	  * configure.ac:
	  update for new GSTPB_PLUGINS_DIR

2006-05-18 19:34:47 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Check for X11
	  Original commit message from CVS:
	  * configure.ac:
	  Check for X11
	  * sys/v4l2/gstv4l2object.c: (gst_v4l2_class_probe_devices):
	  * sys/v4l2/gstv4l2object.h:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_iface_supported):
	  * sys/v4l2/gstv4l2src.h:
	  * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_open):
	  * sys/v4l2/gstv4l2xoverlay.h:
	  Code cleanups, fix debug macros

2006-05-18 14:45:33 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  rtp/gst/gstrtph263pay.c: Properly set static caps for H263 at 34.
	  Original commit message from CVS:
	  2006-05-18  Philippe Kalaf  <philippe.kalaf at collabora.co.uk>
	  * rtp/gst/gstrtph263pay.c:
	  Properly set static caps for H263 at 34.

2006-05-18 12:46:08 +0000  James Doc Livingston <doclivingston@gmail.com>

	  ext/taglib/gsttaglibmux.c: Merge event tags and tag setter tags correctly (#339918). Also, don't leak taglist in case...
	  Original commit message from CVS:
	  Patch by: James "Doc" Livingston  <doclivingston gmail com>
	  * ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag):
	  Merge event tags and tag setter tags correctly (#339918). Also,
	  don't leak taglist in case of an error.

2006-05-17 18:09:06 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	* common:
	* gst/rtp/gstrtph263pay.c:
	  Fixed caps for H263 (not the same as H263+)
	  Original commit message from CVS:
	  Fixed caps for H263 (not the same as H263+)

2006-05-17 12:36:26 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/law/mulaw-decode.c: We can only do caps intersection if the othercaps are non-empty and not
	  Original commit message from CVS:
	  * gst/law/mulaw-decode.c: (mulawdec_getcaps):
	  We can only do caps intersection if the othercaps are non-empty and not
	  ANY. Else we return the pad template (base_caps).

2006-05-17 11:20:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/gstjpegdec.c: Fix crash when outputting debugging information for certain pictures (always good to use the r...
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
	  Fix crash when outputting debugging information for certain
	  pictures (always good to use the right struct member for
	  the number of records in an array).

2006-05-17 08:10:31 +0000  Jindrich Makovicka <jindrich.makivicka@itonis.tv>

	  gst/matroska/ebml-read.c: Don't create unnecessary sub-buffers all the time. Dramatically improves performance with m...
	  Original commit message from CVS:
	  Patch by: Jindrich Makovicka  <jindrich.makivicka at itonis tv>
	  * gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes),
	  (gst_ebml_read_pull_bytes), (gst_ebml_read_element_id),
	  (gst_ebml_read_element_length), (gst_ebml_read_buffer),
	  (gst_ebml_read_bytes), (gst_ebml_read_uint), (gst_ebml_read_sint),
	  (gst_ebml_read_float), (gst_ebml_read_ascii),
	  (gst_ebml_read_binary):
	  Don't create unnecessary sub-buffers all the time. Dramatically
	  improves performance with multiple concurrently running
	  matroskademux instances (#341818) (and avoids doing
	  unnecessarily inefficient things in the general case).

2006-05-16 17:20:04 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/libpng/gstpngenc.c: In snapshot mode, we always return GST_FLOW_UNEXPECTED whatever the return value of gst_pad_p...
	  Original commit message from CVS:
	  * ext/libpng/gstpngenc.c: (gst_pngenc_chain):
	  In snapshot mode, we always return GST_FLOW_UNEXPECTED whatever the
	  return value of gst_pad_push_event().

2006-05-16 14:07:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/autodetect/: Make the name of the child element be based on the name of the parent, so that debug output is more ...
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_find_best):
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_find_best):
	  Make the name of the child element be based on the name of the
	  parent, so that debug output is more useful.
	  * gst/id3demux/id3v2frames.c: (find_utf16_bom),
	  (parse_insert_string_field), (parse_split_strings):
	  Rework string parsing to always walk over BOM markers in UTF16
	  strings, using the endianness indicated by the innermost one,
	  then trying the opposite endianness if that fails to convert
	  to valid UTF-8. Fixes #341774

2006-05-16 13:31:02 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/libpng/Makefile.am: Add LIBPNG_CFLAGS.
	  Original commit message from CVS:
	  2006-05-16  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  Patch from: Matthieu <matthieu at fluendo dot com>
	  * ext/libpng/Makefile.am:
	  Add LIBPNG_CFLAGS.

2006-05-15 11:20:21 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  update with latest changes
	  Original commit message from CVS:
	  update with latest changes

2006-05-15 09:00:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gstid3v2mux.cc: Add support for writing images (APIC frames) into ID3v2 tags (picture type always set to '...
	  Original commit message from CVS:
	  * ext/taglib/gstid3v2mux.cc:
	  Add support for writing images (APIC frames) into ID3v2
	  tags (picture type always set to 'other' for now though).

2006-05-14 12:50:07 +0000  Michael Smith <msmith@xiph.org>

	  gst/wavparse/gstwavparse.c: Update docs; wavparse implements push and pull modes.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c:
	  Update docs; wavparse implements push and pull modes.

2006-05-12 18:10:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Ooops, bitten by the copy-and-paste design paradigm, fixes seek again.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_index_next),
	  (gst_avi_demux_parse_index), (gst_avi_demux_massage_index),
	  (gst_avi_demux_handle_seek), (gst_avi_demux_loop):
	  Ooops, bitten by the copy-and-paste design paradigm, fixes
	  seek again.

2006-05-12 18:04:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.*: Some cleanups, prepare to use GstSegment.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	  (gst_avi_demux_index_next), (gst_avi_demux_handle_src_query),
	  (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_subindex),
	  (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
	  (gst_avi_demux_stream_index), (gst_avi_demux_stream_scan),
	  (gst_avi_demux_massage_index),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_push_event), (gst_avi_demux_stream_header),
	  (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
	  (gst_avi_demux_loop):
	  * gst/avi/gstavidemux.h:
	  Some cleanups, prepare to use GstSegment.
	  Fix error in entry walking code.
	  Fix VBR detection.
	  Smarter timestamp calculation code.
	  Uniform error/eos handling.

2006-05-12 17:44:15 +0000  Michael Smith <msmith@xiph.org>

	  gst/wavparse/gstwavparse.c: Fix use of uninitialised values if we're NOT seeking in ready.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt),
	  (gst_wavparse_perform_seek), (gst_wavparse_stream_headers):
	  Fix use of uninitialised values if we're NOT seeking in ready.
	  Fix typos.

2006-05-12 08:23:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/Makefile.am: Add CFLAGS and LIBS for libgstbase, fixes build on
	  Original commit message from CVS:
	  * gst/wavparse/Makefile.am:
	  Add CFLAGS and LIBS for libgstbase, fixes build on
	  Cygwin (#341489).

2006-05-12 08:21:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: Some more debug info. No need to check whether the string returned by g_convert() is real...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (parse_insert_string_field):
	  Some more debug info. No need to check whether the string
	  returned by g_convert() is really UTF-8 - either it is or
	  we get NULL returned.

2006-05-11 17:59:59 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2colorbalance.c:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2element.c:
	* sys/v4l2/gstv4l2element.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/gstv4l2xoverlay.h:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  Changes proposed by Wingo in bug #338818.
	  Original commit message from CVS:
	  Changes proposed by Wingo in bug #338818.

2006-05-11 09:09:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Figure out the real audio type in mp4a boxes by parsing the optional descriptors in the option...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak),
	  (gst_qtdemux_handle_esds):
	  Figure out the real audio type in mp4a boxes by parsing the
	  optional descriptors in the optional esds box. Promote the
	  default AAC to mp3 when indicated. Fixes #330632.

2006-05-10 17:44:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Parse version 2 sample descriptions.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_dump_unknown),
	  (qtdemux_parse_trak), (gst_qtdemux_handle_esds):
	  Parse version 2 sample descriptions.
	  Don't #define gst_util_dump_mem(), use something more
	  specific instead to avoid confusion.

2006-05-10 13:51:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3v2frames.c: Fix parsing of numeric genre strings some more, by ensuring that we only try and parse st...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3v2_genre_fields_to_taglist):
	  Fix parsing of numeric genre strings some more, by ensuring that
	  we only try and parse strings that a) Start with '(' and b) Consist
	  only of digits.
	  Also, when finding an escaping '((' sequence, bust it back to '(' by
	  swallowing the first parenthesis

2006-05-10 11:17:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/esd/esdsink.*: Move the esd_get_server_info() into gst_esdsink_open() and fail with a decent error message on err...
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_finalize), (gst_esdsink_getcaps),
	  (gst_esdsink_open), (gst_esdsink_close):
	  * ext/esd/esdsink.h:
	  Move the esd_get_server_info() into gst_esdsink_open() and fail
	  with a decent error message on errors.

2006-05-10 10:29:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Const-ify GEnumValue arrays.
	  Original commit message from CVS:
	  * ext/esd/esdmon.c: (gst_esdmon_depths_get_type),
	  (gst_esdmon_channels_get_type):
	  * ext/gconf/gstgconfaudiosink.c: (gst_gconf_profile_get_type):
	  * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_method_get_type):
	  * ext/libcaca/gstcacasink.c: (gst_cacasink_dither_get_type):
	  * ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type):
	  * gst/alpha/gstalpha.c: (gst_alpha_method_get_type):
	  * gst/rtp/gstrtpilbcdepay.c: (gst_ilbc_mode_get_type):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
	  * gst/videobox/gstvideobox.c: (gst_video_box_fill_get_type):
	  * gst/videofilter/gstvideoflip.c: (gst_video_flip_method_get_type):
	  * gst/videomixer/videomixer.c:
	  (gst_video_mixer_background_get_type):
	  Const-ify GEnumValue arrays.

2006-05-09 14:08:15 +0000  Mark Nauwelaerts <manauw@skynet.bet>

	  gst/avi/gstavimux.c: Work around gst_buffer_make_metadata_writable() bug that results in avimux marking all frames in...
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet bet>
	  * gst/avi/gstavimux.c: (gst_avi_mux_do_audio_buffer),
	  (gst_avi_mux_do_video_buffer):
	  Work around gst_buffer_make_metadata_writable() bug that
	  results in avimux marking all frames in the index as
	  keyframes (#340859).

2006-05-08 19:21:18 +0000  Martin Rubli <martin_rubli@logitech.com>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  Fix fourcc name printed out. Patch from Martin Rubli.
	  Original commit message from CVS:
	  Fix fourcc name printed out. Patch from Martin Rubli.

2006-05-08 15:20:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Don't cause side effects in a debugging function.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_query),
	  (qtdemux_dump_mvhd):
	  Don't cause side effects in a debugging function.
	  Also report duration in push mode since we can.

2006-05-08 14:35:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspurl.c: Make parsing of urls suck slightly less.
	  Original commit message from CVS:
	  * gst/rtsp/rtspurl.c: (rtsp_url_parse):
	  Make parsing of urls suck slightly less.

2006-05-08 11:53:03 +0000  Edward Hervey <bilboed@bilboed.com>

	  autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize.
	  Original commit message from CVS:
	  * autogen.sh: (CONFIGURE_DEF_OPT):
	  libtoolize on Darwin/MacOSX is called glibtoolize.

2006-05-08 10:59:05 +0000  Jens Granseuer <jensgr@gmx.net>

	  C89 compliance fixes. Fixes #340980
	  Original commit message from CVS:
	  Patch by: Jens Granseuer <jensgr at gmx dot net>
	  * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_init):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_dispose):
	  C89 compliance fixes. Fixes #340980

2006-05-06 11:38:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.*: Remove tag writing from lame (which was completely broken anyway, #329184). Leaving GstTagSetter ...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_get_type),
	  (gst_lame_release_memory), (gst_lame_init), (gst_lame_sink_event),
	  (gst_lame_setup), (gst_lame_change_state):
	  * ext/lame/gstlame.h:
	  Remove tag writing from lame (which was completely broken
	  anyway, #329184). Leaving GstTagSetter interface around for
	  now, albeit non-functional. Should be removed completely
	  in 0.11. Use the 'id3v2mux' plugin from -good for writing
	  tags.

2006-05-06 09:01:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.*: Handle segment seeks that include the end of the file as stop point properly: when the decoder...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_loop):
	  * ext/flac/gstflacdec.h:
	  Handle segment seeks that include the end of the file as stop point
	  properly: when the decoder hits EOS we want to send a SEGMENT_DONE
	  message instead of an EOS event in case we're in segment seek
	  mode (fixes #340699).

2006-05-06 00:14:09 +0000  Maciej Katafiasz <mathrick@mathrick.org>

	* ChangeLog:
	* ext/cairo/gsttextoverlay.c:
	* ext/flac/gstflacdec.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/wavpack/gstwavpackdec.c:
	* gst/apetag/gstapedemux.c:
	* gst/debug/breakmydata.c:
	* gst/debug/testplugin.c:
	* gst/matroska/ebml-write.c:
	* gst/multipart/multipartdemux.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	  Add semicolons after GST_BOILERPLATE[_FULL] so that indent doesn't mess up following lines.
	  Original commit message from CVS:
	  Add semicolons after GST_BOILERPLATE[_FULL] so that indent doesn't mess up following lines.

2006-05-05 20:12:59 +0000  Martin Rubli <martin_rubli@logitech.com>

	* sys/v4l2/gstv4l2element.c:
	* sys/v4l2/gstv4l2element.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	* tests/icles/v4l2src-test.c:
	  Some changes proposed by wingo in bug #338818 (but not everything yet). Patch from Martin Rubli to fix framerate dete...
	  Original commit message from CVS:
	  Some changes proposed by wingo in bug #338818 (but not everything yet). Patch from Martin Rubli to fix framerate detection.

2006-05-05 08:23:39 +0000  Andres Salomon <dilinger@debian.org>

	  ext/lame/gstlame.c: Fix typo (comma vs. semicolon) (#340710).
	  Original commit message from CVS:
	  Patch by: Andres Salomon  <dilinger at debian org>
	  * ext/lame/gstlame.c: (gst_lame_sink_event):
	  Fix typo (comma vs. semicolon) (#340710).

2006-05-04 17:27:27 +0000  Michal Benes <michal.benes@xeris.cz>

	  gst/matroska/matroska-demux.c: Don't leak caps when freeing the stream context (#340623).
	  Original commit message from CVS:
	  Patch by: Michal Benes  <michal dot benes at xeris dot cz>
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset):
	  Don't leak caps when freeing the stream context (#340623).

2006-05-04 15:40:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS

=== release 0.10.3 ===

2006-05-04 15:36:02 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* win32/common/config.h:
	  Really release 0.10.3
	  Original commit message from CVS:
	  Really release 0.10.3

2006-05-04 15:28:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* docs/plugins/inspect/plugin-qtdemux.xml:
	  Really release 0.10.3 this time
	  Original commit message from CVS:
	  Really release 0.10.3 this time

2006-05-04 15:05:00 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-ladspa.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-qtdemux.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* win32/common/config.h:
	  Release 0.10.3
	  Original commit message from CVS:
	  Release 0.10.3

2006-05-03 18:44:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-05-03 18:41:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-mux.c: Don't strcmp() NULL strings.
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_stream_is_vorbis_header),
	  (gst_matroska_mux_write_data):
	  Don't strcmp() NULL strings.
	  Only start new clusters on video keyframes, not on any
	  random audio buffer that doesn't have the DELTA_UNIT
	  flag set (fixes 'make check' again).

2006-05-03 14:51:50 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/matroska/matroska-mux.c: Don't misinterpret GST_CLOCK_TIME_NONE as very high timestamp value and then dead-lock w...
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet be>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_best_pad),
	  (gst_matroska_mux_stream_is_vorbis_header),
	  (gst_matroska_mux_write_data):
	  Don't misinterpret GST_CLOCK_TIME_NONE as very high timestamp
	  value and then dead-lock when muxing vorbis audio streams
	  (the three vorbis header buffers carry no timestamp, and it
	  would try to mux these after all video buffers). Fixes #340346.
	  Improve clustering: start a new cluster also whenever we get
	  a keyframe.

2006-05-03 14:30:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/qtdemux/qtdemux.c: Clean up one piece of logic slightly and remove a dead code block.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  Clean up one piece of logic slightly and remove a
	  dead code block.

2006-05-03 14:28:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  add win32 stuff
	  Original commit message from CVS:
	  * Makefile.am:
	  * configure.ac:
	  * win32/common/config.h.in:
	  add win32 stuff

2006-05-03 14:26:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  add win32 stuff
	  Original commit message from CVS:
	  * Makefile.am:
	  * configure.ac:
	  * win32/common/config.h.in:
	  add win32 stuff

2006-05-02 22:34:52 +0000  Michael Smith <msmith@xiph.org>

	  ext/cairo/gsttimeoverlay.c: Fix timeoverlay for non-multiple-of-4 widths. This fourcc crap
	  Original commit message from CVS:
	  * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	  Fix timeoverlay for non-multiple-of-4 widths. This fourcc crap
	  SUCKS.

2006-05-02 21:52:48 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	  Fix get_caps func to work when no framerate is available and the caps isn't simple.
	  Original commit message from CVS:
	  Fix get_caps func to work when no framerate is available and the caps isn't simple.

2006-05-02 18:50:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: don't leak caps-string
	  Original commit message from CVS:
	  * gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps):
	  * gst/debug/negotiation.c: (gst_negotiation_update_caps):
	  * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
	  don't leak caps-string

2006-05-02 15:46:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/gstid3demux.c: Let core insert default error message for TYPE_NOT_FOUND errors, it's just as good as our...
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
	  (gst_id3demux_sink_activate):
	  Let core insert default error message for TYPE_NOT_FOUND
	  errors, it's just as good as our own and has the added
	  bonus of being translated.

2006-05-02 15:40:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Post an error message when we get an EOS event and were not able to find out the type of stream.
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_init),
	  (gst_tag_demux_sink_event):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_init),
	  (gst_id3demux_sink_event):
	  Post an error message when we get an EOS event and were not
	  able to find out the type of stream.
	  * tests/check/elements/id3v2mux.c: (fill_mp3_buffer), (got_buffer),
	  (test_taglib_id3mux_with_tags):
	  Decrease num-buffers to 16 per iteration again, otherwise the
	  many memcpy()s and reallocations in the test will hammer slow
	  CPUs completely and make the test timeout.

2006-05-02 13:24:38 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: figure out where plugins-base plugins are
	  Original commit message from CVS:
	  * configure.ac:
	  figure out where plugins-base plugins are
	  * tests/check/Makefile.am:
	  use plugins-base plugins, so we have typefind functions
	  * tests/check/elements/id3v2mux.c: (test_taglib_id3mux_with_tags):
	  increase num-buffers, this makes sure the test errors out instead
	  of timing out when no typefind functions are present

2006-05-02 13:01:50 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/wavparse/gstwavparse.c:
	  fix docs for wavparse
	  Original commit message from CVS:
	  fix docs for wavparse

2006-05-01 21:37:51 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2colorbalance.c:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/v4l2_calls.c:
	* tests/icles/v4l2src-test.c:
	  Few improvements to move to good.
	  Original commit message from CVS:
	  Few improvements to move to good.

2006-05-01 11:46:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  docs/plugins/Makefile.am: also check .cc files for gtk-doc markup
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  also check .cc files for gtk-doc markup
	  * configure.ac:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * tests/check/Makefile.am:
	  * tests/check/elements/id3v2mux.c: (id3v2mux_suite), (main):
	  * ext/Makefile.am:
	  * ext/taglib/Makefile.am:
	  * ext/taglib/gstid3v2mux.h:
	  * ext/taglib/gsttaglibmux.c:
	  * ext/taglib/gsttaglibmux.h:
	  move taglib-based id3v2muxer to -good.  Fixes #336110.

2006-05-01 11:45:15 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-icydemux.xml:
	  add icydemux inspection
	  Original commit message from CVS:
	  add icydemux inspection

2006-05-01 11:43:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/POTFILES.in:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  add ximagesrc for translation
	  Original commit message from CVS:
	  add ximagesrc for translation

2006-04-30 16:16:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/taglib/gstid3v2mux.cc:
	* ext/taglib/gsttaglibmux.c:
	  small cleanups
	  Original commit message from CVS:
	  small cleanups

2006-04-30 15:32:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/taglib/gstid3v2mux.cc:
	  fix docs
	  Original commit message from CVS:
	  fix docs

2006-04-30 14:55:15 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-qtdemux.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	  update to latest version
	  Original commit message from CVS:
	  update to latest version

2006-04-29 18:46:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gsttaglib.cc: Post an error message on the bus in the (extremely unlikely) case of an error.
	  Original commit message from CVS:
	  * ext/taglib/gsttaglib.cc:
	  Post an error message on the bus in the (extremely unlikely)
	  case of an error.

2006-04-29 18:18:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/: Split the actual ID3v2 tag rendering code into its own subclass.
	  Original commit message from CVS:
	  * ext/taglib/Makefile.am:
	  * ext/taglib/gstid3v2mux.cc:
	  * ext/taglib/gstid3v2mux.h:
	  * ext/taglib/gsttaglib.cc:
	  * ext/taglib/gsttaglib.h:
	  Split the actual ID3v2 tag rendering code into
	  its own subclass.

2006-04-29 16:14:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: ... and fix multichannel/WAVFORMATEX support again.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  ... and fix multichannel/WAVFORMATEX support again.

2006-04-28 23:09:17 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.*: Add push (streaming) mode to wavparse (fixes #337625)
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	  (gst_wavparse_class_init), (gst_wavparse_dispose),
	  (gst_wavparse_reset), (gst_wavparse_init),
	  (gst_wavparse_create_sourcepad), (gst_wavparse_parse_adtl),
	  (gst_wavparse_parse_cues), (gst_wavparse_parse_file_header),
	  (gst_wavparse_stream_init), (gst_wavparse_perform_seek),
	  (gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk),
	  (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
	  (gst_wavparse_send_event), (gst_wavparse_add_src_pad),
	  (gst_wavparse_stream_data), (gst_wavparse_loop),
	  (gst_wavparse_chain), (gst_wavparse_srcpad_event),
	  (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
	  (gst_wavparse_change_state), (plugin_init):
	  * gst/wavparse/gstwavparse.h:
	  Add push (streaming) mode to wavparse (fixes #337625)

2006-04-28 21:43:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/id3v2mux.c:
	  element renamed
	  Original commit message from CVS:
	  element renamed

2006-04-28 19:22:46 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  add plugin docs for ximagesrc
	  Original commit message from CVS:
	  add plugin docs for ximagesrc

2006-04-28 19:15:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  add ximagesrc icles test
	  Original commit message from CVS:
	  * configure.ac:
	  * tests/Makefile.am:
	  add ximagesrc icles test

2006-04-28 18:57:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  Move ximagesrc plug-in to good after review.  Fixes #336756.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_class_init),
	  (gst_cmml_enc_push_clip):
	  * sys/Makefile.am:
	  * sys/ximage/Makefile.am:
	  * sys/ximage/gstximagesrc.c:
	  Move ximagesrc plug-in to good after review.  Fixes #336756.

2006-04-28 16:51:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	  borgify naming
	  Original commit message from CVS:
	  borgify naming

2006-04-28 16:46:52 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* sys/ximage/gstximagesrc.c:
	  doc tweaks
	  Original commit message from CVS:
	  doc tweaks

2006-04-28 16:15:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* sys/ximage/Makefile.am:
	* sys/ximage/gstximagesrc.c:
	  clean up Makefile.am
	  Original commit message from CVS:
	  clean up Makefile.am

2006-04-28 15:33:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/taglib/gsttaglibmux.c:
	* ext/taglib/gsttaglibmux.h:
	  pedantic cleanups
	  Original commit message from CVS:
	  pedantic cleanups

2006-04-28 14:57:57 +0000  Michael Smith <msmith@xiph.org>

	  gst/icydemux/gsticydemux.*: Fix event handling: cache events when typefinding and forward later.
	  Original commit message from CVS:
	  * gst/icydemux/gsticydemux.c: (gst_icydemux_reset),         (gst_icydemux_init), (gst_icydemux_sink_setcaps),
	  (gst_icydemux_add_srcpad), (gst_icydemux_parse_and_send_tags),
	  (gst_icydemux_handle_event), (gst_icydemux_send_cached_events),
	  (gst_icydemux_typefind_or_forward), (gst_icydemux_add_meta),
	  (gst_icydemux_chain), (gst_icydemux_send_tag_event):
	  * gst/icydemux/gsticydemux.h:
	  Fix event handling: cache events when typefinding and forward later.

2006-04-28 14:55:20 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/osxaudio/gstosxaudiosink.c: Register osxaudiosrc to the plugin.
	  Original commit message from CVS:
	  2006-04-28  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/osxaudio/gstosxaudiosink.c:
	  (plugin_init):
	  Register osxaudiosrc to the plugin.
	  * sys/osxaudio/gstosxaudiosrc.c:
	  (gst_osx_audio_src_osxelement_do_init),
	  (gst_osx_audio_src_base_init), (gst_osx_audio_src_class_init),
	  (gst_osx_audio_src_init), (gst_osx_audio_src_set_property),
	  (gst_osx_audio_src_get_property),
	  (gst_osx_audio_src_create_ringbuffer), (gst_osx_audio_src_io_proc),
	  (gst_osx_audio_src_osxelement_init):
	  * sys/osxaudio/gstosxaudiosrc.h:
	  Port of osxaudiosrc to 0.10.
	  * sys/osxaudio/Makefile.am:
	  Add osxaudiosrc

2006-04-28 12:00:39 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* ChangeLog:
	  commit Changelog for previous commit
	  Original commit message from CVS:
	  commit Changelog for previous commit

2006-04-28 11:57:39 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* sys/osxaudio/gstosxringbuffer.c:
	* sys/osxaudio/gstosxringbuffer.h:
	  Forgot to commit, quick commit be4 apple dies
	  Original commit message from CVS:
	  Forgot to commit, quick commit be4 apple dies

2006-04-28 11:37:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: Recognise and skip any byte order marker (BOM) in
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (has_utf16_bom),
	  (parse_split_strings):
	  Recognise and skip any byte order marker (BOM) in
	  UTF-16 strings.

2006-04-27 16:05:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add docs for both avidemux and avimux.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-avi.xml:
	  * gst/avi/gstavidemux.c:
	  * gst/avi/gstavimux.c:
	  Add docs for both avidemux and avimux.

2006-04-27 14:51:06 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/avi/: Port AVI muxer to GStreamer-0.10 (#332031).
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet dot be>
	  * gst/avi/Makefile.am:
	  * gst/avi/gstavi.c: (plugin_init):
	  * gst/avi/gstavimux.c: (gst_avi_mux_get_type),
	  (gst_avi_mux_base_init), (gst_avi_mux_finalize),
	  (gst_avi_mux_class_init), (gst_avi_mux_init),
	  (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
	  (gst_avi_mux_pad_link), (gst_avi_mux_pad_unlink),
	  (gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
	  (gst_avi_mux_write_tag), (gst_avi_mux_riff_get_avi_header),
	  (gst_avi_mux_riff_get_avix_header),
	  (gst_avi_mux_riff_get_video_header),
	  (gst_avi_mux_riff_get_audio_header), (gst_avi_mux_add_index),
	  (gst_avi_mux_write_index), (gst_avi_mux_bigfile),
	  (gst_avi_mux_start_file), (gst_avi_mux_stop_file),
	  (gst_avi_mux_restart_file), (gst_avi_mux_handle_event),
	  (gst_avi_mux_fill_queue), (gst_avi_mux_send_pad_data),
	  (gst_avi_mux_strip_buffer), (gst_avi_mux_do_audio_buffer),
	  (gst_avi_mux_do_video_buffer), (gst_avi_mux_do_one_buffer),
	  (gst_avi_mux_loop), (gst_avi_mux_collect_pads),
	  (gst_avi_mux_get_property), (gst_avi_mux_set_property),
	  (gst_avi_mux_change_state):
	  * gst/avi/gstavimux.h:
	  Port AVI muxer to GStreamer-0.10 (#332031).
	  * tests/check/Makefile.am:
	  * tests/check/elements/avimux.c:
	  * tests/check/elements/.cvsignore:
	  Add unit test for AVI muxer.

2006-04-26 21:29:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.*: reverted patch #337625 for the price of 1 hour sleep
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	  (gst_wavparse_class_init), (gst_wavparse_reset),
	  (gst_wavparse_init), (gst_wavparse_create_sourcepad),
	  (gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
	  (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	  (gst_wavparse_send_event), (gst_wavparse_add_src_pad),
	  (gst_wavparse_stream_data), (gst_wavparse_loop),
	  (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate),
	  (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state),
	  (plugin_init):
	  * gst/wavparse/gstwavparse.h:
	  reverted patch #337625 for the price of 1 hour sleep

2006-04-26 20:11:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/wavparse/gstwavparse.*: correct partial implementation of push mode (from my last commit)
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	  (gst_wavparse_class_init), (gst_wavparse_reset),
	  (gst_wavparse_init), (gst_wavparse_create_sourcepad),
	  (gst_wavparse_parse_adtl), (gst_wavparse_parse_cues),
	  (gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
	  (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	  (gst_wavparse_stream_data), (gst_wavparse_loop),
	  (gst_wavparse_chain), (plugin_init):
	  * gst/wavparse/gstwavparse.h:
	  correct partial implementation of push mode
	  (from my last commit)

2006-04-26 17:37:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/esd/esdsink.c: Fix compile problem by defining ESD_MAX_WRITE_SIZE if it is not in esd.h
	  Original commit message from CVS:
	  * ext/esd/esdsink.c:
	  Fix compile problem by defining ESD_MAX_WRITE_SIZE if
	  it is not in esd.h

2006-04-26 17:08:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/auparse/gstauparse.*: Rewrite auparse to suck a little bit less: make source pad dynamic, so decodebin/playbin wo...
	  Original commit message from CVS:
	  * gst/auparse/gstauparse.c: (gst_au_parse_base_init),
	  (gst_au_parse_class_init), (gst_au_parse_init),
	  (gst_au_parse_reset), (gst_au_parse_add_srcpad),
	  (gst_au_parse_remove_srcpad), (gst_au_parse_parse_header),
	  (gst_au_parse_chain), (gst_au_parse_src_convert),
	  (gst_au_parse_src_query), (gst_au_parse_handle_seek),
	  (gst_au_parse_sink_event), (gst_au_parse_src_event),
	  (gst_au_parse_change_state):
	  * gst/auparse/gstauparse.h:
	  Rewrite auparse to suck a little bit less: make source pad
	  dynamic, so decodebin/playbin work with non-raw formats
	  like alaw/mulaw; add query function for duration/position
	  queries; check whether we have enough data before attempting
	  to parse the header (instead of crashing when that is not the
	  case); work around audioconvert sucking by swapping endianness
	  to the native endianness ourselves for float formats; send
	  initial newsegment event. Fixes #161712.

2006-04-26 16:29:38 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/osxaudio/: Port of osxaudiosink to 0.10
	  Original commit message from CVS:
	  2006-04-26  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/osxaudio/Makefile.am:
	  * sys/osxaudio/gstosxaudioelement.c:
	  (gst_osx_audio_element_get_type),
	  (gst_osx_audio_element_class_init):
	  * sys/osxaudio/gstosxaudioelement.h:
	  * sys/osxaudio/gstosxaudiosink.c:
	  (gst_osx_audio_sink_osxelement_do_init),
	  (gst_osx_audio_sink_base_init), (gst_osx_audio_sink_class_init),
	  (gst_osx_audio_sink_init), (gst_osx_audio_sink_set_property),
	  (gst_osx_audio_sink_get_property), (gst_osx_audio_sink_getcaps),
	  (gst_osx_audio_sink_create_ringbuffer),
	  (gst_osx_audio_sink_io_proc), (gst_osx_audio_sink_osxelement_init),
	  (plugin_init):
	  * sys/osxaudio/gstosxaudiosink.h:
	  Port of osxaudiosink to 0.10

2006-04-26 08:55:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/esd/esdsink.c: Always write ESD_BUF_SIZE bytes and use ESD_MAX_WRITE_SIZE as the size of the ringbuffer. This sho...
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_prepare), (gst_esdsink_delay):
	  Always write ESD_BUF_SIZE bytes and use ESD_MAX_WRITE_SIZE as
	  the size of the ringbuffer. This should fix hangs with older
	  esd sound servers.

2006-04-25 21:56:38 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Define GstElementDetails as const and also static (when defined as global)
	  Original commit message from CVS:
	  * ext/amrwb/gstamrwbdec.c:
	  * ext/amrwb/gstamrwbenc.c:
	  * ext/amrwb/gstamrwbparse.c:
	  * ext/arts/gst_arts.c:
	  * ext/artsd/gstartsdsink.c:
	  * ext/audiofile/gstafparse.c:
	  * ext/audiofile/gstafsink.c:
	  * ext/audiofile/gstafsrc.c:
	  * ext/audioresample/gstaudioresample.c:
	  * ext/bz2/gstbz2dec.c:
	  * ext/bz2/gstbz2enc.c:
	  * ext/cdaudio/gstcdaudio.c:
	  * ext/directfb/dfbvideosink.c:
	  * ext/divx/gstdivxdec.c:
	  * ext/divx/gstdivxenc.c:
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
	  * ext/faac/gstfaac.c: (gst_faac_base_init):
	  * ext/faad/gstfaad.c:
	  * ext/gsm/gstgsmdec.c:
	  * ext/gsm/gstgsmenc.c:
	  * ext/hermes/gsthermescolorspace.c:
	  * ext/ivorbis/vorbisfile.c:
	  * ext/lcs/gstcolorspace.c:
	  * ext/libfame/gstlibfame.c:
	  * ext/libmms/gstmms.c: (gst_mms_base_init):
	  * ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
	  * ext/nas/nassink.c: (gst_nassink_base_init):
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/sdl/sdlaudiosink.c:
	  * ext/sdl/sdlvideosink.c:
	  * ext/shout/gstshout.c:
	  * ext/snapshot/gstsnapshot.c:
	  * ext/sndfile/gstsf.c:
	  * ext/swfdec/gstswfdec.c:
	  * ext/tarkin/gsttarkindec.c:
	  * ext/tarkin/gsttarkinenc.c:
	  * ext/theora/theoradec.c:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
	  * ext/xvid/gstxviddec.c:
	  * ext/xvid/gstxvidenc.c:
	  * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init):
	  * gst/chart/gstchart.c:
	  * gst/colorspace/gstcolorspace.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init):
	  * gst/festival/gstfestival.c:
	  * gst/filter/gstbpwsinc.c:
	  * gst/filter/gstiir.c:
	  * gst/filter/gstlpwsinc.c:
	  * gst/freeze/gstfreeze.c:
	  * gst/games/gstpuzzle.c: (gst_puzzle_base_init):
	  * gst/librfb/gstrfbsrc.c:
	  * gst/mixmatrix/mixmatrix.c:
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  * gst/mpeg2sub/gstmpeg2subt.c:
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  * gst/multifilesink/gstmultifilesink.c:
	  * gst/overlay/gstoverlay.c:
	  * gst/passthrough/gstpassthrough.c:
	  * gst/playondemand/gstplayondemand.c:
	  * gst/qtdemux/qtdemux.c:
	  * gst/rtjpeg/gstrtjpegdec.c:
	  * gst/rtjpeg/gstrtjpegenc.c:
	  * gst/smooth/gstsmooth.c:
	  * gst/smoothwave/gstsmoothwave.c:
	  * gst/spectrum/gstspectrum.c:
	  * gst/speed/gstspeed.c:
	  * gst/stereo/gststereo.c:
	  * gst/switch/gstswitch.c:
	  * gst/tta/gstttadec.c: (gst_tta_dec_base_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_base_init):
	  * gst/vbidec/gstvbidec.c:
	  * gst/videocrop/gstvideocrop.c:
	  * gst/videodrop/gstvideodrop.c:
	  * gst/virtualdub/gstxsharpen.c:
	  * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init):
	  * gst/y4m/gsty4mencode.c:
	  * sys/cdrom/gstcdplayer.c:
	  * sys/directdraw/gstdirectdrawsink.c:
	  * sys/directsound/gstdirectsoundsink.c:
	  * sys/glsink/glimagesink.c:
	  * sys/qcam/gstqcamsrc.c:
	  * sys/v4l2/gstv4l2src.c:
	  * sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init):
	  * sys/ximagesrc/ximagesrc.c:
	  Define GstElementDetails as const and also static (when defined as
	  global)

2006-04-25 21:56:38 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Define GstElementDetails as const and also static (when defined as global)
	  Original commit message from CVS:
	  * ext/amrwb/gstamrwbdec.c:
	  * ext/amrwb/gstamrwbenc.c:
	  * ext/amrwb/gstamrwbparse.c:
	  * ext/arts/gst_arts.c:
	  * ext/artsd/gstartsdsink.c:
	  * ext/audiofile/gstafparse.c:
	  * ext/audiofile/gstafsink.c:
	  * ext/audiofile/gstafsrc.c:
	  * ext/audioresample/gstaudioresample.c:
	  * ext/bz2/gstbz2dec.c:
	  * ext/bz2/gstbz2enc.c:
	  * ext/cdaudio/gstcdaudio.c:
	  * ext/directfb/dfbvideosink.c:
	  * ext/divx/gstdivxdec.c:
	  * ext/divx/gstdivxenc.c:
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
	  * ext/faac/gstfaac.c: (gst_faac_base_init):
	  * ext/faad/gstfaad.c:
	  * ext/gsm/gstgsmdec.c:
	  * ext/gsm/gstgsmenc.c:
	  * ext/hermes/gsthermescolorspace.c:
	  * ext/ivorbis/vorbisfile.c:
	  * ext/lcs/gstcolorspace.c:
	  * ext/libfame/gstlibfame.c:
	  * ext/libmms/gstmms.c: (gst_mms_base_init):
	  * ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
	  * ext/nas/nassink.c: (gst_nassink_base_init):
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/sdl/sdlaudiosink.c:
	  * ext/sdl/sdlvideosink.c:
	  * ext/shout/gstshout.c:
	  * ext/snapshot/gstsnapshot.c:
	  * ext/sndfile/gstsf.c:
	  * ext/swfdec/gstswfdec.c:
	  * ext/tarkin/gsttarkindec.c:
	  * ext/tarkin/gsttarkinenc.c:
	  * ext/theora/theoradec.c:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
	  * ext/xvid/gstxviddec.c:
	  * ext/xvid/gstxvidenc.c:
	  * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init):
	  * gst/chart/gstchart.c:
	  * gst/colorspace/gstcolorspace.c:
	  * gst/deinterlace/gstdeinterlace.c:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init):
	  * gst/festival/gstfestival.c:
	  * gst/filter/gstbpwsinc.c:
	  * gst/filter/gstiir.c:
	  * gst/filter/gstlpwsinc.c:
	  * gst/freeze/gstfreeze.c:
	  * gst/games/gstpuzzle.c: (gst_puzzle_base_init):
	  * gst/librfb/gstrfbsrc.c:
	  * gst/mixmatrix/mixmatrix.c:
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  * gst/mpeg2sub/gstmpeg2subt.c:
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  * gst/multifilesink/gstmultifilesink.c:
	  * gst/overlay/gstoverlay.c:
	  * gst/passthrough/gstpassthrough.c:
	  * gst/playondemand/gstplayondemand.c:
	  * gst/qtdemux/qtdemux.c:
	  * gst/rtjpeg/gstrtjpegdec.c:
	  * gst/rtjpeg/gstrtjpegenc.c:
	  * gst/smooth/gstsmooth.c:
	  * gst/smoothwave/gstsmoothwave.c:
	  * gst/spectrum/gstspectrum.c:
	  * gst/speed/gstspeed.c:
	  * gst/stereo/gststereo.c:
	  * gst/switch/gstswitch.c:
	  * gst/tta/gstttadec.c: (gst_tta_dec_base_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_base_init):
	  * gst/vbidec/gstvbidec.c:
	  * gst/videocrop/gstvideocrop.c:
	  * gst/videodrop/gstvideodrop.c:
	  * gst/virtualdub/gstxsharpen.c:
	  * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init):
	  * gst/y4m/gsty4mencode.c:
	  * sys/cdrom/gstcdplayer.c:
	  * sys/directdraw/gstdirectdrawsink.c:
	  * sys/directsound/gstdirectsoundsink.c:
	  * sys/glsink/glimagesink.c:
	  * sys/qcam/gstqcamsrc.c:
	  * sys/v4l2/gstv4l2src.c:
	  * sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init):
	  * sys/ximagesrc/ximagesrc.c:
	  Define GstElementDetails as const and also static (when defined as
	  global)

2006-04-25 21:39:46 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Define GstElementDetails as const and also static (when defined as global)
	  Original commit message from CVS:
	  * ext/aalib/gstaasink.c:
	  * ext/annodex/gstcmmldec.c:
	  * ext/annodex/gstcmmlenc.c:
	  * ext/cairo/gsttextoverlay.c:
	  * ext/cairo/gsttimeoverlay.c:
	  * ext/cdio/gstcdiocddasrc.c:
	  * ext/dv/gstdvdec.c:
	  * ext/dv/gstdvdemux.c:
	  * ext/esd/esdmon.c:
	  * ext/esd/esdsink.c:
	  * ext/flac/gstflacenc.c:
	  * ext/flac/gstflactag.c:
	  * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
	  * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
	  * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
	  * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
	  * ext/gdk_pixbuf/pixbufscale.c:
	  * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
	  * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
	  * ext/jpeg/gstjpegdec.c:
	  * ext/jpeg/gstjpegenc.c:
	  * ext/jpeg/gstsmokedec.c:
	  * ext/jpeg/gstsmokeenc.c:
	  * ext/libcaca/gstcacasink.c:
	  * ext/libmng/gstmngdec.c:
	  * ext/libmng/gstmngenc.c:
	  * ext/libpng/gstpngdec.c:
	  * ext/libpng/gstpngenc.c:
	  * ext/mikmod/gstmikmod.c:
	  * ext/raw1394/gstdv1394src.c:
	  * ext/shout2/gstshout2.c: (gst_shout2send_init):
	  * ext/shout2/gstshout2.h:
	  * ext/speex/gstspeexdec.c:
	  * ext/speex/gstspeexenc.c:
	  * gst/alpha/gstalpha.c:
	  * gst/alpha/gstalphacolor.c:
	  * gst/apetag/gstapedemux.c:
	  * gst/auparse/gstauparse.c:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_base_init):
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_base_init):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
	  * gst/avi/gstavimux.c: (gst_avimux_base_init):
	  * gst/cutter/gstcutter.c:
	  * gst/debug/breakmydata.c:
	  * gst/debug/efence.c:
	  * gst/debug/gstnavigationtest.c:
	  * gst/debug/gstnavseek.c:
	  * gst/debug/negotiation.c:
	  * gst/debug/progressreport.c:
	  * gst/debug/testplugin.c:
	  * gst/effectv/gstaging.c:
	  * gst/effectv/gstdice.c:
	  * gst/effectv/gstedge.c:
	  * gst/effectv/gstquark.c:
	  * gst/effectv/gstrev.c:
	  * gst/effectv/gstshagadelic.c:
	  * gst/effectv/gstvertigo.c:
	  * gst/effectv/gstwarp.c:
	  * gst/flx/gstflxdec.c:
	  * gst/goom/gstgoom.c:
	  * gst/icydemux/gsticydemux.c:
	  * gst/id3demux/gstid3demux.c:
	  * gst/interleave/deinterleave.c:
	  * gst/interleave/interleave.c:
	  * gst/law/alaw-decode.c: (gst_alawdec_base_init):
	  * gst/law/alaw-encode.c: (gst_alawenc_base_init):
	  * gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
	  * gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
	  * gst/level/gstlevel.c:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
	  * gst/median/gstmedian.c:
	  * gst/monoscope/gstmonoscope.c:
	  * gst/multipart/multipartdemux.c:
	  * gst/multipart/multipartmux.c:
	  * gst/oldcore/gstaggregator.c:
	  * gst/oldcore/gstfdsink.c:
	  * gst/oldcore/gstmd5sink.c:
	  * gst/oldcore/gstmultifilesrc.c:
	  * gst/oldcore/gstpipefilter.c:
	  * gst/oldcore/gstshaper.c:
	  * gst/oldcore/gststatistics.c:
	  * gst/rtp/gstasteriskh263.c:
	  * gst/rtp/gstrtpL16depay.c:
	  * gst/rtp/gstrtpL16pay.c:
	  * gst/rtp/gstrtpamrdepay.c:
	  * gst/rtp/gstrtpamrpay.c:
	  * gst/rtp/gstrtpdepay.c:
	  * gst/rtp/gstrtpgsmpay.c:
	  * gst/rtp/gstrtph263pay.c:
	  * gst/rtp/gstrtph263pdepay.c:
	  * gst/rtp/gstrtph263ppay.c:
	  * gst/rtp/gstrtpilbcdepay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmp4vdepay.c:
	  * gst/rtp/gstrtpmp4vpay.c:
	  * gst/rtp/gstrtpmpadepay.c:
	  * gst/rtp/gstrtpmpapay.c:
	  * gst/rtp/gstrtppcmadepay.c:
	  * gst/rtp/gstrtppcmapay.c:
	  * gst/rtp/gstrtppcmudepay.c:
	  * gst/rtp/gstrtppcmupay.c:
	  * gst/rtp/gstrtpspeexdepay.c:
	  * gst/rtp/gstrtpspeexpay.c:
	  * gst/rtsp/gstrtpdec.c:
	  * gst/rtsp/gstrtspsrc.c:
	  * gst/smpte/gstsmpte.c:
	  * gst/udp/gstdynudpsink.c:
	  * gst/udp/gstmultiudpsink.c:
	  * gst/udp/gstudpsink.c:
	  * gst/udp/gstudpsrc.c:
	  * gst/videobox/gstvideobox.c:
	  * gst/videofilter/gstgamma.c: (gst_gamma_base_init):
	  * gst/videofilter/gstvideobalance.c:
	  * gst/videofilter/gstvideoflip.c:
	  * gst/videofilter/gstvideotemplate.c:
	  (gst_videotemplate_base_init):
	  * gst/videomixer/videomixer.c:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	  (gst_wavparse_class_init), (gst_wavparse_dispose),
	  (gst_wavparse_reset), (gst_wavparse_init),
	  (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
	  (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
	  (gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
	  (gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
	  (gst_wavparse_chain), (gst_wavparse_srcpad_event),
	  (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
	  (gst_wavparse_change_state):
	  * gst/wavparse/gstwavparse.h:
	  * sys/oss/gstossmixerelement.c:
	  * sys/oss/gstosssink.c:
	  * sys/oss/gstosssrc.c:
	  * sys/osxaudio/gstosxaudioelement.c:
	  * sys/osxaudio/gstosxaudiosink.c:
	  * sys/osxaudio/gstosxaudiosrc.c:
	  * sys/sunaudio/gstsunaudiomixer.c:
	  * sys/sunaudio/gstsunaudiosink.c:
	  Define GstElementDetails as const and also static (when defined as
	  global)

2006-04-25 17:57:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/gstjpegdec.c: Source pad has fixed caps. If we don't set this, bad things happen when the window is resized.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
	  Source pad has fixed caps. If we don't set this, bad
	  things happen when the window is resized.

2006-04-25 16:38:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Handle case where the TrackType ebml chunk does not come before the
	  Original commit message from CVS:
	  * gst/matroska/Makefile.am:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_handle_src_event):
	  * gst/matroska/matroska-ids.c:
	  (gst_matroska_track_init_video_context),
	  (gst_matroska_track_init_audio_context),
	  (gst_matroska_track_init_subtitle_context),
	  (gst_matroska_track_init_complex_context):
	  * gst/matroska/matroska-ids.h:
	  Handle case where the TrackType ebml chunk does not come before the
	  TrackInfoAudio or TrackInfoVideo ebml chunk (#339446). Ignore QoS
	  events.

2006-04-25 16:09:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: It's codec_data, not codec_info.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_setcaps):
	  It's codec_data, not codec_info.

2006-04-25 11:45:00 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/matroska/matroska-demux.c: Handle codec_data for VfW compatibility codec IDs (#339451)
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet dot be>
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
	  Handle codec_data for VfW compatibility codec IDs (#339451)
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_video_pad_setcaps):
	  Same here, handle codec_data and add additional caps we can handle
	  now to the pad template (huffyuv, dv and h263 video) (#339451)

2006-04-25 11:09:24 +0000  Josef Zlomek <josef.zlomek@itonis.tv>

	  gst/matroska/matroska-mux.c: Fix timestamping of B-frames, use signed integers, do some rounding (#339678).
	  Original commit message from CVS:
	  Patch by: Josef Zlomek  <josef dot zlomek at itonis dot tv>
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_create_buffer_header),
	  (gst_matroska_mux_write_data):
	  Fix timestamping of B-frames, use signed integers, do
	  some rounding (#339678).

2006-04-24 18:30:55 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* ext/annodex/gstcmmlparser.c:
	  just make it compile with --disable-gst-debug.
	  Original commit message from CVS:
	  just make it compile with --disable-gst-debug.

2006-04-23 15:55:30 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/matroska/matroska-demux.c: Fix a bad conversion using gst_guint64_to_gdouble. fabs ((gdouble) demux->index[entry]...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
	  Fix a bad conversion using gst_guint64_to_gdouble.
	  fabs ((gdouble) demux->index[entry].time - (gdouble) seek_pos) can not be
	  replaced by fabs (gst_guint64_to_gdouble (demux->index[entry].time - seek_pos)) as the
	  difference could be negative. fabs (gst_guint64_to_gdouble (demux->index[entry].time) -
	  gst_guint64_to_gdouble (seek_pos)) is the good solution. Thanks to Tim who has seen my
	  mistake.

2006-04-22 15:32:48 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/matroska/matroska-demux.c: Use gst_guint64_to_gdouble for conversions
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
	  Use gst_guint64_to_gdouble for conversions
	  * win32/vs6/gst_plugins_good.dsw:
	  * win32/vs6/libgsticydemux.dsp:
	  Add a project file for icydemux

2006-04-21 18:07:10 +0000  Fabrizio Gennari <fabrizio.ge@tiscali.it>

	  gst/avi/gstavidemux.c: When splitting audio chunks, the block alignment is not taken in consideration, so the smaller...
	  Original commit message from CVS:
	  Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_parse_index), (gst_avi_demux_massage_index):
	  When splitting audio chunks, the block alignment is not taken in
	  consideration, so the smaller chunks could be of size which is
	  not a multiple of the block alignment. Fixes #336904

2006-04-21 17:59:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/raw1394/gstdv1394src.c: Use scale functions
	  Original commit message from CVS:
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_convert):
	  Use scale functions

2006-04-21 17:27:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/dv/gstdv.c: Fix build.
	  Original commit message from CVS:
	  * ext/dv/gstdv.c: (plugin_init):
	  Fix build.

2006-04-21 17:15:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/debug/progressreport.c: Add 'format' property to force querying to a particular format.
	  Original commit message from CVS:
	  * gst/debug/progressreport.c: (gst_progress_report_finalize),
	  (gst_progress_report_class_init), (gst_progress_report_init),
	  (gst_progress_report_do_query), (gst_progress_report_report),
	  (gst_progress_report_set_property),
	  (gst_progress_report_get_property):
	  Add 'format' property to force querying to a particular format.

2006-04-21 15:50:28 +0000  Andy Wingo <wingo@pobox.com>

	  ext/dv/gstdv.c (plugin_init): libdv is a marginal decoder, at best, on big endian systems. Drop its rank in that case...
	  Original commit message from CVS:
	  2006-04-21  Andy Wingo  <wingo@pobox.com>
	  * ext/dv/gstdv.c (plugin_init): libdv is a marginal decoder, at
	  best, on big endian systems. Drop its rank in that case. OTOH on
	  x86 it's quite fine. See changes from today in gst-ffmpeg as well.

2006-04-21 12:40:41 +0000  Ed Catmur <ed@catmur.co.uk>

	  ext/lame/gstlame.c: Don't crash if we get an EOS event before the encoder has been set up (#339287).
	  Original commit message from CVS:
	  Patch by: Ed Catmur  <ed at catmur dot co dot uk>
	  * ext/lame/gstlame.c: (gst_lame_sink_event):
	  Don't crash if we get an EOS event before the encoder
	  has been set up (#339287).

2006-04-21 09:27:11 +0000  Michael Smith <msmith@xiph.org>

	  Add icydemux, and tests.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/icydemux/Makefile.am:
	  * gst/icydemux/gsticydemux.c: (gst_icydemux_get_type),
	  (gst_icydemux_base_init), (gst_icydemux_class_init),
	  (gst_icydemux_reset), (gst_icydemux_init),
	  (gst_icydemux_sink_setcaps), (gst_icydemux_dispose),
	  (gst_icydemux_add_srcpad), (gst_icydemux_remove_srcpad),
	  (unicodify), (gst_icydemux_unicodify),
	  (gst_icydemux_parse_and_send_tags),
	  (gst_icydemux_typefind_or_forward), (gst_icydemux_add_meta),
	  (gst_icydemux_chain), (gst_icydemux_change_state),
	  (gst_icydemux_send_tag_event), (plugin_init):
	  * gst/icydemux/gsticydemux.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/icydemux.c: (typefind_succeed),
	  (plugin_init), (icydemux_found_pad), (create_icydemux),
	  (cleanup_icydemux), (push_data), (GST_START_TEST),
	  (icydemux_suite), (main):
	  Add icydemux, and tests.

2006-04-20 17:48:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.c: Post SEGMENT_DONE message in TIME format.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_loop):
	  Post SEGMENT_DONE message in TIME format.

2006-04-20 17:29:56 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	  Added a couple of ifdefs to make it compile with other kernels.
	  Original commit message from CVS:
	  Added a couple of ifdefs to make it compile with other kernels.

2006-04-20 16:33:55 +0000  Fabrizio Gennari <fabrizio.ge@tiscali.it>

	  gst/avi/gstavidemux.c: Fix index creation when we have to scan the file to create an index. There may be other types ...
	  Original commit message from CVS:
	  Patch by: Fabrizio Gennari  <fabrizio dot ge at tiscali dot it>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_peek_tag),
	  (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan):
	  Fix index creation when we have to scan the file to create
	  an index. There may be other types of RIFF 'LIST' chunks than
	  'movi' and we need to skip them properly as well or we'll end up
	  reading garbage (#336889). Some other cosmetic changes.

2006-04-20 14:21:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.c: Add support for segment seeks (fixes #338290). Also demote some recurring debug message from D...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_loop),
	  (gst_flac_dec_handle_seek_event):
	  Add support for segment seeks (fixes #338290). Also demote
	  some recurring debug message from DEBUG to LOG level.

2006-04-20 13:23:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Set DISCONT flag on first buffer after a discontinuity.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	  (gst_matroskademux_do_index_seek),
	  (gst_matroska_demux_handle_seek_event),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock):
	  * gst/matroska/matroska-ids.h:
	  Set DISCONT flag on first buffer after a discontinuity.
	  Fix newsegment events sent when seeking and honour KEY_UNIT
	  seek flag. Create pad with bogus caps if we don't recognise
	  the stream codec id.
	  * gst/matroska/matroska-demux.h:
	  Fix GObject macros.

2006-04-20 11:00:16 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  gst/matroska/matroska-demux.c: Handle end of segment properly when set; don't dead-lock when posting start of segment...
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts  <manauw at skynet dot be>
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop):
	  Handle end of segment properly when set; don't dead-lock when
	  posting start of segment message when doing a segment seek.
	  Fixes #338810.

2006-04-20 09:48:05 +0000  j^ <j@bootlab.org>

	  gst/qtdemux/qtdemux.c: Never treat video streams as an audio stream.
	  Original commit message from CVS:
	  Patch by: j^ <j at bootlab dot org>
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
	  (qtdemux_video_caps):
	  Never treat video streams as an audio stream.
	  Add qtdrw mime type.
	  Fixes #339041

2006-04-20 09:11:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: Make mpeg2 aac audio work: create artificial private codec data chunk which faad2 seem...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps),
	  (gst_matroska_demux_plugin_init):
	  Make mpeg2 aac audio work: create artificial private codec data
	  chunk which faad2 seems to require, just as we do for mpeg4 aac.
	  Also call gst_riff_init(). Partially fixes #338767.

2006-04-19 15:16:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavenc/gstwavenc.*: Set caps on first outgoing buffer, so that it doesn't error out immediately with a non-negoti...
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_base_init),
	  (gst_wavenc_class_init), (gst_wavenc_init),
	  (gst_wavenc_create_header_buf), (gst_wavenc_push_header),
	  (gst_wavenc_sink_setcaps), (get_id_from_name), (gst_wavenc_event),
	  (gst_wavenc_chain), (gst_wavenc_change_state):
	  * gst/wavenc/gstwavenc.h:
	  Set caps on first outgoing buffer, so that it doesn't error out
	  immediately with a non-negotiated error (#338716). Rewrite and
	  clean up a bit; fix setcaps function to parse things properly;
	  fix sink caps (8bit audio is unsigned and doesn't have depth);
	  use boilerplate macros; remove unused properties stuff.

2006-04-19 09:27:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: For VBR audio, don't try to calculate the samples_per_frame.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  For VBR audio, don't try to calculate the samples_per_frame.
	  Fixes #338935.

2006-04-18 18:14:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gdk_pixbuf/gstgdkpixbuf.c: Leave JPEG decoding to our jpegdec plugin. gdkpixbufdec cannot handle MJPEG streams an...
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c:
	  Leave JPEG decoding to our jpegdec plugin. gdkpixbufdec cannot
	  handle MJPEG streams and might be autoplugged for those if the
	  user doesn't have jpegdec installed (resulting in a cryptic error
	  message about huffman tables). Better to disable JPEG decoding here
	  and let the user figure out that she needs to install jpegdec.

2006-04-18 18:04:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gdk_pixbuf/gstgdkpixbuf.*: Make work with packetised/framed input (e.g. png-in-quicktime). Use
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
	  (gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_init),
	  (gst_gdk_pixbuf_flush), (gst_gdk_pixbuf_chain):
	  * ext/gdk_pixbuf/gstgdkpixbuf.h:
	  Make work with packetised/framed input (e.g. png-in-quicktime). Use
	  GST_ELEMENT_ERROR when we return GST_FLOW_ERROR. Add some
	  GST_DEBUG_FUNCPTR here and there. Use GST_LOG for recurring
	  debug messages. Fix boilerplate macros.

2006-04-18 17:29:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gdk_pixbuf/gstgdkpixbuf.c: No need to special-case for Gdk-2.0 any longer, we require
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_get_capslist),
	  (gst_gdk_pixbuf_set_property), (gst_gdk_pixbuf_get_property):
	  No need to special-case for Gdk-2.0 any longer, we require
	  Gdk 2.2 or newer; minor clean-ups.

2006-04-18 17:17:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Rewrite a bit: use GstBaseSink::start and stop instead of a state change function; use GST_ELEMENT_ERROR for error re...
	  Original commit message from CVS:
	  * ext/shout2/gstshout2.c: (gst_shout2send_base_init),
	  (gst_shout2send_class_init), (gst_shout2send_init),
	  (set_shout_metadata), (gst_shout2send_set_metadata),
	  (gst_shout2send_event), (gst_shout2send_start),
	  (gst_shout2send_connect), (gst_shout2send_stop),
	  (gst_shout2send_render), (gst_shout2send_set_property),
	  (gst_shout2send_get_property), (gst_shout2send_setcaps),
	  (plugin_init):
	  * ext/shout2/gstshout2.h:
	  * po/POTFILES.in:
	  Rewrite a bit: use GstBaseSink::start and stop instead of a state
	  change function; use GST_ELEMENT_ERROR for error reporting, not
	  g_error() or GST_ERROR(); don't unref caps in setcaps function,
	  will cause crashes or assertion failures; remove (unused) "sync"
	  property, basesink already has such a property; misc. other
	  minor fixes and cleanups.

2006-04-18 14:15:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add translatable error message for when we cannot connect to the sound server, as "Cannot open resource for writing" ...
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_open), (gst_esdsink_prepare):
	  * ext/esd/gstesd.c: (plugin_init):
	  * po/POTFILES.in:
	  Add translatable error message for when we cannot
	  connect to the sound server, as "Cannot open resource
	  for writing" isn't really an acceptable message to show
	  to the user in this case.

2006-04-18 13:32:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/gst-i18n-plugin.h: Remove bogus file that doesn't belong here.
	  Original commit message from CVS:
	  * sys/oss/gst-i18n-plugin.h:
	  Remove bogus file that doesn't belong here.

2006-04-17 19:57:10 +0000  Philippe Valembois <lephilousophe@users.sf.net>

	  ext/shout2/gstshout2.*: Handle tags being received before the connection to the server is established properly (see #...
	  Original commit message from CVS:
	  Patch by: Philippe Valembois
	  * ext/shout2/gstshout2.c: (gst_shout2send_init),
	  (gst_shout2send_set_metadata), (gst_shout2send_event),
	  (gst_shout2send_render), (gst_shout2send_change_state):
	  * ext/shout2/gstshout2.h:
	  Handle tags being received before the connection to
	  the server is established properly (see #338636).

2006-04-17 19:43:32 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	  Just added a gtk-doc comment.
	  Original commit message from CVS:
	  Just added a gtk-doc comment.

2006-04-17 19:12:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/shout2/gstshout2.c: Don't crash in case the connection to the server fails: don't set pointer to NULL by assignin...
	  Original commit message from CVS:
	  * ext/shout2/gstshout2.c: (gst_shout2send_render):
	  Don't crash in case the connection to the server fails:
	  don't set pointer to NULL by assigning FALSE; error out
	  properly by using GST_ELEMENT_ERROR and returning
	  GST_FLOW_ERROR (fixes #338636). Lastly, free connection
	  before resetting the pointer.

2006-04-17 10:01:51 +0000  Alex Lancaster <alexlan@fedoraproject.org>

	  gst/id3demux/id3tags.c: (Fixes #338713)
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c:
	  Recognise TCO (Genre) tags in ID3v2.2. Patch by Alex Lancaster
	  (Fixes #338713)

2006-04-13 21:45:57 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  Fixed some memory leaks.
	  Original commit message from CVS:
	  Fixed some memory leaks.

2006-04-13 09:15:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* gst/rtp/Makefile.am:
	* gst/rtp/gstasteriskh263.h:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpdepay.h:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtpmp4gpay.h:
	* gst/rtp/gstrtpmp4vdepay.h:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadepay.h:
	* gst/rtp/gstrtpmpapay.h:
	* gst/rtp/gstrtppcmadepay.h:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmudepay.h:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtppcmupay.h:
	* gst/rtp/gstrtpspeexdepay.h:
	* gst/rtp/gstrtpspeexpay.h:
	  reverting rtp patches to fix freeze break on -base as explained on the list
	  Original commit message from CVS:
	  reverting rtp patches to fix freeze break on -base as explained on the list

2006-04-13 09:01:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtp/: Fix GObject macros.
	  Original commit message from CVS:
	  * gst/rtp/gstasteriskh263.h:
	  * gst/rtp/gstrtpL16depay.h:
	  * gst/rtp/gstrtpL16pay.h:
	  * gst/rtp/gstrtpamrdepay.h:
	  * gst/rtp/gstrtpamrpay.h:
	  * gst/rtp/gstrtpdepay.h:
	  * gst/rtp/gstrtpgsmdepay.h:
	  * gst/rtp/gstrtpgsmpay.h:
	  * gst/rtp/gstrtph263pay.h:
	  * gst/rtp/gstrtph263pdepay.h:
	  * gst/rtp/gstrtph263ppay.h:
	  * gst/rtp/gstrtpilbcdepay.h:
	  * gst/rtp/gstrtpilbcpay.h:
	  * gst/rtp/gstrtpmp4gpay.h:
	  * gst/rtp/gstrtpmp4vdepay.h:
	  * gst/rtp/gstrtpmp4vpay.h:
	  * gst/rtp/gstrtpmpadepay.h:
	  * gst/rtp/gstrtpmpapay.h:
	  * gst/rtp/gstrtppcmadepay.h:
	  * gst/rtp/gstrtppcmapay.h:
	  * gst/rtp/gstrtppcmudepay.h:
	  * gst/rtp/gstrtppcmupay.h:
	  * gst/rtp/gstrtpspeexdepay.h:
	  * gst/rtp/gstrtpspeexpay.h:
	  Fix GObject macros.

2006-04-13 03:42:51 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst/rtp/: Ported mulaw and alaw payloaders to use new base class
	  Original commit message from CVS:
	  2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
	  * gst/rtp/gstrtppcmapay.c:
	  * gst/rtp/gstrtppcmapay.h:
	  * gst/rtp/gstrtppcmupay.c:
	  * gst/rtp/gstrtppcmupay.h:
	  Ported mulaw and alaw payloaders to use new base class
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c:
	  * gst/rtp/gstrtpilbcpay.c:
	  * gst/rtp/gstrtpilbcpay.h:
	  * gst/rtp/gstrtpilbcdepay.c:
	  * gst/rtp/gstrtpilbcdepay.h:
	  Added new iLBC payloader/depayloader. Payloader uses new audio payload base
	  class.

2006-04-12 21:57:02 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c:
	  Fix to work in read mode.
	  Original commit message from CVS:
	  Fix to work in read mode.

2006-04-12 09:42:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/gdk_pixbuf/gstgdkpixbuf.c: Some cleanups.
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
	  (gst_gdk_pixbuf_get_capslist), (gst_gdk_pixbuf_sink_getcaps),
	  (gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_init),
	  (gst_gdk_pixbuf_flush), (gst_gdk_pixbuf_sink_event),
	  (gst_gdk_pixbuf_chain):
	  Some cleanups.
	  Added RGBA as a possible output format.
	  Correctly free the supported mimetypes.
	  deprecate silent arg, it's not used.
	  Return result from _alloc_buffer to peer.

2006-04-11 18:03:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/rtp/gstrtpmp4vdepay.c: Don't leak memory allocated by gst_buffer_new_and_alloc() by overwriting GST_BUFFER_MALLOC...
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_process):
	  Don't leak memory allocated by gst_buffer_new_and_alloc() by
	  overwriting GST_BUFFER_MALLOCDATA.

2006-04-11 15:27:31 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  fix version number macro
	  Original commit message from CVS:
	  fix version number macro

2006-04-11 09:35:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/libpng/gstpngdec.*: Handle more than one frame if the content is framed, like with png-in-quicktime (#331917).
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_init),
	  (user_endrow_callback), (user_end_callback),
	  (gst_pngdec_caps_create_and_set), (gst_pngdec_chain),
	  (gst_pngdec_sink_setcaps), (gst_pngdec_sink_event),
	  (gst_pngdec_libpng_clear), (gst_pngdec_change_state):
	  * ext/libpng/gstpngdec.h:
	  Handle more than one frame if the content is framed,
	  like with png-in-quicktime (#331917).

2006-04-10 19:55:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  sys/oss/: - the user-visible error strings were in the wrong category
	  Original commit message from CVS:
	  * sys/oss/Makefile.am:
	  * sys/oss/common.h:
	  * sys/oss/gstosssink.c: (gst_oss_sink_init), (gst_oss_sink_open),
	  (gst_oss_sink_prepare), (gst_oss_sink_unprepare):
	  * sys/oss/gstosssrc.c: (gst_oss_src_prepare),
	  (gst_oss_src_unprepare):
	  - the user-visible error strings were in the wrong category
	  - and the messages were not marked for translation
	  - which is actually a good thing, because they were exactly
	  the kind of message you would never want anyone to see
	  - the macros were using variables that didn't exist in the macro
	  arguments
	  - and they were obviously copied from each other and then modified
	  - so a common header makes sense

2006-04-10 17:16:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Fix parsing of newer stsd chunks again.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  Fix parsing of newer stsd chunks again.

2006-04-10 16:09:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/ebml-read.c: Don't try to modify read-only data.
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_read_sint):
	  Don't try to modify read-only data.
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock):
	  Fix comment (won't crash any longer now).

2006-04-10 15:48:55 +0000  Michael Smith <msmith@xiph.org>

	  ext/annodex/gstcmmlenc.c: Use copies of header buffers for caps to avoid circular refcounting problems (as in theorad...
	  Original commit message from CVS:
	  * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_set_header_on_caps):
	  Use copies of header buffers for caps to avoid circular refcounting
	  problems (as in theoradec, vorbisdec).
	  * tests/check/elements/cmmldec.c: (GST_START_TEST):
	  Fix a typo in test that meant it was testing the wrong thing.
	  * tests/check/elements/cmmlenc.c: (check_headers):
	  Fix refcount checks now that we use buffer-copies for caps.

2006-04-10 15:43:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: Use static pad templates with ANY caps for audio and video source pads and get rid of ...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init),
	  (gst_matroska_demux_handle_seek_event),
	  (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
	  (gst_matroska_demux_subtitle_caps),
	  (gst_matroska_demux_plugin_init):
	  Use static pad templates with ANY caps for audio and video
	  source pads and get rid of a lot of unnecessary (and partially
	  broken) code for the template caps. Clean up caps finding
	  functions. Fixes playback of audio files/streams that do not
	  contain the sample rate and/or number of channels in the audio
	  context (happens a lot with vorbis/mp3 .mka files it seems).
	  Fixes #337183.
	  Also add myself to copyright holders.

2006-04-10 15:29:21 +0000  Michael Smith <msmith@xiph.org>

	  ext/annodex/gstcmmlutils.c: Use g_list_delete_link () instead of g_list_remove_link () so that we free the link as we...
	  Original commit message from CVS:
	  * ext/annodex/gstcmmlutils.c: (gst_cmml_track_list_del_clip):
	  Use g_list_delete_link () instead of g_list_remove_link () so that
	  we free the link as well as the contained data.

2006-04-10 14:20:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Fix framerate calculation.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
	  (qtdemux_parse_trak):
	  Fix framerate calculation.

2006-04-10 10:10:55 +0000  Ryan Lortie (desrt) <desrt@destr.ca>

	  gst/avi/gstavidemux.c: Fix some crashers with empty chunks. (Fixes #337749)
	  Original commit message from CVS:
	  Patch by: Ryan Lortie (desrt) <desrt at destr dot ca>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
	  (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
	  (gst_avi_demux_stream_header):
	  Fix some crashers with empty chunks. (Fixes #337749)

2006-04-10 08:31:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: force mono 8000 Hz on AMR samples.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  force mono 8000 Hz on AMR samples.

2006-04-09 18:30:51 +0000  Sébastien Moutte <sebastien@moutte.net>

	  ext/neon/gstneonhttpsrc.c: remove atoll by using g_ascii_strtoull (atoll is not supported on WIN32)
	  Original commit message from CVS:
	  * ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_start):
	  remove atoll by using g_ascii_strtoull (atoll is not supported on WIN32)
	  * sys/directdraw/gstdirectdrawsink.c:
	  * sys/directsound/gstdirectsoundsink.c:
	  done some cleans in sources
	  * win32/vs6:
	  add project files for neon, qtdemux

2006-04-09 17:31:37 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/level/gstlevel.c: use G_GINT64_CONSTANT for INT64 constants
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_set_caps),(gst_level_transform_ip):
	  use G_GINT64_CONSTANT for INT64 constants
	  * gst/videofilter/gstvideobalance.c:
	  define rint for WIN32 #define rint(x) (floor((x)+0.5))
	  * win32/vs6/libgstavi.dsp:
	  add missing libraries for the link and remove avimux.c from
	  the project as it isn't ported to 0.10 yet

2006-04-09 14:00:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/ebml-read.c: Even better would be if we actually did the right thing here (also, G_GUINT64_CONSTANT only...
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_read_sint):
	  Even better would be if we actually did the right thing
	  here (also, G_GUINT64_CONSTANT only exists since GLib-2.10).

2006-04-09 13:52:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/ebml-read.c: Can't just replace 1LL with 1L here just because MSVC doesn't support it, as it might lead ...
	  Original commit message from CVS:
	  * gst/matroska/ebml-read.c: (gst_ebml_read_sint):
	  Can't just replace 1LL with 1L here just because MSVC doesn't
	  support it, as it might lead to incorrect results when doing the
	  bitshifting here. Using GLib's G_GUINT64_CONSTANT() macro to
	  force a 64-bit constant in a way that all compilers are happy with.

2006-04-08 21:48:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
	  Original commit message from CVS:
	  * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init):
	  * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init):
	  * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init):
	  * ext/arts/gst_arts.c: (gst_arts_class_init):
	  * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init):
	  * ext/audiofile/gstafsink.c: (gst_afsink_class_init):
	  * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init):
	  * ext/audioresample/gstaudioresample.c:
	  * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init):
	  * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init):
	  * ext/divx/gstdivxdec.c: (gst_divxdec_class_init):
	  * ext/hermes/gsthermescolorspace.c:
	  (gst_hermes_colorspace_class_init):
	  * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init):
	  * ext/jack/gstjack.c: (gst_jack_class_init):
	  * ext/jack/gstjackbin.c: (gst_jack_bin_class_init):
	  * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init):
	  * ext/libfame/gstlibfame.c: (gst_fameenc_class_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init):
	  * ext/nas/nassink.c: (gst_nassink_class_init):
	  * ext/shout/gstshout.c: (gst_icecastsend_class_init):
	  * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
	  * ext/sndfile/gstsf.c: (gst_sf_class_init):
	  * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init),
	  (gst_swfdec_class_init):
	  * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init):
	  * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init):
	  * gst/chart/gstchart.c: (gst_chart_class_init):
	  * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init):
	  * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init):
	  * gst/festival/gstfestival.c: (gst_festival_class_init):
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
	  * gst/filter/gstiir.c: (gst_iir_class_init):
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
	  * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init):
	  * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init):
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  (gst_system_encode_class_init):
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  (gst_mp1videoparse_class_init):
	  * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init):
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  (gst_mp3parse_class_init):
	  * gst/overlay/gstoverlay.c: (gst_overlay_class_init):
	  * gst/passthrough/gstpassthrough.c: (passthrough_class_init):
	  * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init):
	  * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init):
	  * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init):
	  * gst/smooth/gstsmooth.c: (gst_smooth_class_init):
	  * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init):
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
	  * gst/stereo/gststereo.c: (gst_stereo_class_init):
	  * gst/switch/gstswitch.c: (gst_switch_class_init):
	  * gst/tta/gstttadec.c: (gst_tta_dec_class_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_class_init):
	  * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init):
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init):
	  * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init):
	  * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init):
	  * sys/cdrom/gstcdplayer.c: (cdplayer_class_init):
	  * sys/directsound/gstdirectsoundsink.c:
	  (gst_directsoundsink_class_init):
	  * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init):
	  * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init):
	  * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init):
	  * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init):
	  * sys/v4l2/gstv4l2colorbalance.c:
	  (gst_v4l2_color_balance_channel_class_init):
	  * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init),
	  (gst_v4l2_tuner_norm_class_init):
	  * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init):
	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)

2006-04-08 21:48:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
	  Original commit message from CVS:
	  * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init):
	  * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init):
	  * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init):
	  * ext/arts/gst_arts.c: (gst_arts_class_init):
	  * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init):
	  * ext/audiofile/gstafsink.c: (gst_afsink_class_init):
	  * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init):
	  * ext/audioresample/gstaudioresample.c:
	  * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init):
	  * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init):
	  * ext/divx/gstdivxdec.c: (gst_divxdec_class_init):
	  * ext/hermes/gsthermescolorspace.c:
	  (gst_hermes_colorspace_class_init):
	  * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init):
	  * ext/jack/gstjack.c: (gst_jack_class_init):
	  * ext/jack/gstjackbin.c: (gst_jack_bin_class_init):
	  * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init):
	  * ext/libfame/gstlibfame.c: (gst_fameenc_class_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init):
	  * ext/nas/nassink.c: (gst_nassink_class_init):
	  * ext/shout/gstshout.c: (gst_icecastsend_class_init):
	  * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
	  * ext/sndfile/gstsf.c: (gst_sf_class_init):
	  * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init),
	  (gst_swfdec_class_init):
	  * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init):
	  * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init):
	  * gst/chart/gstchart.c: (gst_chart_class_init):
	  * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init):
	  * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init):
	  * gst/festival/gstfestival.c: (gst_festival_class_init):
	  * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
	  * gst/filter/gstiir.c: (gst_iir_class_init):
	  * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
	  * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init):
	  * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init):
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  (gst_system_encode_class_init):
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  (gst_mp1videoparse_class_init):
	  * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init):
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  (gst_mp3parse_class_init):
	  * gst/overlay/gstoverlay.c: (gst_overlay_class_init):
	  * gst/passthrough/gstpassthrough.c: (passthrough_class_init):
	  * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init):
	  * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init):
	  * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init):
	  * gst/smooth/gstsmooth.c: (gst_smooth_class_init):
	  * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init):
	  * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
	  * gst/stereo/gststereo.c: (gst_stereo_class_init):
	  * gst/switch/gstswitch.c: (gst_switch_class_init):
	  * gst/tta/gstttadec.c: (gst_tta_dec_class_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_class_init):
	  * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init):
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init):
	  * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init):
	  * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init):
	  * sys/cdrom/gstcdplayer.c: (cdplayer_class_init):
	  * sys/directsound/gstdirectsoundsink.c:
	  (gst_directsoundsink_class_init):
	  * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init):
	  * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init):
	  * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init):
	  * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init):
	  * sys/v4l2/gstv4l2colorbalance.c:
	  (gst_v4l2_color_balance_channel_class_init):
	  * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init),
	  (gst_v4l2_tuner_norm_class_init):
	  * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init):
	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)

2006-04-08 21:21:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
	  Original commit message from CVS:
	  * ext/aalib/gstaasink.c: (gst_aasink_class_init):
	  * ext/esd/esdsink.c: (gst_esdsink_class_init):
	  * ext/flac/gstflactag.c: (gst_flac_tag_class_init):
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init):
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init):
	  * ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init):
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init):
	  * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
	  * ext/libmng/gstmngdec.c: (gst_mngdec_class_init):
	  * ext/libmng/gstmngenc.c: (gst_mngenc_class_init):
	  * ext/libpng/gstpngdec.c: (gst_pngdec_class_init):
	  * ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
	  * ext/mikmod/gstmikmod.c: (gst_mikmod_class_init):
	  * ext/shout2/gstshout2.c: (gst_shout2send_class_init):
	  * ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
	  * gst/alpha/gstalpha.c: (gst_alpha_class_init):
	  * gst/avi/gstavimux.c: (gst_avimux_class_init):
	  * gst/debug/efence.c: (gst_efence_class_init):
	  * gst/debug/negotiation.c: (gst_negotiation_class_init):
	  * gst/flx/gstflxdec.c: (gst_flxdec_class_init):
	  * gst/goom/gstgoom.c: (gst_goom_class_init):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_class_init):
	  * gst/interleave/deinterleave.c: (deinterleave_class_init):
	  * gst/interleave/interleave.c: (interleave_class_init):
	  * gst/law/alaw-decode.c: (gst_alawdec_class_init):
	  * gst/law/alaw-encode.c: (gst_alawenc_class_init):
	  * gst/law/mulaw-encode.c: (gst_mulawenc_class_init):
	  * gst/median/gstmedian.c: (gst_median_class_init):
	  * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init):
	  * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init):
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init):
	  * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init):
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init):
	  * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init):
	  * gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init):
	  * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init):
	  * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init):
	  * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init):
	  * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init):
	  * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init):
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init):
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init):
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init):
	  * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init):
	  * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init):
	  * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init):
	  * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init):
	  * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init):
	  * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init):
	  * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init):
	  * gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init):
	  * gst/smpte/gstsmpte.c: (gst_smpte_class_init):
	  * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init):
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
	  * gst/udp/gstudpsink.c: (gst_udpsink_class_init):
	  * gst/videomixer/videomixer.c: (gst_videomixer_class_init):
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
	  * sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init):
	  * sys/oss/gstosssink.c: (gst_oss_sink_class_init):
	  * sys/osxaudio/gstosxaudioelement.c:
	  (gst_osxaudioelement_class_init):
	  * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init):
	  * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init):
	  * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init):
	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)

2006-04-08 19:06:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix more broken GObject macros
	  Original commit message from CVS:
	  * ext/mikmod/gstmikmod.h:
	  * gst/level/gstlevel.h:
	  Fix more broken GObject macros

2006-04-08 18:41:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix broken GObject macros
	  Original commit message from CVS:
	  * ext/xine/gstxine.h:
	  * gst-libs/gst/play/play.h:
	  * sys/v4l2/gstv4l2element.h:
	  * sys/ximagesrc/ximageutil.h:
	  Fix broken GObject macros

2006-04-08 18:25:55 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix broken GObject macros
	  Original commit message from CVS:
	  * ext/annodex/gstcmmldec.h:
	  * ext/annodex/gstcmmlenc.h:
	  * ext/annodex/gstcmmltag.h:
	  * ext/cairo/gsttextoverlay.h:
	  * ext/ladspa/gstsignalprocessor.h:
	  * gst/matroska/ebml-read.h:
	  * gst/matroska/ebml-write.h:
	  * sys/osxaudio/gstosxaudioelement.h:
	  Fix broken GObject macros

2006-04-08 18:23:04 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Don't make rounding errors in timestamp/duration calculations.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
	  (gst_qtdemux_chain), (gst_qtdemux_add_stream), (qtdemux_dump_stsz),
	  (qtdemux_dump_stco), (qtdemux_parse_trak):
	  Don't make rounding errors in timestamp/duration calculations.
	  Fix timestamps for AMR and IMA4.  Fixes (#337436).
	  Create a dummy segment even when there is no edit list.

2006-04-08 13:09:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.c: Don't try to seek beyond the end of the file (would occasionally display error dialogs in tote...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_handle_seek_event):
	  Don't try to seek beyond the end of the file (would
	  occasionally display error dialogs in totem when seeking
	  to the end) (#335869). Will still throw an error though
	  if the file is truncated and the total_samples value in
	  the stream header is wrong.

2006-04-07 18:15:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.*: If the stream header doesn't contain the total number of samples, search for the last flac fra...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_calculate_crc8),
	  (gst_flac_dec_scan_got_frame), (gst_flac_dec_scan_for_last_block),
	  (gst_flac_dec_metadata_callback):
	  * ext/flac/gstflacdec.h:
	  If the stream header doesn't contain the total number of samples,
	  search for the last flac frame at the end of the file and calculate
	  the total duration from that frame's offset (fixes #337609).

2006-04-07 15:53:43 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  Typo fix, s/XFree86/X11 and added doc blurb saying that it fixates to 25fps
	  Original commit message from CVS:
	  2006-04-07  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/amrwb/amrwb-code/Makefile.am:
	  * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_recalc),
	  (gst_ximagesrc_create), (gst_ximagesrc_set_property):
	  Typo fix, s/XFree86/X11 and added doc blurb saying that it fixates to
	  25fps

2006-04-07 15:47:27 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  tests/icles/ximagesrc-test.c: Actually assert that pipeline goes to playing
	  Original commit message from CVS:
	  2006-04-07  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * tests/icles/ximagesrc-test.c: (main):
	  Actually assert that pipeline goes to playing

2006-04-07 15:27:40 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximagesrc/ximagesrc.c: Fix typo, C++ style comments and other small cleanups
	  Original commit message from CVS:
	  2006-04-07  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_recalc),
	  (composite_pixel), (gst_ximagesrc_ximage_get),
	  (gst_ximagesrc_create), (gst_ximagesrc_set_property):
	  Fix typo, C++ style comments and other small cleanups

2006-04-07 10:48:19 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: Don't unref the GstPadTemplate returned by gst_element_class_get_pad_template().
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream):
	  Don't unref the GstPadTemplate returned by
	  gst_element_class_get_pad_template().

2006-04-06 19:16:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Added full edit list support.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
	  (gst_qtdemux_handle_src_query), (gst_qtdemux_find_index),
	  (gst_qtdemux_find_keyframe), (gst_qtdemux_find_segment),
	  (gst_qtdemux_move_stream), (gst_qtdemux_perform_seek),
	  (gst_qtdemux_do_seek), (gst_qtdemux_change_state),
	  (gst_qtdemux_activate_segment),
	  (gst_qtdemux_prepare_current_sample), (gst_qtdemux_advance_sample),
	  (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop),
	  (qtdemux_parse_trak):
	  Added full edit list support.
	  Avoid overflows in prologue image detection code.
	  Avoid roundoff errors in timestamp calculations.

2006-04-06 11:35:26 +0000  j^ <j@bootlab.org>

	  Unify the long descriptions in the plugin details (#337263).
	  Original commit message from CVS:
	  Patch by: j^  <j at bootlab dot org>
	  * ext/amrwb/gstamrwbdec.c:
	  * ext/amrwb/gstamrwbenc.c:
	  * ext/amrwb/gstamrwbparse.c:
	  * ext/arts/gst_arts.c:
	  * ext/artsd/gstartsdsink.c:
	  * ext/audiofile/gstafparse.c:
	  * ext/audiofile/gstafsink.c:
	  * ext/audiofile/gstafsrc.c:
	  * ext/cdaudio/gstcdaudio.c:
	  * ext/directfb/dfbvideosink.c:
	  * ext/divx/gstdivxdec.c:
	  * ext/divx/gstdivxenc.c:
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
	  * ext/faac/gstfaac.c: (gst_faac_base_init):
	  * ext/faad/gstfaad.c:
	  * ext/gsm/gstgsmdec.c:
	  * ext/gsm/gstgsmenc.c:
	  * ext/hermes/gsthermescolorspace.c:
	  * ext/ivorbis/vorbisfile.c:
	  * ext/lcs/gstcolorspace.c:
	  * ext/libfame/gstlibfame.c:
	  * ext/libmms/gstmms.c: (gst_mms_base_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
	  * ext/nas/nassink.c: (gst_nassink_base_init):
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/polyp/polypsink.c: (gst_polypsink_base_init):
	  * ext/sdl/sdlaudiosink.c:
	  * ext/sdl/sdlvideosink.c:
	  * ext/shout/gstshout.c:
	  * ext/snapshot/gstsnapshot.c:
	  * ext/sndfile/gstsf.c:
	  * ext/tarkin/gsttarkindec.c:
	  * ext/tarkin/gsttarkinenc.c:
	  * ext/theora/theoradec.c:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
	  * ext/xvid/gstxviddec.c:
	  * ext/xvid/gstxvidenc.c:
	  * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init):
	  * gst/chart/gstchart.c:
	  * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init):
	  * gst/festival/gstfestival.c:
	  * gst/filter/gstiir.c:
	  * gst/filter/gstlpwsinc.c:
	  * gst/freeze/gstfreeze.c:
	  * gst/games/gstpuzzle.c: (gst_puzzle_base_init):
	  * gst/mixmatrix/mixmatrix.c:
	  * gst/mpeg1sys/gstmpeg1systemencode.c:
	  * gst/mpeg1videoparse/gstmp1videoparse.c:
	  * gst/mpeg2sub/gstmpeg2subt.c:
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  * gst/multifilesink/gstmultifilesink.c:
	  * gst/overlay/gstoverlay.c:
	  * gst/passthrough/gstpassthrough.c:
	  * gst/playondemand/gstplayondemand.c:
	  * gst/qtdemux/qtdemux.c:
	  * gst/rtjpeg/gstrtjpegdec.c:
	  * gst/rtjpeg/gstrtjpegenc.c:
	  * gst/smooth/gstsmooth.c:
	  * gst/tta/gstttadec.c: (gst_tta_dec_base_init):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_base_init):
	  * gst/videocrop/gstvideocrop.c:
	  * gst/videodrop/gstvideodrop.c:
	  * gst/virtualdub/gstxsharpen.c:
	  * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init):
	  * gst/y4m/gsty4mencode.c:
	  Unify the long descriptions in the plugin details (#337263).

2006-04-06 09:14:30 +0000  Brian Cameron <brian.cameron@sun.com>

	  sys/sunaudio/gstsunaudiosink.*: Use spec->segsize and spec->segtotal in the prepare function to initialise the ring b...
	  Original commit message from CVS:
	  Patch by: Brian Cameron  <brian dot cameron at sun dot com>
	  * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_init),
	  (gst_sunaudiosink_prepare), (gst_sunaudiosink_write):
	  * sys/sunaudio/gstsunaudiosink.h:
	  Use spec->segsize and spec->segtotal in the prepare function
	  to initialise the ring buffer instead of using the buffer-time
	  property (#337421).

2006-04-06 08:52:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump core requirements to CVS for gst_pad_query_peer_duration() which is used by speexdec.
	  Original commit message from CVS:
	  * configure.ac:
	  Bump core requirements to CVS for gst_pad_query_peer_duration()
	  which is used by speexdec.

2006-04-05 18:27:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/speex/: Fix seeking and duration queries (#337033); clean up and refactor a bit.
	  Original commit message from CVS:
	  * ext/speex/gstspeex.c: (plugin_init):
	  * ext/speex/gstspeexdec.c: (gst_speex_dec_class_init),
	  (gst_speex_dec_reset), (gst_speex_dec_init), (speex_dec_convert),
	  (speex_get_sink_query_types), (speex_dec_sink_query),
	  (speex_get_src_query_types), (speex_dec_src_query),
	  (speex_dec_src_event), (speex_dec_sink_event),
	  (speex_dec_chain_parse_header), (speex_dec_chain_parse_comments),
	  (speex_dec_chain_parse_data), (speex_dec_chain),
	  (gst_speex_dec_get_property), (gst_speex_dec_set_property),
	  (speex_dec_change_state):
	  * ext/speex/gstspeexdec.h:
	  Fix seeking and duration queries (#337033); clean up and
	  refactor a bit.

2006-04-05 12:41:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/raw1394/gstdv1394src.c: distinguish between device not found and could not open for reading
	  Original commit message from CVS:
	  * ext/raw1394/gstdv1394src.c:
	  distinguish between device not found and could not open for
	  reading

2006-04-05 08:36:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: Use duration as segment stop position if none is explicitly configured.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek),
	  (gst_qtdemux_do_seek), (gst_qtdemux_loop_state_movie),
	  (gst_qtdemux_loop):
	  Use duration as segment stop position if none is
	  explicitly configured.
	  Also perform EOS when we run past the segment stop.

2006-04-04 11:20:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: More cleanups, added comments.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_go_back),
	  (gst_qtdemux_perform_seek), (gst_qtdemux_do_seek),
	  (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop),
	  (gst_qtdemux_chain), (qtdemux_parse_tree), (qtdemux_parse_trak):
	  More cleanups, added comments.
	  Mark discontinuities on outgoing buffers.
	  Post better errors when something goes wrong.
	  Handle EOS and segment end properly.

2006-04-04 08:31:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.*: Handle stss boxes so we can mark and find keyframes.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
	  (gst_qtdemux_push_event), (gst_qtdemux_go_back),
	  (gst_qtdemux_perform_seek), (gst_qtdemux_do_seek),
	  (gst_qtdemux_handle_src_event), (plugin_init),
	  (gst_qtdemux_change_state), (gst_qtdemux_loop_state_movie),
	  (gst_qtdemux_loop), (gst_qtdemux_chain),
	  (qtdemux_sink_activate_pull), (gst_qtdemux_add_stream),
	  (qtdemux_parse), (qtdemux_parse_tree), (qtdemux_parse_trak),
	  (qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num),
	  (qtdemux_tag_add_gnre), (gst_qtdemux_handle_esds):
	  * gst/qtdemux/qtdemux.h:
	  Handle stss boxes so we can mark and find keyframes.
	  Implement correct accurate and keyframe seeking.
	  Use _DEBUG_OBJECT when possible.

2006-04-03 13:29:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/.gitignore:
	  ignore more
	  Original commit message from CVS:
	  ignore more

2006-04-03 13:28:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* pkgconfig/Makefile.am:
	  fix dist
	  Original commit message from CVS:
	  fix dist

2006-04-03 09:02:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  add a .pc file so other modules can use good plugins in tests
	  Original commit message from CVS:
	  * Makefile.am:
	  * configure.ac:
	  * pkgconfig/.cvsignore:
	  * pkgconfig/Makefile.am:
	  * pkgconfig/gstreamer-plugins-good-uninstalled.pc.in:
	  add a .pc file so other modules can use good plugins in tests

2006-04-01 16:50:49 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* docs/plugins/inspect/plugin-qtdemux.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* ext/taglib/gsttaglibmux.c:
	* tests/check/elements/id3v2mux.c:
	  add taglib checks and docs
	  Original commit message from CVS:
	  add taglib checks and docs

2006-04-01 15:30:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/upload.mak:
	  disable use of AS_LIBTOOL_TAGS, it doesn't work correctly
	  Original commit message from CVS:
	  disable use of AS_LIBTOOL_TAGS, it doesn't work correctly

2006-04-01 14:03:03 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-ladspa.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  adding inspect files
	  Original commit message from CVS:
	  adding inspect files

2006-04-01 10:15:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/icles/ximagesrc-test.c:
	  5 second timeout
	  Original commit message from CVS:
	  5 second timeout

2006-04-01 10:14:26 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/icles/.gitignore:
	* tests/icles/Makefile.am:
	* tests/icles/ximagesrc-test.c:
	  rename test
	  Original commit message from CVS:
	  rename test

2006-04-01 10:09:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/audiofx/gststereo.c:
	  rework build; add translations for v4l2
	  Original commit message from CVS:
	  rework build; add translations for v4l2

2006-04-01 10:09:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/equalizer/gstiirequalizer.c:
	* gst/qtdemux/qtdemux.c:
	* gst/spectrum/gstspectrum.c:
	* gst/videocrop/gstvideocrop.c:
	* sys/directdraw/gstdirectdrawplugin.c:
	* sys/directsound/gstdirectsoundplugin.c:
	* sys/v4l2/gstv4l2.c:
	* sys/ximage/gstximagesrc.c:
	  rework build; add translations for v4l2
	  Original commit message from CVS:
	  rework build; add translations for v4l2

2006-04-01 09:56:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: clean up, use AS_VERSION and AS_NANO
	  Original commit message from CVS:
	  * configure.ac:
	  clean up, use AS_VERSION and AS_NANO
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
	  use PACKAGE_VERSION define
	  * po/af.po:
	  * po/az.po:
	  * po/cs.po:
	  * po/en_GB.po:
	  * po/hu.po:
	  * po/it.po:
	  * po/nb.po:
	  * po/nl.po:
	  * po/or.po:
	  * po/sq.po:
	  * po/sr.po:
	  * po/sv.po:
	  * po/uk.po:
	  * po/vi.po:
	  updated

2006-04-01 09:54:39 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: rework similarly to other modules
	  Original commit message from CVS:
	  * configure.ac:
	  rework similarly to other modules
	  * ext/a52dec/gsta52dec.c:
	  * ext/amrnb/amrnb.c:
	  * ext/dvdnav/dvdnavsrc.c:
	  * ext/dvdread/dvdreadsrc.c:
	  * ext/lame/gstlame.c:
	  * ext/mad/gstid3tag.c:
	  * ext/mpeg2dec/gstmpeg2dec.c:
	  * ext/sidplay/gstsiddec.cc:
	  * gst/asfdemux/gstasf.c:
	  * gst/dvdlpcmdec/gstdvdlpcmdec.c:
	  * gst/dvdsub/gstdvdsubdec.c:
	  * gst/iec958/ac3iec.c:
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  * gst/mpegstream/gstmpegstream.c:
	  * gst/realmedia/rmdemux.c: (plugin_init):
	  use the correct defines

2006-03-31 17:52:36 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  Add tests and fix PAR caps issue to ximagesrc
	  Original commit message from CVS:
	  2006-03-31  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * Makefile.am:
	  * configure.ac:
	  * sys/ximagesrc/ximagesrc.c:
	  (gst_ximagesrc_ximage_get),
	  (gst_ximagesrc_get_caps), (gst_ximagesrc_class_init):
	  * sys/ximagesrc/ximageutil.c:
	  * tests/Makefile.am:
	  * tests/icles/Makefile.am:
	  * tests/icles/ximagesrc-test.c: (terminate_playback), (main):
	  Add tests and fix PAR caps issue to ximagesrc

2006-03-31 16:32:47 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximagesrc/ximagesrc.c: Add docs to ximagesrc
	  Original commit message from CVS:
	  2006-03-31  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximagesrc/ximagesrc.c:
	  Add docs to ximagesrc

2006-03-31 15:21:35 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  sys/ximagesrc/: Fix ximagesrc so a) the cursor doesnt trail and b) there are no yellow rectangles with the cursor
	  Original commit message from CVS:
	  2006-03-31  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * sys/ximagesrc/ximagesrc.c: (composite_pixel),
	  (gst_ximagesrc_ximage_get), (gst_ximagesrc_set_property),
	  (gst_ximagesrc_get_caps), (gst_ximagesrc_class_init):
	  * sys/ximagesrc/ximagesrc.h:
	  * sys/ximagesrc/ximageutil.c: (ximageutil_xcontext_get):
	  * sys/ximagesrc/ximageutil.h:
	  Fix ximagesrc so a) the cursor doesnt trail and b) there are no
	  yellow rectangles with the cursor

2006-03-30 23:46:42 +0000  Sébastien Moutte <sebastien@moutte.net>

	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstalaw.dsp:
	* win32/vs6/libgstalpha.dsp:
	* win32/vs6/libgstalphacolor.dsp:
	* win32/vs6/libgstapetag.dsp:
	* win32/vs6/libgstauparse.dsp:
	* win32/vs6/libgstautodetect.dsp:
	* win32/vs6/libgstavi.dsp:
	* win32/vs6/libgstcutter.dsp:
	* win32/vs6/libgsteffectv.dsp:
	* win32/vs6/libgstflx.dsp:
	* win32/vs6/libgstgoom.dsp:
	* win32/vs6/libgstid3demux.dsp:
	* win32/vs6/libgstinterleave.dsp:
	* win32/vs6/libgstjpeg.dsp:
	* win32/vs6/libgstlevel.dsp:
	* win32/vs6/libgstmatroska.dsp:
	* win32/vs6/libgstmedian.dsp:
	* win32/vs6/libgstmonoscope.dsp:
	* win32/vs6/libgstmulaw.dsp:
	* win32/vs6/libgstmultipart.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstsmpte.dsp:
	* win32/vs6/libgstspeex.dsp:
	* win32/vs6/libgstvideobalance.dsp:
	* win32/vs6/libgstvideobox.dsp:
	* win32/vs6/libgstvideoflip.dsp:
	* win32/vs6/libgstvideomixer.dsp:
	* win32/vs6/libgstwavenc.dsp:
	* win32/vs6/libgstwavparse.dsp:
	  I'm too lazy to comment this
	  Original commit message from CVS:
	  *** empty log message ***

2006-03-30 23:37:16 +0000  Sébastien Moutte <sebastien@moutte.net>

	  ext\jpeg\smokecodec.c: use of GST_DEBUG instead of DEBUG(a...) for WIN32
	  Original commit message from CVS:
	  * ext\jpeg\smokecodec.c:
	  use of GST_DEBUG instead of DEBUG(a...) for WIN32
	  * ext\speex\gstspeexenc.c: (gst_speexenc_set_header_on_caps):
	  move first instruction after all variables declarations
	  * gst\alpha\gstalpha.c:
	  * gst\effectv\gstshagadelic.c:
	  * gst\smpte\paint.c:
	  * gst\videofilter\gstvideobalance.c:
	  define M_PI if it's not defined (it's not defined on WIN32)
	  * gst\cutter\gstcutter.c: (gst_cutter_chain):
	  * gst\id3demux\id3v2frames.c: (parse_relative_volume_adjustment_two):
	  * gst\level\gstlevel.c: (gst_level_set_property), (gst_level_transform_ip):
	  * gst\matroska\matroska-demux.c: (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_video_caps):
	  * gst\matroska\matroska-mux.c: (gst_matroska_mux_start), (gst_matroska_mux_finish):
	  * gst\wavparse\gstwavparse.c: (gst_wavparse_stream_data):
	  use gst_guint64_to_gdouble for conversions
	  * gst\goom\filters.c: (setPixelRGB_):
	  fix a debug which was using undefined variable
	  * gst\level\gstlevel.c: (gst_level_set_caps), (gst_level_transform_ip):
	  * gst\matroska\ebml-read.c: (gst_ebml_read_sint):
	  replace LL suffix with L suffix (LL isn't supported by MSVC6.0)
	  * win32/vs6:
	  add vs6 projects files for most of plugins-good

2006-03-30 15:37:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  better/unified long descriptions
	  Original commit message from CVS:
	  * ext/aalib/gstaasink.c:
	  * ext/annodex/gstcmmldec.c:
	  * ext/annodex/gstcmmlenc.c:
	  * ext/cairo/gsttextoverlay.c:
	  * ext/cairo/gsttimeoverlay.c:
	  * ext/cdio/gstcdiocddasrc.c:
	  * ext/dv/gstdvdec.c:
	  * ext/esd/esdmon.c:
	  * ext/esd/esdsink.c:
	  * ext/flac/gstflacdec.c:
	  * ext/flac/gstflacenc.c:
	  * ext/flac/gstflactag.c:
	  * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
	  * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
	  * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
	  * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
	  * ext/gdk_pixbuf/gstgdkpixbuf.c:
	  * ext/gdk_pixbuf/pixbufscale.c:
	  * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
	  * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
	  * ext/jpeg/gstjpegdec.c:
	  * ext/jpeg/gstjpegenc.c:
	  * ext/jpeg/gstsmokedec.c:
	  * ext/jpeg/gstsmokeenc.c:
	  * ext/libcaca/gstcacasink.c:
	  * ext/libmng/gstmngdec.c:
	  * ext/libmng/gstmngenc.c:
	  * ext/libpng/gstpngdec.c:
	  * ext/libpng/gstpngenc.c:
	  * ext/mikmod/gstmikmod.c:
	  * ext/raw1394/gstdv1394src.c:
	  * ext/shout2/gstshout2.c:
	  * ext/speex/gstspeexdec.c:
	  * ext/speex/gstspeexenc.c:
	  * gst/alpha/gstalpha.c:
	  * gst/alpha/gstalphacolor.c:
	  * gst/auparse/gstauparse.c:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_base_init):
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_base_init):
	  * gst/avi/gstavimux.c: (gst_avimux_base_init):
	  * gst/cutter/gstcutter.c:
	  * gst/debug/breakmydata.c:
	  * gst/debug/efence.c:
	  * gst/debug/gstnavigationtest.c:
	  * gst/debug/negotiation.c:
	  * gst/debug/progressreport.c:
	  * gst/debug/testplugin.c:
	  * gst/effectv/gstaging.c:
	  * gst/effectv/gstdice.c:
	  * gst/effectv/gstedge.c:
	  * gst/effectv/gstquark.c:
	  * gst/effectv/gstrev.c:
	  * gst/effectv/gstvertigo.c:
	  * gst/effectv/gstwarp.c:
	  * gst/flx/gstflxdec.c:
	  * gst/goom/gstgoom.c:
	  * gst/interleave/deinterleave.c:
	  * gst/interleave/interleave.c:
	  * gst/law/alaw-decode.c: (gst_alawdec_base_init):
	  * gst/law/alaw-encode.c: (gst_alawenc_base_init):
	  * gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
	  * gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
	  * gst/level/gstlevel.c:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
	  * gst/median/gstmedian.c:
	  * gst/monoscope/gstmonoscope.c:
	  * gst/multipart/multipartdemux.c:
	  * gst/multipart/multipartmux.c:
	  * gst/oldcore/gstmd5sink.c:
	  * gst/oldcore/gstmultifilesrc.c:
	  * gst/oldcore/gstpipefilter.c:
	  * gst/oldcore/gstshaper.c:
	  * gst/oldcore/gststatistics.c:
	  * gst/rtp/gstasteriskh263.c:
	  * gst/rtp/gstrtpL16depay.c:
	  * gst/rtp/gstrtpL16pay.c:
	  * gst/rtp/gstrtpamrdepay.c:
	  * gst/rtp/gstrtpamrpay.c:
	  * gst/rtp/gstrtpdepay.c:
	  * gst/rtp/gstrtpgsmpay.c:
	  * gst/rtp/gstrtph263pay.c:
	  * gst/rtp/gstrtph263pdepay.c:
	  * gst/rtp/gstrtph263ppay.c:
	  * gst/rtp/gstrtpmp4gpay.c:
	  * gst/rtp/gstrtpmp4vdepay.c:
	  * gst/rtp/gstrtpmp4vpay.c:
	  * gst/rtp/gstrtpmpadepay.c:
	  * gst/rtp/gstrtpmpapay.c:
	  * gst/rtp/gstrtppcmadepay.c:
	  * gst/rtp/gstrtppcmapay.c:
	  * gst/rtp/gstrtppcmudepay.c:
	  * gst/rtp/gstrtppcmupay.c:
	  * gst/rtp/gstrtpspeexdepay.c:
	  * gst/rtp/gstrtpspeexpay.c:
	  * gst/rtsp/gstrtpdec.c:
	  * gst/smpte/gstsmpte.c:
	  * gst/videobox/gstvideobox.c:
	  * gst/videofilter/gstgamma.c: (gst_gamma_base_init):
	  * gst/videofilter/gstvideobalance.c:
	  * gst/videofilter/gstvideoflip.c:
	  * gst/videofilter/gstvideotemplate.c:
	  (gst_videotemplate_base_init):
	  * gst/videomixer/videomixer.c:
	  * gst/wavenc/gstwavenc.c:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init):
	  better/unified long descriptions
	  Fixed #336602
	  Some cleanups to auparse, don't send multiple newsegments.

2006-03-29 16:06:50 +0000  Michael Dominic K <mdk@mdk.org.pl>

	  ext/dv/gstdvdemux.*: Seek in READY patch. Only works for pull based mode.
	  Original commit message from CVS:
	  From a patch by: Michael Dominic K. <mdk at mdk dot org dot pl>
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_class_init),
	  (gst_dvdemux_reset), (gst_dvdemux_src_convert),
	  (gst_dvdemux_send_event), (gst_dvdemux_flush), (gst_dvdemux_loop),
	  (gst_dvdemux_sink_activate_pull), (gst_dvdemux_change_state):
	  * ext/dv/gstdvdemux.h:
	  Seek in READY patch. Only works for pull based mode.
	  Fixes #323880

2006-03-28 16:06:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.*: Make xingheader property non-functional, it's broken anyway after all (use xingmux instead).
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_set_property),
	  (gst_lame_get_property), (gst_lame_setup):
	  * ext/lame/gstlame.h:
	  Make xingheader property non-functional, it's broken anyway
	  after all (use xingmux instead).

2006-03-28 15:10:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.c: On EOS, flush encoder and send remaining data. Fix return value handling in sink event function.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_event):
	  On EOS, flush encoder and send remaining data. Fix
	  return value handling in sink event function.

2006-03-27 17:06:45 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/v4l2src_calls.c:
	  Small fix, now pwc driver can tell about its buffers.
	  Original commit message from CVS:
	  Small fix, now pwc driver can tell about its buffers.

2006-03-27 14:09:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gdk_pixbuf/gstgdkpixbuf.c: Fix two crashers: don't unref the same caps twice, and set pixbuf loader to NULL after...
	  Original commit message from CVS:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_flush),
	  (gst_gdk_pixbuf_event):
	  Fix two crashers: don't unref the same caps twice, and
	  set pixbuf loader to NULL after freeing it.

2006-03-27 14:00:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/speex/gstspeexenc.*: Don't leak adapter.
	  Original commit message from CVS:
	  * ext/speex/gstspeexenc.c: (gst_speexenc_class_init),
	  (gst_speexenc_finalize), (gst_speexenc_sink_setcaps),
	  (gst_speexenc_chain):
	  * ext/speex/gstspeexenc.h:
	  Don't leak adapter.
	  A push *always* takes ownership of the buffer, even on
	  errors.
	  Small cleanups.

2006-03-26 19:56:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gsttaglib.*: Fix newsegment event handling a bit. We need to cache the first newsegment event, because we ...
	  Original commit message from CVS:
	  * ext/taglib/gsttaglib.cc:
	  * ext/taglib/gsttaglib.h:
	  Fix newsegment event handling a bit. We need to
	  cache the first newsegment event, because we can't
	  adjust offsets yet when we get it, as we don't
	  know the size of the tag yet for sure at that point.
	  Also do some minor cleaning up here and there and add
	  some debug statements.

2006-03-26 12:24:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/gstid3demux.c: Create source pad without leaking.
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
	  Create source pad without leaking.

2006-03-25 21:57:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gsttaglib.cc: We do not want to proxy the caps on the sink pad; our source pad should have application/x-i...
	  Original commit message from CVS:
	  * ext/taglib/gsttaglib.cc:
	  We do not want to proxy the caps on the sink pad; our
	  source pad should have application/x-id3 caps; also,
	  don't use already-freed strings in debug messages;
	  finally, adjust buffer offsets on buffers sent out.

2006-03-25 13:02:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/gstv4l2src.c: Older kernels don't seem to have this particular v4l2 format, so comment out until this gets f...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c:
	  Older kernels don't seem to have this particular v4l2 format,
	  so comment out until this gets fixed properly (and make
	  buildbots happy).

2006-03-25 05:31:28 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* common:
	* sys/v4l2/gstv4l2colorbalance.c:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2element.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  Just make few things more robust and also some identation.
	  Original commit message from CVS:
	  Just make few things more robust and also some identation.

2006-03-24 19:41:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/flac/: Spifify a bit.
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_handle_seek_event):
	  * ext/flac/gstflacdec.h:
	  * ext/flac/gstflacenc.h:
	  Spifify a bit.
	  Fix deadly lock order error in seeking code, STREAM_LOCK
	  cannot be taken within LOCK and the streaming variables are
	  protected with the STREAM_LOCK anyway.

2006-03-24 18:56:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: this patch combines the global init_frames with the stream init_frames. Rationale being that t...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_index),
	  (gst_avi_demux_stream_index), (gst_avi_demux_stream_scan),
	  (gst_avi_demux_massage_index), (gst_avi_demux_handle_seek):
	  this patch combines the global init_frames with the stream
	  init_frames. Rationale being that the global delay should
	  be subtracted from any stream delay.
	  Fixes #335858.

2006-03-24 17:11:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: use DEBUG_FUNCPTR for collectpads
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_init):
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_init):
	  * gst/smpte/gstsmpte.c: (gst_smpte_init):
	  * gst/videomixer/videomixer.c: (gst_videomixer_init):
	  use DEBUG_FUNCPTR for collectpads

2006-03-24 09:54:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/gstjpegenc.c: Don't crash when encoding images where the number of rows isn't a multiple of 2*DCTSIZE. Add s...
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_init), (gst_jpegenc_chain):
	  Don't crash when encoding images where the number of rows isn't
	  a multiple of 2*DCTSIZE. Add some GST_DEBUG_FUNCPTR.

2006-03-23 21:28:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  More state change function fixes.
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_change_state):
	  * gst/interleave/deinterleave.c: (deinterleave_change_state):
	  * gst/interleave/interleave.c: (interleave_change_state):
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_change_state):
	  More state change function fixes.

2006-03-23 20:12:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/esd/esdsink.*: Fix esd choppy playback by configuring audiosink correctly. Fixes #325191
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_class_init),
	  (gst_esdsink_getcaps), (gst_esdsink_open), (gst_esdsink_close),
	  (gst_esdsink_prepare), (gst_esdsink_unprepare),
	  (gst_esdsink_delay), (gst_esdsink_reset):
	  * ext/esd/esdsink.h:
	  Fix esd choppy playback by configuring audiosink
	  correctly. Fixes #325191

2006-03-23 19:57:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/libpng/gstpngdec.c: Make state change function thread-safe.
	  Original commit message from CVS:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_change_state):
	  Make state change function thread-safe.

2006-03-23 16:50:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: Don't try to read beyond the end of the file just because the header claims a bigger size...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_get_upstream_size),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	  Don't try to read beyond the end of the file just because
	  the header claims a bigger size (like with truncated files).

2006-03-23 15:36:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.*: Delay source pad creation until we have the first chunk of media data, so the we can exam...
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
	  (gst_wavparse_stream_data), (gst_wavparse_loop):
	  * gst/wavparse/gstwavparse.h:
	  Delay source pad creation until we have the first chunk of
	  media data, so the we can examine the data and adjust the
	  caps accordingly if required. This makes playback of .wav
	  files with DTS-declared-as-PCM content work (#313266).

2006-03-22 19:50:56 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add videobalance plugn
	  Original commit message from CVS:
	  add videobalance plugn

2006-03-22 13:02:11 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  mention fixed bug number in the changelog
	  Original commit message from CVS:
	  mention fixed bug number in the changelog

2006-03-22 13:00:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/: Don't attempt typefinding on too-short buffers that have been completely trimmed away.
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Don't attempt typefinding on too-short buffers that have been
	  completely trimmed away.
	  * gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
	  Improve the debug output

2006-03-21 18:12:59 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/esd/esdsink.c: Some cleanups.
	  Original commit message from CVS:
	  * ext/esd/esdsink.c: (gst_esdsink_class_init), (gst_esdsink_init),
	  (gst_esdsink_finalize), (gst_esdsink_getcaps), (gst_esdsink_open),
	  (gst_esdsink_close), (gst_esdsink_prepare), (gst_esdsink_write),
	  (gst_esdsink_set_property), (gst_esdsink_get_property):
	  Some cleanups.
	  Reset fd to -1 when we close them.

2006-03-21 16:19:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: the OPTIONS request result is optional so don't fail on it.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	  the OPTIONS request result is optional so don't
	  fail on it.

2006-03-21 14:53:36 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/: gcc 4.1 unreferenced pointer fixes.
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_reset):
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_reset):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_create_sourcepad),
	  (gst_wavparse_stream_headers), (gst_wavparse_send_event),
	  (gst_wavparse_change_state):
	  gcc 4.1 unreferenced pointer fixes.

2006-03-21 13:07:31 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/wavparse/gstwavparse.c: Fix block alignment calculation. Alignment should be done before adding the byte offset w...
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek):
	  Fix block alignment calculation. Alignment should be done before
	  adding the byte offset where the data starts (#335231).

2006-03-20 18:34:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/matroska/ebml-write.c: Ensure that we set correct caps on buffers that are transferred direct from the input.
	  Original commit message from CVS:
	  * gst/matroska/ebml-write.c: (gst_ebml_write_element_push):
	  Ensure that we set correct caps on buffers that are transferred
	  direct from the input.

2006-03-20 17:38:48 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/goom/: Free filter data when cleaning up. (Fixes: #334995)
	  Original commit message from CVS:
	  * gst/goom/filters.c: (zoomFilterDestroy):
	  * gst/goom/goom_core.c: (goom_close):
	  Free filter data when cleaning up. (Fixes: #334995)

2006-03-20 08:59:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gsttaglib.h: Fix left-over gst_my_filter_get_type.
	  Original commit message from CVS:
	  * ext/taglib/gsttaglib.h:
	  Fix left-over gst_my_filter_get_type.

2006-03-17 16:34:36 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* sys/ximage/gstximagesrc.c:
	  Have a show mouse pointer property and use it if we can
	  Original commit message from CVS:
	  Have a show mouse pointer property and use it if we can

2006-03-17 15:33:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Don't compile udp and rtsp plugins on win32 (mingw) or other systems that don't have <sys/socket.h> for...
	  Original commit message from CVS:
	  * configure.ac:
	  Don't compile udp and rtsp plugins on win32 (mingw) or other
	  systems that don't have <sys/socket.h> for some reason (#316203).

2006-03-16 17:28:07 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* ChangeLog:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gstdv1394src.h:
	  Change bus reset handler so it reports useful information such as whether the device being used connected or disconne...
	  Original commit message from CVS:
	  Change bus reset handler so it reports useful information such as
	  whether the device being used connected or disconnected

2006-03-16 16:06:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: We only care about gain and peak data for the master volume.
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c:
	  (parse_relative_volume_adjustment_two):
	  We only care about gain and peak data for the master volume.

2006-03-16 13:22:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/id3v2frames.c: Read replay gain tags (#323721).
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	  (parse_id_string), (parse_unique_file_identifier),
	  (parse_relative_volume_adjustment_two), (id3v2_tag_to_taglist):
	  Read replay gain tags (#323721).

2006-03-15 23:19:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump requirements to gst-plugins-base CVS because of buggy gst_tag_from_id3_user_tag() in 0.10.5.
	  Original commit message from CVS:
	  * configure.ac:
	  Bump requirements to gst-plugins-base CVS because
	  of buggy gst_tag_from_id3_user_tag() in 0.10.5.

2006-03-15 22:30:24 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	* ChangeLog:
	* gst/rtp/gstrtppcmadepay.c:
	  Fixed one of the caps in the code from mulaw to alaw.
	  Original commit message from CVS:
	  Fixed one of the caps in the code from mulaw to alaw.

2006-03-15 16:21:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/apetag/gsttagdemux.c: Ensure that we set caps on the buffers we pass.
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
	  Ensure that we set caps on the buffers we pass.
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
	  (gst_id3demux_sink_activate):
	  Ensure that we set caps on the buffers we pass.
	  Use STREAM, TYPE_NOT_FOUND as the error class when
	  typefinding fails.

2006-03-15 16:17:12 +0000  Edward Hervey <bilboed@bilboed.com>

	  Fix memleak with gst_static_pad_template_get().
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_init):
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_init), (gst_dvdemux_add_pads):
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init):
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
	  (gst_jpeg_dec_setcaps):
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_init):
	  * ext/jpeg/gstsmokedec.c: (gst_smokedec_init):
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init):
	  * ext/libmng/gstmngdec.c: (gst_mngdec_init),
	  (gst_mngdec_src_getcaps):
	  * ext/libpng/gstpngdec.c: (gst_pngdec_init),
	  (gst_pngdec_caps_create_and_set):
	  * ext/libpng/gstpngenc.c: (gst_pngenc_init):
	  * ext/mikmod/gstmikmod.c: (gst_mikmod_init):
	  * ext/speex/gstspeexdec.c: (gst_speex_dec_init):
	  * gst/alpha/gstalpha.c: (gst_alpha_init):
	  * gst/auparse/gstauparse.c: (gst_au_parse_init):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_init),
	  (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream):
	  * gst/cutter/gstcutter.c: (gst_cutter_init):
	  * gst/debug/efence.c: (gst_efence_init), (gst_efence_getrange),
	  (gst_efence_checkgetrange):
	  * gst/debug/negotiation.c: (gst_negotiation_init):
	  * gst/flx/gstflxdec.c: (gst_flxdec_init):
	  * gst/goom/gstgoom.c: (gst_goom_init):
	  * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_init):
	  * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_init):
	  * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_init):
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init):
	  * gst/rtp/gstrtpdepay.c: (gst_rtp_depay_init):
	  * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_init):
	  * gst/rtsp/gstrtpdec.c: (gst_rtpdec_init):
	  * gst/smpte/gstsmpte.c: (gst_smpte_init):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
	  (gst_wavparse_create_sourcepad):
	  Fix memleak with gst_static_pad_template_get().
	  This uses gst_pad_new_from_static_template() instead.
	  Fixes #333512

2006-03-15 15:08:20 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Let's not forget to chain up to the parent dispose.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_dispose):
	  Let's not forget to chain up to the parent dispose.

2006-03-15 14:39:25 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Series of memleak fixes:
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
	  (gst_qtdemux_init), (gst_qtdemux_dispose),
	  (gst_qtdemux_add_stream), (qtdemux_parse_trak):
	  Series of memleak fixes:
	  - Unref the GstAdapter in finalize.
	  - Use gst_pad_new_from_static_template(), shorter and safer.
	  - Free unused QtDemuxStream when not used.

2006-03-15 13:43:42 +0000  Christophe Fergeau <teuf@gnome.org>

	  ext/lame/gstlame.c: use GST_DEBUG_FUNCPTR more often.
	  Original commit message from CVS:
	  Patch by: Christophe Fergeau  <teuf gnome org>
	  * ext/lame/gstlame.c: (gst_lame_release_memory),
	  (gst_lame_finalize), (gst_lame_class_init),
	  (gst_lame_sink_setcaps), (gst_lame_init), (gst_lame_sink_event),
	  (gst_lame_change_state):
	  Fix some memory leaks (#333345), use GST_DEBUG_FUNCPTR more often.

2006-03-14 17:56:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump -base requirement to 0.10.5 for gst_tag_from_id3_user_tag(), used by id3demux.
	  Original commit message from CVS:
	  * configure.ac:
	  Bump -base requirement to 0.10.5 for gst_tag_from_id3_user_tag(),
	  used by id3demux.
	  * gst/id3demux/gstid3demux.c: (plugin_init):
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	  (parse_user_text_identification_frame),
	  (parse_unique_file_identifier):
	  Add support for UFID and TXXX frames and extract musicbrainz tags.

2006-03-14 17:24:03 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/v4l2/gstv4l2src.c: Initialization of the debugging category should be as early as possible, moving it from _class...
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_base_init),
	  (gst_v4l2src_class_init):
	  Initialization of the debugging category should be as early as possible,
	  moving it from _class_init() to beginning of _base_init().

2006-03-14 15:28:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Catch short reads, like they might happen with truncated files (see #305279); remove unnecessa...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
	  Catch short reads, like they might happen with truncated
	  files (see #305279); remove unnecessary indentation.

2006-03-14 14:18:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Fix DIB image inversion for pictures with a depth != 8 (#305279).
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_invert):
	  Fix DIB image inversion for pictures with a
	  depth != 8 (#305279).

2006-03-14 09:23:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/gstjpegdec.*: Fix durations on outgoing buffers after seeking in MJPEG files (#334083); some minor clean-ups.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_class_init),
	  (gst_jpeg_dec_chain), (gst_jpeg_dec_change_state):
	  * ext/jpeg/gstjpegdec.h:
	  Fix durations on outgoing buffers after seeking
	  in MJPEG files (#334083); some minor clean-ups.

2006-03-13 18:28:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Implement seek in READY (re-fixes #327658)
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
	  (gst_wavparse_change_state):
	  Implement seek in READY (re-fixes #327658)

2006-03-13 17:22:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/gsttaglib.cc: Add gtk-doc blurb (unused for the time being); match registered plugin name to the filename ...
	  Original commit message from CVS:
	  * ext/taglib/gsttaglib.cc:
	  Add gtk-doc blurb (unused for the time being); match registered
	  plugin name to the filename of the plugin (taglibmux => taglib)

2006-03-13 15:49:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps):
	  * ext/esd/esdmon.c: (gst_esdmon_get):
	  * ext/flac/gstflactag.c: (gst_flac_tag_chain):
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
	  (gst_gdk_pixbuf_sink_getcaps):
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps),
	  (gst_jpegenc_setcaps):
	  * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
	  (gst_smokeenc_setcaps):
	  * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink),
	  (gst_mngdec_src_getcaps):
	  * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink),
	  (gst_mngenc_chain):
	  * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps):
	  * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink):
	  * ext/speex/gstspeexdec.c: (speex_dec_convert),
	  (speex_dec_src_event), (speex_dec_chain):
	  * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect),
	  (gst_avimux_audsinkconnect), (gst_avimux_handle_event):
	  * gst/debug/negotiation.c: (gst_negotiation_getcaps),
	  (gst_negotiation_pad_link), (gst_negotiation_chain):
	  * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler),
	  (gst_flxdec_chain):
	  * gst/interleave/deinterleave.c: (deinterleave_sink_link),
	  (deinterleave_chain):
	  * gst/law/mulaw-encode.c: (mulawenc_setcaps):
	  * gst/median/gstmedian.c: (gst_median_link):
	  * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect),
	  (gst_monoscope_chain):
	  * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect):
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps):
	  * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain):
	  * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get):
	  close #333784 unref the result of gst_pad_get_parent()
	  by: Christophe Fergeau.

2006-03-13 10:05:09 +0000  Julien Moutte <julien@moutte.net>

	  Fix build of v4l2 (sigh)
	  Original commit message from CVS:
	  2006-03-13  Julien MOUTTE  <julien@moutte.net>
	  * docs/plugins/gst-plugins-bad-plugins-decl-list.txt:
	  * sys/v4l2/Makefile.am: Fix build of v4l2 (sigh)

2006-03-12 15:33:00 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/v4l2/v4l2src_calls.c: g_atomic_int_set is only available in glib-0.10, use gst_atomic_int_et instead.
	  Original commit message from CVS:
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_capture_init),
	  (gst_v4l2src_buffer_pool_free):
	  g_atomic_int_set is only available in glib-0.10, use gst_atomic_int_et
	  instead.

2006-03-12 15:25:51 +0000  Edward Hervey <bilboed@bilboed.com>

	  sys/v4l2/gstv4l2element.h: Remove tim's addition of "_stdint.h" since it doesn't make the PPC buildbot happy.
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2element.h:
	  Remove tim's addition of "_stdint.h" since it doesn't make the PPC
	  buildbot happy.
	  I will just use the same comment Ronald used when he added these lines:
	  Yet Another Hack (tm) for kernel header borkedness.

2006-03-12 15:02:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/taglib/: Add support for writing MusicBrainz IDs.
	  Original commit message from CVS:
	  * ext/taglib/Makefile.am:
	  * ext/taglib/gsttaglib.cc:
	  * ext/taglib/gsttaglib.h:
	  Add support for writing MusicBrainz IDs.

2006-03-12 14:43:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/v4l2/gstv4l2element.h: Include "_stdint.h" in an attempt to make the
	  Original commit message from CVS:
	  * sys/v4l2/gstv4l2element.h:
	  Include "_stdint.h" in an attempt to make the
	  PPC-buildbot happy.

2006-03-12 11:00:33 +0000  Christophe Fergeau <teuf@gnome.org>

	  ext/lame/gstlame.c: mark the xing-header property as BROKEN (see http://bugzilla.gnome.org/show_bug.cgi?id=330317#c19...
	  Original commit message from CVS:
	  2006-03-12  Christophe Fergeau  <teuf@gnome.org>
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * ext/lame/gstlame.c: (gst_lame_class_init): mark the xing-header
	  property as BROKEN (see
	  http://bugzilla.gnome.org/show_bug.cgi?id=330317#c19 for an
	  explanation why it's broken).

2006-03-11 22:50:03 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2element.c:
	* sys/v4l2/gstv4l2element.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/gstv4l2xoverlay.h:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  V4L2 ported to 0.10.
	  Original commit message from CVS:
	  V4L2 ported to 0.10.

2006-03-11 10:58:08 +0000  Alex Lancaster <alexlan@fedoraproject.org>

	  ext/taglib/gsttaglib.cc: and add support for TCOP (copyright)
	  Original commit message from CVS:
	  2006-03-11  Christophe Fergeau  <teuf@gnome.org>
	  Patch by: Alex Lancaster
	  * ext/taglib/gsttaglib.cc: fix writing of TPOS tags (album number),
	  and add support for TCOP (copyright)

2006-03-09 20:02:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Fix build with gcc-4.1 (#327355).
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_send_event):
	  Fix build with gcc-4.1 (#327355).

2006-03-09 17:44:17 +0000  Christophe Fergeau <teuf@gnome.org>

	  new id3v2 muxer based on TagLib
	  Original commit message from CVS:
	  2006-03-09  Christophe Fergeau  <teuf@gnome.org>
	  reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * configure.ac:
	  * ext/Makefile.am:
	  * ext/taglib/Makefile.am:
	  * ext/taglib/gsttaglib.cc:
	  * ext/taglib/gsttaglib.h: new id3v2 muxer based on TagLib

2006-03-09 11:47:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/dv/gstdvdemux.c: Handle events in push mode better, can now do non-flushing seeks in push mode as well.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event),
	  (gst_dvdemux_convert_segment), (gst_dvdemux_demux_frame):
	  Handle events in push mode better, can now do non-flushing
	  seeks in push mode as well.

2006-03-08 12:16:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Extract disc number and count from files that use 'disk' instead of 'disc' as node identifier ...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
	  Extract disc number and count from files that use
	  'disk' instead of 'disc' as node identifier for that
	  (fixes #332066).

2006-03-07 17:31:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstdynudpsink.c: Applied patch from Kai Vehmanen, fixes #333624.
	  Original commit message from CVS:
	  * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init):
	  Applied patch from Kai Vehmanen, fixes #333624.

2006-03-06 22:22:45 +0000  Julien Moutte <julien@moutte.net>

	  ext/libpng/gstpngdec.c: Implement paletted and grayscale png files handling. (#150363).
	  Original commit message from CVS:
	  2006-03-06  Julien MOUTTE  <julien@moutte.net>
	  * ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
	  Implement paletted and grayscale png files handling.
	  (#150363).

2006-03-06 00:10:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/speex/gstspeexenc.c: fix a tag list assert follow gst-plugins-base/ext/ogg/README; set OFFSET and OFFSET_END.  Mu...
	  Original commit message from CVS:
	  * ext/speex/gstspeexenc.c: (gst_speexenc_set_header_on_caps),
	  (gst_speexenc_chain):
	  fix a tag list assert
	  follow gst-plugins-base/ext/ogg/README; set OFFSET
	  and OFFSET_END.  Muxes correctly with gst-plugins-base
	  > 0.9.3

2006-03-05 13:03:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Add support for '3IVD' fourcc (#333403).
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add support for '3IVD' fourcc (#333403).

2006-03-04 20:11:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/: Use new typefind helper functions here as well, and do typefinding in pull-mode if upstream supports t...
	  Original commit message from CVS:
	  * gst/id3demux/Makefile.am:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
	  (gst_id3demux_chain), (gst_id3demux_sink_activate):
	  Use new typefind helper functions here as well, and
	  do typefinding in pull-mode if upstream supports that.

2006-03-04 18:57:37 +0000  Benjamin Pineau <ben.pineau@gmail.com>

	  sys/sunaudio/: Remove unused variables, breaks build from CVS
	  Original commit message from CVS:
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  (gst_sunaudiomixer_ctrl_get_volume),
	  (gst_sunaudiomixer_ctrl_set_volume):
	  * sys/sunaudio/gstsunaudiomixertrack.c:
	  (gst_sunaudiomixer_track_new):
	  Remove unused variables, breaks build from CVS
	  with -Werror (#333392, patch by: Benjamin Pineau)

2006-03-03 23:45:23 +0000  Sébastien Moutte <sebastien@moutte.net>

	  sys/: sinks are now using GST_RANK_PRIMARY to be used with autodectection
	  Original commit message from CVS:
	  * sys/directdraw:
	  * sys/directsound:
	  sinks are now using GST_RANK_PRIMARY to be used with autodectection
	  * win32/vs6:
	  project files updated to fix some bugs
	  * win32/vs7:
	  * win32/vs8:
	  vs7 and vs8 project files added

2006-03-03 18:36:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Added wavparse docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Added wavparse docs.
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
	  (gst_wavparse_reset), (gst_wavparse_init),
	  (gst_wavparse_create_sourcepad), (gst_wavparse_parse_file_header),
	  (gst_wavparse_stream_init), (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_send_event),
	  (gst_wavparse_stream_data), (gst_wavparse_loop),
	  (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull),
	  (gst_wavparse_change_state):
	  * gst/wavparse/gstwavparse.h:
	  Implement seek in READY (fixes #327658)
	  Added docs and did some cleanups.

2006-03-03 17:51:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.*: If we have an index, use a duration based on the index instead of blindly trusting the informa...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	  (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_calculate_durations_from_index),
	  (gst_avi_demux_stream_header):
	  * gst/avi/gstavidemux.h:
	  If we have an index, use a duration based on the index instead
	  of blindly trusting the information in the stream headers
	  (fixes #331817).

2006-03-03 15:50:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Added smoke and jpeg to the docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  Added smoke and jpeg to the docs.
	  * ext/jpeg/Makefile.am:
	  * ext/jpeg/gstjpeg.c: (plugin_init):
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
	  * ext/jpeg/gstjpegenc.h:
	  * ext/jpeg/gstsmokedec.c: (gst_smokedec_init),
	  (gst_smokedec_chain):
	  * ext/jpeg/gstsmokedec.h:
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
	  * ext/jpeg/gstsmokeenc.h:
	  * ext/jpeg/smokecodec.h:
	  Port smokedec (fixes #331905).
	  Added some docs.
	  Some cleanups.

2006-03-03 14:39:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Added videobalance and videoflip to the docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  Added videobalance and videoflip to the docs.
	  * gst/videofilter/Makefile.am:
	  * gst/videofilter/gstvideobalance.c:
	  (gst_video_balance_update_tables_planar411),
	  (gst_video_balance_is_passthrough),
	  (gst_video_balance_update_properties), (oil_tablelookup_u8),
	  (gst_video_balance_planar411_ip), (gst_video_balance_set_caps),
	  (gst_video_balance_transform_ip), (gst_video_balance_base_init),
	  (gst_video_balance_finalize), (gst_video_balance_class_init),
	  (gst_video_balance_init), (gst_video_balance_interface_supported),
	  (gst_video_balance_interface_init),
	  (gst_video_balance_colorbalance_list_channels),
	  (gst_video_balance_colorbalance_set_value),
	  (gst_video_balance_colorbalance_get_value),
	  (gst_video_balance_colorbalance_init),
	  (gst_video_balance_set_property), (gst_video_balance_get_property),
	  (gst_video_balance_get_type), (plugin_init):
	  * gst/videofilter/gstvideobalance.h:
	  Ported to 0.10. (Fixes #326160)
	  Added docs.
	  * gst/videofilter/gstvideoflip.c:
	  * gst/videofilter/gstvideoflip.h:
	  Added docs.

2006-03-03 11:07:41 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Use GST_WARNING instead of GST_ERROR for all the too short/long atoms when parsing.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak):
	  Use GST_WARNING instead of GST_ERROR for all the too short/long atoms
	  when parsing.
	  Also let's be a bit less vulgar in our warning messages :)

2006-03-02 15:14:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump requirements to current core and -base CVS (core for new typefind helper API, and -base for the
	  Original commit message from CVS:
	  * configure.ac:
	  Bump requirements to current core and -base CVS
	  (core for new typefind helper API, and -base for the
	  WAVFORMATEX support that was added to libgstriff and
	  is needed by wavparse).
	  * gst/apetag/Makefile.am:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain),
	  (gst_tag_demux_sink_activate):
	  Use new typefind helpers for typefinding instead of our
	  home-grown stuff; also, do typefinding in pull-mode if
	  upstream supports that.

2006-02-28 11:59:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Can't divide through zero (suppress warning in case of stream with one single still picture) (...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  Can't divide through zero (suppress warning in case of
	  stream with one single still picture) (see #327083)

2006-02-28 10:40:01 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	  remove conflict indicator
	  Original commit message from CVS:
	  remove conflict indicator

2006-02-28 10:39:08 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	  add missing entry
	  Original commit message from CVS:
	  add missing entry

2006-02-28 10:29:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Use DEBUG_OBJECT more.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data),
	  (gst_wavparse_pad_convert), (gst_wavparse_srcpad_event),
	  (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull):
	  Use DEBUG_OBJECT more.

2006-02-28 10:22:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Added dvdec and dvdemux to docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Added dvdec and dvdemux to docs.
	  * ext/dv/gstdvdec.c: (gst_dvdec_base_init), (gst_dvdec_chain):
	  Added docs.
	  Check frame sizes so we don't crash when don't have enough
	  data.
	  Send nice error messages on error.
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_base_init),
	  (gst_dvdemux_class_init), (gst_dvdemux_init),
	  (gst_dvdemux_finalize), (gst_dvdemux_reset),
	  (gst_dvdemux_src_convert), (gst_dvdemux_sink_convert),
	  (gst_dvdemux_src_query), (gst_dvdemux_sink_query),
	  (gst_dvdemux_push_event), (gst_dvdemux_handle_sink_event),
	  (gst_dvdemux_convert_src_pair), (gst_dvdemux_convert_sink_pair),
	  (gst_dvdemux_convert_src_to_sink), (gst_dvdemux_handle_push_seek),
	  (gst_dvdemux_do_seek), (gst_dvdemux_handle_pull_seek),
	  (gst_dvdemux_handle_src_event), (gst_dvdemux_demux_audio),
	  (gst_dvdemux_demux_video), (gst_dvdemux_demux_frame),
	  (gst_dvdemux_flush), (gst_dvdemux_chain), (gst_dvdemux_loop),
	  (gst_dvdemux_sink_activate_push), (gst_dvdemux_sink_activate_pull),
	  (gst_dvdemux_sink_activate), (gst_dvdemux_change_state):
	  * ext/dv/gstdvdemux.h:
	  Added docs.
	  Implement pull mode.
	  Fix memleaks.
	  Reduce memcpy for the video demuxing.

2006-02-28 09:21:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/annodex/: Add a little extra debug. Make the decoder not return NOT_LINKED, as we want to continue decoding all C...
	  Original commit message from CVS:
	  * ext/annodex/gstcmmldec.c: (gst_cmml_dec_sink_event),
	  (gst_cmml_dec_new_buffer), (gst_cmml_dec_parse_preamble),
	  (gst_cmml_dec_parse_head), (gst_cmml_dec_push_clip):
	  * ext/annodex/gstcmmlparser.c: (gst_cmml_parser_parse_chunk):
	  Add a little extra debug. Make the decoder not return NOT_LINKED,
	  as we want to continue decoding all CMML and emitting tags.

2006-02-27 14:37:29 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add annodex plugin
	  Original commit message from CVS:
	  add annodex plugin

2006-02-27 14:00:18 +0000  Michael Smith <msmith@xiph.org>

	  ext/annodex/gstskeltag.*: Deleted; these files aren't used any more either.
	  Original commit message from CVS:
	  * ext/annodex/gstskeltag.c:
	  * ext/annodex/gstskeltag.h:
	  Deleted; these files aren't used any more either.

2006-02-25 20:37:29 +0000  Julien Moutte <julien@moutte.net>

	  ext/Makefile.am: Fix dist-check.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * ext/Makefile.am: Fix dist-check.

2006-02-25 19:36:24 +0000  Julien Moutte <julien@moutte.net>

	  ext/annodex/gstcmmlenc.c: Fix another memleak.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_push_clip): Fix another
	  memleak.

2006-02-25 19:07:41 +0000  Julien Moutte <julien@moutte.net>

	  Fix a memleak in gst_cmml_track_list_add_clip.
	  Original commit message from CVS:
	  2006-02-25  Alessandro Decina <alessandro@nnva.org>
	  * ext/annodex/Makefile.am:
	  * ext/annodex/gstannodex.c:
	  * ext/annodex/gstcmmldec.c:
	  * ext/annodex/gstcmmlenc.c:
	  * ext/annodex/gstcmmlparser.c:
	  * ext/annodex/gstcmmlparser.h:
	  * ext/annodex/gstcmmlutils.c:
	  * tests/check/elements/cmmldec.c:
	  * tests/check/elements/cmmlenc.c:
	  Fix a memleak in gst_cmml_track_list_add_clip.
	  Handle overflows in clip's start and end times.
	  Add the "encoded" parameter to cmmldec and cmmlenc caps.
	  Do not parse junk at the end of a CMML preamble buffer.
	  Register a libxml error handler to not print stuff on stderr.
	  Check for bad clip start and end times in the testsuites.

2006-02-25 11:37:10 +0000  Julien Moutte <julien@moutte.net>

	  ext/annodex/: Fix possible memleaks.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * ext/annodex/gstcmmldec.c: (gst_cmml_dec_class_init),
	  (gst_cmml_dec_finalize), (gst_cmml_dec_change_state):
	  * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_class_init),
	  (gst_cmml_enc_finalize), (gst_cmml_enc_change_state):
	  * ext/annodex/gstcmmlutils.c: (gst_cmml_track_list_destroy): Fix
	  possible memleaks.

2006-02-24 23:52:28 +0000  Julien Moutte <julien@moutte.net>

	  tests/check/: Fix tests so that they use the plugins-base tags.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * tests/check/Makefile.am:
	  * tests/check/elements/cmmldec.c:
	  * tests/check/elements/cmmlenc.c: Fix tests so that they use
	  the plugins-base tags.

2006-02-24 23:36:58 +0000  Julien Moutte <julien@moutte.net>

	  ext/Makefile.am: Re-enable module.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * ext/Makefile.am: Re-enable module.

2006-02-24 23:32:14 +0000  Julien Moutte <julien@moutte.net>

	  tests/check/Makefile.am: Forgot to remove that test.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * tests/check/Makefile.am: Forgot to remove that test.

2006-02-24 23:31:08 +0000  Julien Moutte <julien@moutte.net>

	  Try to fix Annodex plugin.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * ext/annodex/Makefile.am:
	  * ext/annodex/gstannodex.c: (plugin_init):
	  * ext/annodex/gstcmmldec.c:
	  * ext/annodex/gstskeldec.c:
	  * ext/annodex/gstskeldec.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/skeldec.c: Try to fix Annodex plugin.

2006-02-24 23:06:27 +0000  Julien Moutte <julien@moutte.net>

	  tests/check/Makefile.am: Disable those checks as well.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * tests/check/Makefile.am: Disable those checks as well.

2006-02-24 22:49:29 +0000  Julien Moutte <julien@moutte.net>

	  ext/Makefile.am: Disable annodex for now until we figure out how to make it build.
	  Original commit message from CVS:
	  2006-02-24  Julien MOUTTE  <julien@moutte.net>
	  * ext/Makefile.am: Disable annodex for now until we figure out
	  how to make it build.
	  * ext/gdk_pixbuf/Makefile.am: Note for Thomas :
	  Add a rule to your checklist : "please try to at least build
	  what you are going to commit into -good, or if you are too lazy
	  to do that, please check that the buildbots are not crying because
	  of your commit."

2006-02-24 19:51:29 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* configure.ac:
	* ext/Makefile.am:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.h:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/gdk_pixbuf/pixbufscale.h:
	  I'm too lazy to comment this
	  Original commit message from CVS:
	  Gdkpixbuf ported from 0.8 to 0.10 by Renato Filho <renato.filho@indt.org.br>. gst_loader and gdkpixbufanimation still need port.

2006-02-24 19:49:32 +0000  Fabrizio Gennari <fabrizio.ge@tiscali.it>

	  gst/qtdemux/qtdemux.c: Add support for palettised Apple SMC videos (#327075, based on
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
	  (qtdemux_parse_trak), (qtdemux_video_caps):
	  Add support for palettised Apple SMC videos (#327075, based on
	  patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>).

2006-02-24 19:07:10 +0000  Michael Smith <msmith@xiph.org>

	  Add Annodex elements from Alessendro Decina: skeleton and CMML.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * ext/Makefile.am:
	  * ext/annodex/Makefile.am:
	  * ext/annodex/gstannodex.c:
	  * ext/annodex/gstannodex.h:
	  * ext/annodex/gstcmmldec.c:
	  * ext/annodex/gstcmmldec.h:
	  * ext/annodex/gstcmmlenc.c:
	  * ext/annodex/gstcmmlenc.h:
	  * ext/annodex/gstcmmlparser.c:
	  * ext/annodex/gstcmmlparser.h:
	  * ext/annodex/gstcmmltag.c:
	  * ext/annodex/gstcmmltag.h:
	  * ext/annodex/gstcmmlutils.c:
	  * ext/annodex/gstcmmlutils.h:
	  * ext/annodex/gstskeldec.c:
	  * ext/annodex/gstskeldec.h:
	  * ext/annodex/gstskeltag.c:
	  * ext/annodex/gstskeltag.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/cmmldec.c:
	  * tests/check/elements/cmmlenc.c:
	  * tests/check/elements/skeldec.c:
	  Add Annodex elements from Alessendro Decina: skeleton and CMML.
	  Includes tests & docs, oh my! Passes Thomas's -good checklist
	  entirely. Wow.

2006-02-24 17:09:56 +0000  Michael Smith <msmith@xiph.org>

	  autogen.sh: Check for automake 1.9 as well.
	  Original commit message from CVS:
	  * autogen.sh:
	  Check for automake 1.9 as well.

2006-02-24 14:49:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacenc.c: Change min. sample rate to 8kHz to match flacdec's.
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c:
	  Change min. sample rate to 8kHz to match flacdec's.

2006-02-23 20:08:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cdio/Makefile.am: Add GST_BASE_CFLAGS and GST_BASE_LIBS (seems to be required for Cygwin, see #317048)
	  Original commit message from CVS:
	  * ext/cdio/Makefile.am:
	  Add GST_BASE_CFLAGS and GST_BASE_LIBS (seems to be
	  required for Cygwin, see #317048)
	  * gst/rtp/gstasteriskh263.c:
	  Cygwin has includes for both the unix network socket API
	  and the windows API, but only one can be included, so fix
	  includes to only use one or the other, prefering the unxi
	  one (#317048).

2006-02-23 12:21:25 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  rtp/gst/: Separated the G711 payloaders/depayloaders into separate elements for mulaw/alaw. Also removed the old g711...
	  Original commit message from CVS:
	  2006-02-23  Philippe Kalaf  <philippe.kalaf at collabora.co.uk>
	  * rtp/gst/gstrtppcmadepay.c:
	  * rtp/gst/gstrtppcmadepay.h:
	  * rtp/gst/gstgstrtppcmapay.c:
	  * rtp/gst/gstgstrtppcmapay.h:
	  * rtp/gst/gstrtppcmudepay.c:
	  * rtp/gst/gstrtppcmudepay.h:
	  * rtp/gst/gstrtppcmupay.c:
	  * rtp/gst/gstrtppcmupay.h:
	  * rtp/gst/Makefile.am:
	  * rtp/gst/gstrtp.c:
	  * rtp/gst/README:
	  Separated the G711 payloaders/depayloaders into separate elements for
	  mulaw/alaw. Also removed the old g711 payloaders/depayloaders.

2006-02-22 20:22:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/dv/: Ueber spiffify some more, added debug category.
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c: (gst_dvdec_base_init), (gst_dvdec_init),
	  (gst_dvdec_change_state):
	  * ext/dv/gstdvdec.h:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_base_init), (gst_dvdemux_init),
	  (gst_dvdemux_src_convert), (gst_dvdemux_sink_convert),
	  (gst_dvdemux_src_query), (gst_dvdemux_sink_query),
	  (gst_dvdemux_handle_sink_event), (gst_dvdemux_demux_frame),
	  (gst_dvdemux_flush), (gst_dvdemux_chain),
	  (gst_dvdemux_change_state):
	  * ext/dv/gstdvdemux.h:
	  Ueber spiffify some more, added debug category.
	  Use _scale.
	  Use segments, respect playback rate from newsegment.
	  Fix refcount issue.

2006-02-22 09:33:25 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Add 'dvsd' and 'dv25' to list of possible fourcc values for DV Video.
	  Original commit message from CVS:
	  Reviewed by : Edward Hervey <edward@fluendo.com>
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add 'dvsd' and 'dv25' to list of possible fourcc values for DV Video.
	  Add image/png for fourcc 'png '

2006-02-20 21:19:59 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Port ximagesrc to 0.10 (Closes #304795)
	  Original commit message from CVS:
	  * configure.ac:
	  * sys/Makefile.am:
	  * sys/ximagesrc/Makefile.am:
	  * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_return_buf),
	  (gst_ximagesrc_open_display), (gst_ximagesrc_start),
	  (gst_ximagesrc_stop), (gst_ximagesrc_unlock),
	  (gst_ximagesrc_recalc), (composite_pixel),
	  (gst_ximagesrc_ximage_get), (gst_ximagesrc_create),
	  (gst_ximagesrc_set_property), (gst_ximagesrc_get_property),
	  (gst_ximagesrc_clear_bufpool), (gst_ximagesrc_base_init),
	  (gst_ximagesrc_dispose), (gst_ximagesrc_finalize),
	  (gst_ximagesrc_get_caps), (gst_ximagesrc_set_caps),
	  (gst_ximagesrc_fixate), (gst_ximagesrc_class_init),
	  (gst_ximagesrc_init), (plugin_init):
	  * sys/ximagesrc/ximagesrc.h:
	  * sys/ximagesrc/ximageutil.c: (ximageutil_handle_xerror),
	  (ximageutil_check_xshm_calls), (ximageutil_xcontext_get),
	  (ximageutil_xcontext_clear),
	  (ximageutil_calculate_pixel_aspect_ratio),
	  (gst_ximagesrc_buffer_finalize), (gst_ximage_buffer_free),
	  (gst_ximagesrc_buffer_init), (gst_ximagesrc_buffer_class_init),
	  (gst_ximagesrc_buffer_get_type), (gst_ximageutil_ximage_new),
	  (gst_ximageutil_ximage_destroy):
	  * sys/ximagesrc/ximageutil.h:
	  Port ximagesrc to 0.10 (Closes #304795)

=== release 0.10.1 ===

2006-02-20 19:12:10 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: releasing 0.10.1, "Slimy - yet satisfying"
	  Original commit message from CVS:
	  2006-02-20  Jan Schmidt <thaytan@mad.scientist.com>
	  * configure.ac:
	  releasing 0.10.1, "Slimy - yet satisfying"

2006-02-20 13:08:50 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/ladspa/gstsignalprocessor.c: Fix compilation of LADPSA. It doesn't seem to work, and isn't enabled for the build,...
	  Original commit message from CVS:
	  * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_event),
	  (gst_signal_processor_process):
	  Fix compilation of LADPSA. It doesn't seem to work, and isn't
	  enabled for the build, but it helps me win the feature-count
	  competitions ooh yeah.

2006-02-19 16:02:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Use scaling code for added precission and more correct stop position in case scale==0.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_src_convert),
	  (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
	  (gst_avi_demux_parse_file_header), (gst_avi_demux_stream_init),
	  (gst_avi_demux_parse_avih), (gst_avi_demux_parse_superindex),
	  (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_stream_header), (gst_avi_demux_change_state):
	  Use scaling code for added precission and more correct stop
	  position in case scale==0.

2006-02-19 12:09:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/flx/gstflxdec.*: Implement DURATION query.
	  Original commit message from CVS:
	  * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler),
	  (gst_flxdec_chain):
	  * gst/flx/gstflxdec.h:
	  Implement DURATION query.

2006-02-19 11:57:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/flx/: Set MALLOCDATA for the temp buffers so we don't leak.
	  Original commit message from CVS:
	  * gst/flx/flx_color.h:
	  * gst/flx/flx_fmt.h:
	  * gst/flx/gstflxdec.c: (gst_flxdec_init),
	  (gst_flxdec_src_query_handler), (flx_decode_color),
	  (gst_flxdec_chain):
	  * gst/flx/gstflxdec.h:
	  Set MALLOCDATA for the temp buffers so we don't leak.
	  Some debug cleanups.
	  Consume all data in the adapter before leaving the chain
	  function. Fixes #330678.

2006-02-18 20:48:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/: Handle 0 data size in otherwise valid frames.
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
	  * gst/id3demux/id3v2frames.c: (id3v2_genre_fields_to_taglist):
	  Handle 0 data size in otherwise valid frames.
	  Handle numeric strings in 2.4.0 even when not in parentheses

2006-02-18 17:20:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Recognise SSA/ASS and USF subtitle formats and set proper caps when they are found.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_subtitle_caps),
	  (gst_matroska_demux_plugin_init):
	  * gst/matroska/matroska-ids.h:
	  Recognise SSA/ASS and USF subtitle formats and
	  set proper caps when they are found.

2006-02-17 18:25:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Don't GST_LOG timestamps from nonexistent index entries (#331582).
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
	  Don't GST_LOG timestamps from nonexistent index
	  entries (#331582).

2006-02-17 17:54:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/gstjpegdec.c: Fix invalid memory access for some odd-sized images (see image contained in quicktime stream i...
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_decode_direct),
	  (gst_jpeg_dec_chain):
	  Fix invalid memory access for some odd-sized images
	  (see image contained in quicktime stream in #327083);
	  use g_malloc() instead of g_alloca().

2006-02-17 16:28:29 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Check that the size of the returned buffer is of the correct size because the parser assumes t...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header):
	  Check that the size of the returned buffer is of the correct size
	  because the parser assumes that.
	  Fixes #331543.

2006-02-17 15:37:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpamrdepay.c: Patch from Sebastien Cote, fixes #319884
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_chain):
	  Patch from Sebastien Cote, fixes #319884

2006-02-17 11:19:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cdio/gstcdio.c: Init debug category (#331253).
	  Original commit message from CVS:
	  * ext/cdio/gstcdio.c: (plugin_init):
	  Init debug category (#331253).

2006-02-17 10:53:38 +0000  Christian Schaller <uraeus@gnome.org>

	* ext/gconf/gconf.c:
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.c:
	* ext/gconf/gstgconfaudiosink.h:
	* gconf/gstreamer.schemas.in:
	* gst-plugins-good.spec.in:
	  add Jurg's patch for multidevice support
	  Original commit message from CVS:
	  add Jurg's patch for multidevice support

2006-02-16 20:30:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: Pass extra_data to gst_riff_create_audio_caps(), so that
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Pass extra_data to gst_riff_create_audio_caps(), so that
	  WAVEFORMATEX stuff works. Post audio codec name and post
	  it as taglist on the bus. Allow up to 8 channesl for raw
	  PCM in the source pad template caps.

2006-02-16 17:16:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/lame/gstlame.c: Fix up lame a bit.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain),
	  (gst_lame_change_state):
	  Fix up lame a bit.
	  Apply patch #319782 by Gautier Portet.

2006-02-16 16:53:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/multipart/multipartdemux.c: Applied #318663. Gives quite a few false positives in autoscan mode, but it's better ...
	  Original commit message from CVS:
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
	  (gst_multipart_demux_class_init), (gst_multipart_demux_init),
	  (gst_multipart_demux_finalize), (gst_multipart_find_pad_by_mime),
	  (gst_multipart_demux_chain), (gst_multipart_demux_change_state),
	  (gst_multipart_set_property), (gst_multipart_get_property):
	  Applied #318663. Gives quite a few false positives in
	  autoscan mode, but it's better than nothing. Not closing yet.

2006-02-16 14:13:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Update documentation.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/inspect/plugin-udp.xml:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	  (gst_udpsrc_start):
	  Update documentation.
	  Fix args.

2006-02-16 14:02:57 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Don't stop the task if the pad isn't linked.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_event),
	  (gst_qtdemux_loop), (qtdemux_sink_activate_pull):
	  Don't stop the task if the pad isn't linked.

2006-02-16 10:58:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3tags.c: ID3 2.3.0 used synch-safe integers for the tag size, but not for the frame size. (Fixes #331368)
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
	  ID3 2.3.0 used synch-safe integers for the tag size, but not for the
	  frame size. (Fixes #331368)

2006-02-16 10:42:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/README: Updated README.
	  Original commit message from CVS:
	  * gst/rtsp/README:
	  Updated README.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
	  (gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
	  (gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp):
	  * gst/rtsp/gstrtspsrc.h:
	  Make sure the RTP port is an even port an try to allocate
	  another if not.
	  Added retry property to control max retries for port allocation.
	  Make sure RTCP port is RTP port+1.
	  Cleanup when port allocation fails.
	  Fixes #319183.

2006-02-16 09:17:58 +0000  Wouter Paesen <wouter@kangaroot.net>

	  gst/alpha/gstalpha.c: Don't ignore return value of the parent class's state
	  Original commit message from CVS:
	  * gst/alpha/gstalpha.c: (gst_alpha_change_state):
	  Don't ignore return value of the parent class's state
	  change function (#331385, patch by: Wouter Paesen).

2006-02-15 12:17:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Add HAL sound device wrapper plugins. Closes #329106
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * ext/Makefile.am:
	  * ext/hal/Makefile.am:
	  * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init),
	  (gst_hal_audio_sink_class_init), (gst_hal_audio_sink_reset),
	  (gst_hal_audio_sink_init), (gst_hal_audio_sink_dispose),
	  (do_toggle_element), (gst_hal_audio_sink_set_property),
	  (gst_hal_audio_sink_get_property),
	  (gst_hal_audio_sink_change_state):
	  * ext/hal/gsthalaudiosink.h:
	  * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init),
	  (gst_hal_audio_src_class_init), (gst_hal_audio_src_reset),
	  (gst_hal_audio_src_init), (gst_hal_audio_src_dispose),
	  (do_toggle_element), (gst_hal_audio_src_set_property),
	  (gst_hal_audio_src_get_property), (gst_hal_audio_src_change_state):
	  * ext/hal/gsthalaudiosrc.h:
	  * ext/hal/gsthalelements.c: (plugin_init):
	  * ext/hal/gsthalelements.h:
	  * ext/hal/hal.c: (gst_hal_get_string),
	  (gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
	  (gst_hal_get_audio_src):
	  * ext/hal/hal.h:
	  Add HAL sound device wrapper plugins. Closes #329106

2006-02-15 12:13:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: It appears 100% equals 1/1 and not 100/1 ...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_chain):
	  It appears 100% equals 1/1 and not 100/1 ...

2006-02-15 10:15:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Add comment in a fultile attempt to stop the copy-and-paste paradigm leading to duplication of...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
	  Add comment in a fultile attempt to stop the copy-and-paste
	  paradigm leading to duplication of bad code.
	  * gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
	  Mime parameters have to be checked case insensitive

2006-02-15 09:45:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: When buffering MDAT data, show the user something is happening by posting 'buffering' messages...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_post_buffering),
	  (gst_qtdemux_chain):
	  When buffering MDAT data, show the user something is
	  happening by posting 'buffering' messages on the bus.

2006-02-14 23:23:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: Advance stream time for lagging subtitle streams by sending newsegment events with the...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams):
	  Advance stream time for lagging subtitle streams by sending
	  newsegment events with the update flag set.

2006-02-14 18:50:13 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.*: Make push-based work if mdat atom is before moov atom.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
	  (gst_qtdemux_handle_src_query), (gst_qtdemux_change_state),
	  (next_entry_size), (gst_qtdemux_chain):
	  * gst/qtdemux/qtdemux.h:
	  Make push-based work if mdat atom is before moov atom.
	  Don't answer duration query. This should be transformed into replying
	  FALSE to seek events.

2006-02-14 16:58:30 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: There can be bogus data before the hdrl LIST tag in the RIFF header.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header):
	  There can be bogus data before the hdrl LIST tag in the RIFF header.
	  It's hard to say if it's not respecting the AVI specifications or not,
	  but since Google Video is producing AVIs like that and the other player
	  don't seem to complain, I guess we should do the same.

2006-02-14 11:24:53 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Handle the case where data atoms are before moov atoms in push-based mode.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (next_entry_size), (gst_qtdemux_chain):
	  Handle the case where data atoms are before moov atoms in push-based mode.
	  Errors out gracefully.

2006-02-13 22:04:42 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/: QtDemux can now work push-based.
	  Original commit message from CVS:
	  * gst/qtdemux/Makefile.am:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
	  (gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state),
	  (extract_initial_length_and_fourcc),
	  (gst_qtdemux_loop_state_header), (gst_qtdemux_loop_state_movie),
	  (gst_qtdemux_loop_header), (next_entry_size), (gst_qtdemux_chain),
	  (qtdemux_sink_activate), (qtdemux_sink_activate_pull),
	  (qtdemux_sink_activate_push), (qtdemux_parse_trak):
	  * gst/qtdemux/qtdemux.h:
	  QtDemux can now work push-based.
	  It still needs some love for seeking.

2006-02-13 12:00:51 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3v2frames.c: Add more validation to ensure that a char encoding conversion produced a valid UTF-8 string.
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (parse_insert_string_field),
	  (parse_split_strings):
	  Add more validation to ensure that a char encoding conversion
	  produced a valid UTF-8 string.

2006-02-13 10:43:15 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: Properly handle end of segment. Closes #330885.
	  Original commit message from CVS:
	  Reviewed by: Edward Hervey  <edward@fluendo.com>
	  * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
	  Properly handle end of segment. Closes #330885.

2006-02-13 10:36:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4gpay.h: For got to commit this one.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4gpay.h:
	  For got to commit this one.

2006-02-12 18:59:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4gpay.*: Make more things work.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
	  (gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_parse_audio_config),
	  (gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
	  (gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush):
	  * gst/rtp/gstrtpmp4gpay.h:
	  Make more things work.
	  Handle ACC config strings.

2006-02-12 13:10:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/rtp/gstrtpamrpay.c: set timestamps if no incoming timestamps set
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
	  set timestamps if no incoming timestamps set

2006-02-11 13:54:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/apetag/gsttagdemux.c: ... and fix the very same leaks in GstTagDemux.
	  Original commit message from CVS:
	  * gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size),
	  (gst_tag_demux_do_typefind):
	  ... and fix the very same leaks in GstTagDemux.

2006-02-11 13:35:13 +0000  Jon Trowbridge <trow@ximian.com>

	  gst/id3demux/gstid3demux.c:
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size),
	  (gst_id3demux_do_typefind):
	  Fix a couple of mem leaks. (Patch by Jonathan Matthew
	  <jonathan at kaolin dot wh9 dot net>)

2006-02-10 17:37:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4vpay.c: First set options, then set caps or else the baseclass will not know about the options, duh.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_setcaps):
	  First set options, then set caps or else the baseclass
	  will not know about the options, duh.

2006-02-10 17:16:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4vpay.c: Don't waste time looking for a config string if we have codec_info on the incomming caps.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init),
	  (gst_rtp_mp4v_pay_setcaps):
	  Don't waste time looking for a config string if we have codec_info
	  on the incomming caps.

2006-02-10 16:40:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/README: Say something about case-sensitivity of caps vs mime-attributes.
	  Original commit message from CVS:
	  * gst/rtp/README:
	  Say something about case-sensitivity of caps vs mime-attributes.
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init),
	  (gst_rtp_amr_pay_handle_buffer):
	  * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_get_type),
	  (gst_rtp_mp4g_pay_base_init), (gst_rtp_mp4g_pay_class_init),
	  (gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_finalize),
	  (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps),
	  (gst_rtp_mp4g_pay_flush), (gst_rtp_mp4g_pay_handle_buffer),
	  (gst_rtp_mp4g_pay_set_property), (gst_rtp_mp4g_pay_get_property),
	  (gst_rtp_mp4g_pay_plugin_init):
	  * gst/rtp/gstrtpmp4gpay.h:
	  Added beginnings of mpeg4-generic payloader (RFC 3640)

2006-02-09 14:20:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Resurected rtpdec to make rtspsrc happy again.
	  Original commit message from CVS:
	  * gst/rtsp/Makefile.am:
	  * gst/rtsp/gstrtpdec.c: (gst_rtpdec_get_type),
	  (gst_rtpdec_class_init), (gst_rtpdec_init), (gst_rtpdec_getcaps),
	  (gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp),
	  (gst_rtpdec_set_property), (gst_rtpdec_get_property),
	  (gst_rtpdec_change_state):
	  * gst/rtsp/gstrtpdec.h:
	  * gst/rtsp/gstrtsp.c: (plugin_init):
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
	  * gst/rtsp/rtspconnection.c: (read_body),
	  (rtsp_connection_receive):
	  * gst/rtsp/rtspmessage.c: (rtsp_message_dump):
	  Resurected rtpdec to make rtspsrc happy again.
	  Skip attributes from the session id.
	  Don't crash when dumping a message with an empty body.

2006-02-09 14:14:07 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpamrdepay.c: Added more meaningfull warnings when something goes wrong.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_chain):
	  Added more meaningfull warnings when something goes wrong.
	  Clear F bit on outgoing AMR packets.
	  * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init),
	  (gst_rtp_amr_pay_handle_buffer):
	  Added debugging category
	  Support payloading of multiple AMR frames.
	  * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_depay_data):
	  Added some debugging.

2006-02-09 11:25:42 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS

=== release 0.10.2 ===

2006-02-09 11:22:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  Releasing 0.10.2
	  Original commit message from CVS:
	  Releasing 0.10.2

2006-02-08 17:35:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-02-08 17:18:20 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Oops, jumping the gun with the ChangeLog entry
	  Original commit message from CVS:
	  Oops, jumping the gun with the ChangeLog entry

2006-02-08 17:16:46 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Bump core and plugins-base requirement to 0.10.2.2 for API additions (and 1 migration of gst_bin_find_u...
	  Original commit message from CVS:
	  * configure.ac:
	  Bump core and plugins-base requirement to 0.10.2.2
	  for API additions (and 1 migration of gst_bin_find_unconnected_pad)

2006-02-08 17:12:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/: Register musicbrainz tags.
	  Original commit message from CVS:
	  * ext/flac/gstflac.c: (plugin_init):
	  * ext/speex/gstspeex.c: (plugin_init):
	  Register musicbrainz tags.

2006-02-07 18:31:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/qtdemux/qtdemux.c:
	  remove unused var
	  Original commit message from CVS:
	  remove unused var

2006-02-07 18:01:17 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/qtdemux/qtdemux.c: use the correct variable to check if we can calculate the last chunk.  Looks like an obvious b...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
	  (qtdemux_parse_trak):
	  use the correct variable to check if we can calculate
	  the last chunk.  Looks like an obvious bug, and makes
	  the dump of offsets comparable to other tools

2006-02-07 17:54:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/qtdemux/qtdemux.c: clean up some debugging, using _OBJECT, moving recurring messages to LOG level
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
	  (qtdemux_parse_trak):
	  clean up some debugging, using _OBJECT, moving recurring
	  messages to LOG level

2006-02-07 16:23:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gconf/gconf.h: Remove declaration of function that no longer exists.
	  Original commit message from CVS:
	  * ext/gconf/gconf.h:
	  Remove declaration of function that no longer exists.

2006-02-07 13:39:08 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/shout2/gstshout2.c: Make shout2 work for non ogg streams
	  Original commit message from CVS:
	  2006-02-07  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/shout2/gstshout2.c: (gst_shout2send_render),
	  (gst_shout2send_setcaps), (gst_shout2send_change_state):
	  Make shout2 work for non ogg streams

2006-02-06 17:26:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/udp/gstmultiudpsink.*: Updated docs.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	  (gst_multiudpsink_render), (gst_multiudpsink_get_property),
	  (gst_multiudpsink_init_send), (gst_multiudpsink_add),
	  (gst_multiudpsink_remove), (gst_multiudpsink_clear),
	  (gst_multiudpsink_get_stats), (gst_multiudpsink_change_state):
	  * gst/udp/gstmultiudpsink.h:
	  Updated docs.
	  Added properties bytes-served, bytes_to_serve.
	  Post proper error messages,
	  Emit client added signal too.

2006-02-06 15:41:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.*: Some QT demux loving.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_query),
	  (gst_qtdemux_handle_src_event), (gst_qtdemux_loop_header),
	  (qtdemux_inflate), (qtdemux_parse), (qtdemux_parse_trak),
	  (qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num),
	  (qtdemux_tag_add_gnre), (gst_qtdemux_handle_esds),
	  (qtdemux_video_caps), (qtdemux_audio_caps):
	  * gst/qtdemux/qtdemux.h:
	  Some QT demux loving.
	  Handle seeking in a less broken way.
	  Fix AMR caps to match the AMR decoder.
	  Set first timestamp on AMR samples to 0 for now.
	  Remove some \n in DEBUG strings.
	  Use _scale_int for maximum precision.

2006-02-06 15:31:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* common:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/udp/gstmultiudpsink.c:
	  adding docs for multiudpsink
	  Original commit message from CVS:
	  adding docs for multiudpsink

2006-02-06 15:28:56 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/level/gstlevel.c: peak below decay is not necessarily an error, so don't ERROR log
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_transform_ip):
	  peak below decay is not necessarily an error, so don't ERROR log

2006-02-06 15:27:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  cvs versions
	  Original commit message from CVS:
	  cvs versions

2006-02-06 14:25:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/ebml-write.*: Make sure we send a newsegment event in BYTES format before sending buffers (#328531).
	  Original commit message from CVS:
	  * gst/matroska/ebml-write.c: (gst_ebml_write_reset),
	  (gst_ebml_write_flush_cache), (gst_ebml_write_element_push),
	  (gst_ebml_write_seek):
	  * gst/matroska/ebml-write.h:
	  Make sure we send a newsegment event in BYTES format
	  before sending buffers (#328531).

2006-02-06 12:18:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Pass unhandled queries upstream instead of just dropping them (#326446). Update query type arrays here and there.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_src_query),
	  (gst_dvdemux_sink_query):
	  * ext/flac/gstflacdec.c: (gst_flac_dec_src_query):
	  * ext/speex/gstspeexdec.c: (speex_get_query_types),
	  (speex_dec_src_query):
	  * ext/speex/gstspeexenc.c: (gst_speexenc_src_query),
	  (gst_speexenc_sink_query):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_get_src_query_types),
	  (gst_matroska_demux_handle_src_query):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_get_query_types),
	  (gst_wavparse_pad_query):
	  Pass unhandled queries upstream instead of just dropping
	  them (#326446). Update query type arrays here and there.

2006-02-06 11:57:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/matroskamux.c: Collectpads in core got changed and now also holds a reference to any pad that is...
	  Original commit message from CVS:
	  * tests/check/elements/matroskamux.c: (setup_src_pad):
	  Collectpads in core got changed and now also holds a
	  reference to any pad that is part of it. Fix refcount
	  checks in test case accordingly.

2006-02-06 11:41:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/apetag/gstapedemux.h: Fix include, for now GstTagDemux is in the apetag dir.
	  Original commit message from CVS:
	  * gst/apetag/gstapedemux.h:
	  Fix include, for now GstTagDemux is in the apetag dir.

2006-02-06 11:34:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Add cdio plugin to docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-cdio.xml:
	  Add cdio plugin to docs.
	  * ext/cdio/gstcdiocddasrc.c:
	  Add gtk-doc blurb.
	  * ext/cdio/gstcdio.c:
	  The plugin is called 'cdio' not 'cddio'.

2006-02-06 10:56:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add APE tag demuxer (#325649).
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-apetag.xml:
	  * gst/apetag/Makefile.am:
	  * gst/apetag/gstapedemux.c:
	  * gst/apetag/gstapedemux.h:
	  * gst/apetag/gsttagdemux.c:
	  * gst/apetag/gsttagdemux.h:
	  Add APE tag demuxer (#325649).

2006-02-05 22:22:56 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/gconf/: Ignore changing the GConf key to "". Ignore GConf key updates that don't actually change the string.
	  Original commit message from CVS:
	  * ext/gconf/gconf.c: (gst_gconf_get_default_audio_sink),
	  (gst_gconf_get_default_video_sink),
	  (gst_gconf_get_default_audio_src),
	  (gst_gconf_get_default_video_src):
	  * ext/gconf/gconf.h:
	  * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
	  (gst_gconf_audio_sink_init), (gst_gconf_audio_sink_dispose),
	  (do_toggle_element):
	  * ext/gconf/gstgconfaudiosink.h:
	  * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
	  (gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose),
	  (do_toggle_element):
	  * ext/gconf/gstgconfaudiosrc.h:
	  * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
	  (gst_gconf_video_sink_init), (gst_gconf_video_sink_dispose),
	  (do_toggle_element):
	  * ext/gconf/gstgconfvideosink.h:
	  * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
	  (gst_gconf_video_src_init), (gst_gconf_video_src_dispose),
	  (do_toggle_element):
	  * ext/gconf/gstgconfvideosrc.h:
	  Ignore changing the GConf key to "". Ignore GConf key updates
	  that don't actually change the string.
	  For now, ignore the GConf key when the state is > READY, as
	  it breaks streaming. Sometime it will be nice to bring the
	  new sink online even mid-stream, by sending NEWSEGMENT info
	  and possibly prerolling.
	  (Fixes #326736)

2006-02-05 20:43:49 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/goom/: Make goom reentrant by moving all important static variables into instance structures.
	  Original commit message from CVS:
	  * gst/goom/filters.c: (zoomFilterNew), (calculatePXandPY),
	  (setPixelRGB), (setPixelRGB_), (getPixelRGB), (getPixelRGB_),
	  (zoomFilterSetResolution), (zoomFilterDestroy),
	  (zoomFilterFastRGB), (pointFilter):
	  * gst/goom/filters.h:
	  * gst/goom/goom_core.c: (goom_init), (goom_set_resolution),
	  (goom_update), (goom_close):
	  * gst/goom/goom_core.h:
	  * gst/goom/goom_tools.h:
	  * gst/goom/graphic.c:
	  * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
	  (gst_goom_dispose), (gst_goom_src_setcaps), (gst_goom_chain):
	  * gst/goom/gstgoom.h:
	  * gst/goom/lines.c: (goom_lines):
	  * gst/goom/lines.h:
	  Make goom reentrant by moving all important static variables
	  into instance structures.
	  (Fixes #329181)

2006-02-04 15:41:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.*: Third attempt, use gst_pad_is_linked() this time.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_all_source_pads_unlinked),
	  (gst_avi_demux_process_next_entry):
	  * gst/avi/gstavidemux.h:
	  Third attempt, use gst_pad_is_linked() this time.

2006-02-04 13:30:12 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3v2frames.c: Adjust for data length indicators when parsing (Fixes #329810)
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	  (parse_split_strings):
	  Adjust for data length indicators when parsing (Fixes #329810)
	  Fix stupid bug parsing UTF-8 tag text.
	  Output tag strings with multiple fields as multiple tags, so the
	  app gets all the data.

2006-02-03 20:05:20 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* ext/flac/gstflacenc.c:
	  Fixed a bug add in last commit, where no event is send. Thanks Tim to show me.
	  Original commit message from CVS:
	  Fixed a bug add in last commit, where no event is send. Thanks Tim to show me.

2006-02-03 18:07:35 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* ext/flac/gstflacenc.c:
	* gst/matroska/ebml-read.c:
	  Just make it compile with --disable-gst-debug.
	  Original commit message from CVS:
	  Just make it compile with --disable-gst-debug.

2006-02-03 16:55:42 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  update spec file
	  Original commit message from CVS:
	  update spec file

2006-02-03 13:06:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3v2frames.c: Never output a tag with a null contents string.
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
	  (id3v2_tag_to_taglist), (id3v2_genre_string_to_taglist),
	  (id3v2_genre_fields_to_taglist):
	  Never output a tag with a null contents string.

2006-02-02 21:00:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Only pause if all pads are unlinked AND we've tried to send data on all of them at least once.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_all_source_pads_unlinked):
	  Only pause if all pads are unlinked AND we've tried to send data
	  on all of them at least once.

2006-02-02 12:29:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Make loop function/task pause itself when all source pads are unlinked.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_all_source_pads_unlinked),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_loop):
	  Make loop function/task pause itself when all source pads are
	  unlinked.

2006-02-02 10:47:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use new functions from core to render a bin from a string. Fixes build. Up requirements to core CVS.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/gconf/gconf.c: (gst_gconf_render_bin_from_key):
	  Use new functions from core to render a bin from a
	  string. Fixes build. Up requirements to core CVS.

2006-02-01 11:01:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/auparse/gstauparse.c: Don't push buffers into the adapter that we are going to push downstream again without fram...
	  Original commit message from CVS:
	  * gst/auparse/gstauparse.c: (gst_au_parse_chain):
	  Don't push buffers into the adapter that we are going to
	  push downstream again without framing anyway. Also, the
	  adaptor takes ownership of buffers put into it (fixes
	  auparse pushing invalid buffers for .au files with
	  ADPCM contents). Finally, set caps on all outgoing buffers.

2006-01-30 23:13:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/: Someone should kick my butt. Remove ID3v1 tags from the end of the file.
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
	  (gst_id3demux_read_id3v1), (gst_id3demux_sink_activate),
	  (gst_id3demux_send_tag_event):
	  * gst/id3demux/id3tags.c: (id3demux_read_id3v1_tag):
	  Someone should kick my butt. Remove ID3v1 tags from the end of the
	  file.
	  Improve error messages. Send the TAG message as soon as we complete
	  typefinding, instead of waiting until we send the first buffer.
	  Downstream tag event is still sent before the first buffer.

2006-01-29 20:07:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/gstwavpackdec.c: Add debug category, use boilerplate macros, fix handling of widths of 32 bits.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_setcaps),
	  (gst_wavpack_dec_base_init), (gst_wavpack_dec_dispose),
	  (gst_wavpack_dec_class_init), (gst_wavpack_dec_sink_event),
	  (gst_wavpack_dec_init), (gst_wavpack_dec_format_samples),
	  (gst_wavpack_dec_chain), (gst_wavpack_dec_plugin_init):
	  Add debug category, use boilerplate macros, fix handling
	  of widths of 32 bits.
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init),
	  (gst_wavpack_parse_dispose), (gst_wavpack_parse_class_init),
	  (gst_wavpack_parse_index_get_last_entry),
	  (gst_wavpack_parse_index_get_entry_from_sample),
	  (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
	  (gst_wavpack_parse_src_query),
	  (gst_wavpack_parse_scan_to_find_sample),
	  (gst_wavpack_parse_send_newsegment),
	  (gst_wavpack_parse_handle_seek_event),
	  (gst_wavpack_parse_src_event), (gst_wavpack_parse_init),
	  (gst_wavpack_parse_get_upstream_length),
	  (gst_wavpack_parse_pull_buffer),
	  (gst_wavpack_parse_create_src_pad), (gst_wavpack_parse_loop),
	  (gst_wavpack_parse_change_state),
	  (gst_wavepack_parse_sink_activate),
	  (gst_wavepack_parse_sink_activate_pull),
	  (gst_wavpack_parse_plugin_init):
	  * ext/wavpack/gstwavpackparse.h:
	  Rewrite a bit, mostly to fix flow logic and to make seeking work.
	  Fix buffer/event refcounting. Add some debug statements. Add
	  width of 32 to source pad template caps. Use boilerplate macros.

2006-01-27 12:17:56 +0000  Andy Wingo <wingo@pobox.com>

	  ext/dv/: Call dv_set_error_log (dv_decoder_t *, NULL); after dv_decoder_new to not have warings flooding stderr. this...
	  Original commit message from CVS:
	  2006-01-27  Jan Gerber  <j@bootlab.org>
	  Reviewed by: Andy Wingo <wingo@pobox.com>
	  * ext/dv/gstdvdec.c (gst_dvdec_change_state):
	  * ext/dv/gstdvdemux.c (gst_dvdemux_change_state):
	  Call dv_set_error_log (dv_decoder_t *, NULL); after dv_decoder_new
	  to not have warings flooding stderr. this is the suggested way
	  also used in dvgrab and kino. (#328336)

2006-01-27 01:43:07 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/oss/gstosssink.c: Free the device name string when finalised.
	  Original commit message from CVS:
	  * sys/oss/gstosssink.c: (gst_oss_sink_class_init),
	  (gst_oss_sink_init), (gst_oss_sink_finalise):
	  Free the device name string when finalised.

2006-01-26 16:23:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Fix wrong memcpy source pointer.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	  Fix wrong memcpy source pointer.

2006-01-25 22:05:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/id3demux/gstid3demux.c: Don't put function calls in g_return_if_fail() statements, or they'll be replaced with NO...
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_remove_srcpad):
	  Don't put function calls in g_return_if_fail() statements,
	  or they'll be replaced with NOOPs if someone compiles with
	  G_DISABLE_CHECKS defined.

2006-01-25 20:33:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  changelog surgery
	  Original commit message from CVS:
	  changelog surgery

2006-01-25 18:23:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3v2frames.c: Never trust ANY information encoded in a media file, especially when it's giving you size...
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
	  Never trust ANY information encoded in a media file, especially
	  when it's giving you sizes. (Fixes #328452)

2006-01-24 18:03:46 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* gst/rtp/gstrtpg711pay.c:
	  I'm too lazy to comment this
	  Original commit message from CVS:
	  Patch written by Kai Vehmanen <kai.vehmanen@nokia.com> applied. See bug #325148.

2006-01-24 11:58:53 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: More coherent framerate setting on caps.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
	  (gst_qtdemux_add_stream), (qtdemux_parse_trak):
	  More coherent framerate setting on caps.
	  If sample_size is available, use that for the samples' duration in
	  the index. This enables single frame streams to work (and I imagine
	  fixes some other cases).
	  Tested on testsuite, no regression.

2006-01-23 18:39:31 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/matroska/: Added recognition of Real Audio and Video streams in matroska demuxer.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps),
	  (gst_matroska_demux_audio_caps), (gst_matroska_demux_plugin_init):
	  * gst/matroska/matroska-ids.h:
	  Added recognition of Real Audio and Video streams in matroska demuxer.

2006-01-23 18:37:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.*: Contrary to what the const char in the lame API might suggest, lame expects us to keep the string...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_finalize), (gst_lame_class_init),
	  (gst_lame_init), (add_one_tag), (gst_lame_set_metadata):
	  * ext/lame/gstlame.h:
	  Contrary to what the const char in the lame API might suggest,
	  lame expects us to keep the strings we pass to id3tag_set_foo()
	  around; it doesn't free them either though, so we have to store
	  them somewhere and free them later when we can be sure lame
	  doesn't need them any longer.

2006-01-23 15:10:55 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Added codec recognition for: _ VP31 : video/x-vp3 _ AVDJ : image/jpeg _ dvcp, dvc  : video/x-d...
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
	  (qtdemux_video_caps), (qtdemux_audio_caps):
	  Added codec recognition for:
	  _ VP31 : video/x-vp3
	  _ AVDJ : image/jpeg
	  _ dvcp, dvc  : video/x-dv, systemstream=(boolean)false
	  _ 0x6d730017 : audio/x-adpcm, layout=(string)quicktime

2006-01-23 15:02:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/lame/gstlame.c: don't pass an uninitialised string pointer to lame if we don't know how to handle the tag type, a...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (add_one_tag):
	  Fix handling of GST_TAG_DATE (#311679), don't pass an
	  uninitialised string pointer to lame if we don't know
	  how to handle the tag type, and fix minor memory leak.

2006-01-23 14:32:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3v2frames.c: Remove errant break statement, and fix compilation with older GCC.
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
	  Remove errant break statement, and fix compilation with
	  older GCC.

2006-01-23 12:04:12 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Mention that my last commit fixes #328241
	  Original commit message from CVS:
	  Mention that my last commit fixes #328241

2006-01-23 11:06:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/sunaudio/: Export functions that are needed in other parts of the code, makes the mixer actually work; adjust mag...
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * sys/sunaudio/gstsunaudiomixerctrl.c:
	  * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_init):
	  Export functions that are needed in other parts of the code,
	  makes the mixer actually work; adjust magic minimum buffer-time
	  value from 3ms to 5ms to work around stuttering during mp3
	  playback (#327765).

2006-01-23 10:44:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-mux.c: Fix possible deadlock in matroska muxer (#327825).
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_best_pad),
	  (gst_matroska_mux_write_data), (gst_matroska_mux_collected):
	  Fix possible deadlock in matroska muxer (#327825).

2006-01-23 09:59:03 +0000  Jens Granseuer <jensgr@gmx.net>

	  C89 fixes: declare variables at the beginning of a block and
	  Original commit message from CVS:
	  * ext/libpng/gstpngenc.c: (gst_pngenc_chain):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_invert):
	  * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps):
	  * gst/rtsp/sdpmessage.h:
	  * gst/udp/gstdynudpsink.c: (gst_dynudpsink_render):
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_get_stats):
	  C89 fixes: declare variables at the beginning of a block and
	  make gcc-2.9x happy (#328264; patch by: Jens Granseuer
	  <jensgr at gmx dot net>).

2006-01-23 09:22:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/: Rewrite parsing of text tags to handle multiple NULL terminated strings. Parse numeric genre strings a...
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
	  * gst/id3demux/id3tags.h:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	  (parse_comment_frame), (parse_text_identification_frame),
	  (id3v2_tag_to_taglist), (id3v2_are_digits),
	  (id3v2_genre_string_to_taglist), (id3v2_genre_fields_to_taglist),
	  (parse_split_strings), (free_tag_strings):
	  Rewrite parsing of text tags to handle multiple NULL terminated
	  strings. Parse numeric genre strings and ID3v2 type
	  "(3)(6)Alternative" style genre strings.
	  Parse dates that are only YYYY or YYYY-mm format.

2006-01-21 11:43:53 +0000  Fabrizio <fabrizio.ge@tiscali.it>

	  gst/qtdemux/qtdemux.c: 'twos' and 'sowt' fourcc can be 16bit or 8bit audio.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
	  (qtdemux_audio_caps):
	  'twos' and 'sowt' fourcc can be 16bit or 8bit audio.
	  Fix 8bit case (#327133, based on patch by: Fabrizio
	  Gennari <fabrizio dot ge at tiscali dot it>).
	  Also, "G_LITTLE_ENDIAN" and "G_BIG_ENDIAN" are not
	  valid literals for endianness in caps strings,
	  only "LITTLE_ENDIAN" and "BIG_ENDIAN" are valid.

2006-01-20 15:06:28 +0000  Christoph Burghardt <hawkes@web.de>

	  gst/videobox/gstvideobox.c: Don't forget to initialize liboil, otherwise our oil functions
	  Original commit message from CVS:
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init):
	  Don't forget to initialize liboil, otherwise our oil functions
	  will crash (fixes #327871; patch by: Christoph Burghardt
	  <hawkes at web dot de>).

2006-01-19 21:46:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery (last entry may have been slightly misleading)
	  Original commit message from CVS:
	  ChangeLog surgery (last entry may have been slightly misleading)

2006-01-19 21:00:50 +0000  Brian Cameron <brian.cameron@sun.com>

	  configure.ac: just like in the core and gst-plugins-base. Fixes build on Solaris (fixes
	  Original commit message from CVS:
	  * configure.ac:
	  Use plain AS_LIBTOOL_TAGS instead of AS_LIBTOOL_TAGS([CXX]), just
	  like in the core and gst-plugins-base. Fixes build on Solaris (fixes
	  #326683; patch by: Brian Cameron <brian dot cameron at sun dot com>)

2006-01-19 00:10:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cdio/: Fix build for libcdio versions >= 76; give slightly lower rank than cdparanoia.
	  Original commit message from CVS:
	  * ext/cdio/gstcdio.c: (gst_cdio_add_cdtext_field), (plugin_init):
	  * ext/cdio/gstcdio.h:
	  * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_get_cdtext):
	  Fix build for libcdio versions >= 76; give slightly lower rank
	  than cdparanoia.

2006-01-18 19:30:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Port libcdio cdda source, formerly known as cddasrc, now known as cdiocddasrc (fixes #323327). Should also read CD-TE...
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/Makefile.am:
	  * ext/cdio/Makefile.am:
	  * ext/cdio/gstcdio.c:
	  * ext/cdio/gstcdio.h:
	  * ext/cdio/gstcdiocddasrc.c:
	  * ext/cdio/gstcdiocddasrc.h:
	  Port libcdio cdda source, formerly known as cddasrc, now known as
	  cdiocddasrc (fixes #323327). Should also read CD-TEXT if available,
	  but that's not tested (fixes #317658).

2006-01-18 19:08:08 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/wavparse/gstwavparse.c: Fix conversion from TIME to BYTES format (fixes #326864;
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_pad_convert):
	  Fix conversion from TIME to BYTES format (fixes #326864;
	  patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>)

2006-01-18 18:54:02 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* gst/qtdemux/qtdemux.c:
	  Ronald's patch applied. see bug #326318.
	  Original commit message from CVS:
	  Ronald's patch applied. see bug #326318.

2006-01-17 16:45:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.*: Fix seeking for quicktime files. Could still use some more love and sophistication.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
	  (gst_qtdemux_send_event), (gst_qtdemux_handle_src_event),
	  (gst_qtdemux_change_state), (gst_qtdemux_loop_header):
	  * gst/qtdemux/qtdemux.h:
	  Fix seeking for quicktime files. Could still use some more
	  love and sophistication.

2006-01-16 10:23:47 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  update with love
	  Original commit message from CVS:
	  update with love

2006-01-15 20:21:48 +0000  Sergey Scobich <sergey.scobich@gmail.com>

	  gst/id3demux/id3v2frames.c: Fix compilation of id3demux when zlib is not present.
	  Original commit message from CVS:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
	  Fix compilation of id3demux when zlib is not present.
	  (Fixes #326602; patch by: Sergey Scobich)

2006-01-15 14:12:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/esd/Makefile.am: otherwise build will fail for folks with libesd in a non-standard prefix (#327009).
	  Original commit message from CVS:
	  * ext/esd/Makefile.am:
	  Add $(ESD_CFLAGS), otherwise build will fail for folks
	  with libesd in a non-standard prefix (#327009).

2006-01-13 19:29:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	  back to head
	  Original commit message from CVS:
	  back to head

2006-01-13 19:25:40 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/upload.mak:
	  releasing 0.10.1
	  Original commit message from CVS:
	  releasing 0.10.1

2006-01-13 18:37:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/jpeg/gstsmokeenc.c: fix memleak.  Fixes #326618
	  Original commit message from CVS:
	  patch by: Wim Taymans
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
	  fix memleak.  Fixes #326618

2006-01-13 18:35:00 +0000  Mike Smith <msmith@xiph.org>

	  gst/level/gstlevel.c: Fix memleak.  Fixes #326612
	  Original commit message from CVS:
	  2006-01-13  Thomas Vander Stichele  <thomas at apestaart dot org>
	  patch by: Mike Smith
	  * gst/level/gstlevel.c: (gst_level_message_new),
	  (gst_level_message_append_channel):
	  Fix memleak.  Fixes #326612

2006-01-11 11:39:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: prereleasing
	  Original commit message from CVS:
	  * configure.ac:
	  prereleasing
	  * po/af.po:
	  * po/az.po:
	  * po/cs.po:
	  * po/en_GB.po:
	  * po/hu.po:
	  * po/it.po:
	  * po/nb.po:
	  * po/nl.po:
	  * po/or.po:
	  * po/sq.po:
	  * po/sr.po:
	  * po/sv.po:
	  * po/uk.po:
	  * po/vi.po:
	  update translations

2006-01-11 11:04:03 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Add support for Indeo3 video in Quicktime files.
	  Original commit message from CVS:
	  reviewed by: Edward Hervey  <edward@fluendo.com>
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add support for Indeo3 video in Quicktime files.
	  Closes #326524

2006-01-10 12:38:59 +0000  Michael Smith <msmith@xiph.org>

	  gst/level/gstlevel.c: Don't leak filter arrays.
	  Original commit message from CVS:
	  * gst/level/gstlevel.c: (gst_level_class_init),
	  (gst_level_dispose):
	  Don't leak filter arrays.

2006-01-09 17:04:52 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/upload.mak:
	* gst-plugins-good.spec.in:
	* sys/Makefile.am:
	* sys/sunaudio/Makefile.am:
	* sys/sunaudio/gstsunaudio.c:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiomixer.h:
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixertrack.c:
	* sys/sunaudio/gstsunaudiomixertrack.h:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosink.h:
	  add Sun Audio plugin. Verified that nothing breaks and that make check works.
	  Original commit message from CVS:
	  add Sun Audio plugin. Verified that nothing breaks and that make check works.
	  Don't think the docs gets properly built yet, but I don't understand exactly how to enable that.

2006-01-07 20:01:09 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-plugins-good/gst/udp/: Allow udpsrc and dynudpsink to take a sockfd as a parameter. For udpsrc, overrides the por...
	  Original commit message from CVS:
	  2005-01-07  Philippe Khalaf  <philippe.kalaf@collabora.co.uk>
	  * gst-plugins-good/gst/udp/gstdynudpsink.c:
	  * gst-plugins-good/gst/udp/gstudpsrc.c:
	  Allow udpsrc and dynudpsink to take a sockfd as a parameter. For udpsrc,
	  overrides the port or multicast parameters. Fixes bugs #323021.

2006-01-06 16:28:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gconf/: Add new gconfaudiosrc and gconfvideosrc elements (needed for gnome-sound-recorder).
	  Original commit message from CVS:
	  * ext/gconf/Makefile.am:
	  * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
	  (gst_gconf_audio_src_class_init), (gst_gconf_audio_src_reset),
	  (gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose),
	  (do_toggle_element), (cb_toggle_element),
	  (gst_gconf_audio_src_change_state):
	  * ext/gconf/gstgconfaudiosrc.h:
	  * ext/gconf/gstgconfelements.c: (plugin_init):
	  * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
	  (gst_gconf_video_src_class_init), (gst_gconf_video_src_reset),
	  (gst_gconf_video_src_init), (gst_gconf_video_src_dispose),
	  (do_toggle_element), (cb_toggle_element),
	  (gst_gconf_video_src_change_state):
	  * ext/gconf/gstgconfvideosrc.h:
	  Add new gconfaudiosrc and gconfvideosrc elements
	  (needed for gnome-sound-recorder).

2006-01-06 11:46:53 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/id3demux/gstid3demux.c: Add gst_element_no_more_pads() for proper decodebin behaviour.
	  Original commit message from CVS:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
	  Add gst_element_no_more_pads() for proper decodebin behaviour.
	  * gst/id3demux/id3v2frames.c: (parse_comment_frame),
	  (parse_text_identification_frame), (parse_split_strings):
	  Failure to decode some tags is not a GST_ERROR() but a
	  GST_WARNING()
	  When iterating over a chunk of text, check that we haven't gone too
	  far.

2006-01-05 23:17:44 +0000  Sébastien Moutte <sebastien@moutte.net>

	* sys/directdraw/gstdirectdrawplugin.c:
	* sys/directdraw/gstdirectdrawsink.c:
	* sys/directdraw/gstdirectdrawsink.h:
	* sys/directsound/gstdirectsoundplugin.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	* win32/vs6/libgstdirectdraw.dsp:
	* win32/vs6/libgstdirectsound.dsp:
	  added sys/directdraw added sys/directsound added win32/vs6/gst_plugins_bad.dsw added win32/vs6/libgstdirectsound.dsp ...
	  Original commit message from CVS:
	  2006-01-05  Sebastien Moutte  <sebastien@moutte.net>
	  * added sys/directdraw
	  * added sys/directsound
	  * added win32/vs6/gst_plugins_bad.dsw
	  * added win32/vs6/libgstdirectsound.dsp
	  * added win32/vs6/libgstdirectdraw.dsp
	  * added win32/common/config.h

2006-01-05 17:03:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/videobox/gstvideobox.c: call oil_init() when using liboil
	  Original commit message from CVS:
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	  (plugin_init):
	  call oil_init() when using liboil

2006-01-04 17:28:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/jpeg/: Fix leaks.
	  Original commit message from CVS:
	  * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
	  Fix leaks.

2006-01-02 19:38:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.c: Don't g_assert() where we should just return FALSE; remove unnecessary g_assert(); initialize ...
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * ext/flac/gstflacdec.c: (gst_flac_dec_write),
	  (gst_flac_dec_convert_src), (gst_flac_dec_src_query),
	  (gst_flac_dec_change_state):
	  Don't g_assert() where we should just return FALSE; remove
	  unnecessary g_assert(); initialize some fields properly in
	  state change function (fixes #325504). Also, use
	  GST_DEBUG_OBJECT in two more places.

2005-12-30 15:51:05 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: also remove smoothwave's Makefile.am
	  Original commit message from CVS:
	  * configure.ac:
	  also remove smoothwave's Makefile.am
	  * docs/plugins/Makefile.am:
	  fix plugin docs

2005-12-30 15:39:17 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/smoothwave/.gitignore:
	* gst/smoothwave/Makefile.am:
	* gst/smoothwave/README:
	* gst/smoothwave/demo-osssrc.c:
	* gst/smoothwave/gstsmoothwave.c:
	* gst/smoothwave/gstsmoothwave.h:
	  remove old plugin that went bad
	  Original commit message from CVS:
	  remove old plugin that went bad

2005-12-30 15:34:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/examples/Makefile.am: added missing Makefile.am
	  Original commit message from CVS:
	  * tests/examples/Makefile.am:
	  added missing Makefile.am

2005-12-30 15:28:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  moved level-example to tests/examples/level-example
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/level/Makefile.am:
	  * gst/level/level-example.c:
	  * tests/Makefile.am:
	  * tests/examples/level/Makefile.am:
	  * tests/examples/level/level-example.c: (message_handler), (main):
	  moved level-example to tests/examples/level-example
	  * tests/old/examples/level/demo.c: (main):
	  * tests/old/examples/level/plot.c: (main):
	  some initial fixes

2005-12-29 16:36:19 +0000  Michael Smith <msmith@xiph.org>

	  gst/udp/gstmultiudpsink.*: Track packets sent per client in addition to bytes sent; provide this info through get-sta...
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render),
	  (gst_multiudpsink_remove), (gst_multiudpsink_get_stats):
	  * gst/udp/gstmultiudpsink.h:
	  Track packets sent per client in addition to bytes sent; provide
	  this info through get-stats signal

2005-12-29 11:26:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/auparse/gstauparse.c: Can't use gst_object_unref() on a GstAdapter (#325191).
	  Original commit message from CVS:
	  * gst/auparse/gstauparse.c: (gst_au_parse_dispose):
	  Can't use gst_object_unref() on a GstAdapter (#325191).

2005-12-28 18:55:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/id3demux/id3tags.c: If a broken tag has 0 bytes payload, at least still skip the 10 byte header
	  Original commit message from CVS:
	  * gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
	  If a broken tag has 0 bytes payload, at least still skip
	  the 10 byte header

2005-12-22 15:00:41 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-plugins-good/gst/rtp/: Making these depayloaders (H263+ and mpeg4 video) inherit from
	  Original commit message from CVS:
	  2005-12-22  Philippe Khalaf  <burger@speedy.org>
	  * gst-plugins-good/gst/rtp/gstrtph263pdepay.h:
	  * gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
	  * gst-plugins-good/gst/rtp/gstrtpmp4vdepay.h:
	  * gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
	  Making these depayloaders (H263+ and mpeg4 video) inherit from
	  RtpBaseDepayloaderClass. Fixes bugs #323922 and #323908.

2005-12-21 17:15:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  docs/plugins/gst-plugins-good-plugins.*: Regenerate the plugin hiearchy.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  Regenerate the plugin hiearchy.

2005-12-21 15:24:59 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Add documentation for id3demux.
	  Original commit message from CVS:
	  2005-12-21  Jan Schmidt  <thaytan@mad.scientist.com>
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  * gst/id3demux/gstid3demux.c: (gst_id3demux_get_type),
	  (gst_id3demux_base_init), (gst_id3demux_class_init),
	  (gst_id3demux_chain):
	  * gst/id3demux/gstid3demux.h:
	  Add documentation for id3demux.
	  Don't fail if the first buffer is not at offset 0, just
	  attempt to typefind and do pass through
	  Rename the gst_type function from gst_gst_id3demux..

2005-12-20 12:44:25 +0000  Michael Smith <msmith@xiph.org>

	  gst/udp/gstmultiudpsink.*: Collect statistics; return them from get_stats.
	  Original commit message from CVS:
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render),
	  (gst_multiudpsink_add), (gst_multiudpsink_remove),
	  (gst_multiudpsink_get_stats):
	  * gst/udp/gstmultiudpsink.h:
	  Collect statistics; return them from get_stats.

2005-12-19 15:43:30 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: Stupid signedness issue...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
	  Stupid signedness issue...

2005-12-19 15:19:44 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/swfdec/gstswfdec.c: Add debugging category and return GstFlowReturn in the right places
	  Original commit message from CVS:
	  * ext/swfdec/gstswfdec.c: (gst_swfdec_class_init),
	  (gst_swfdec_chain), (gst_swfdec_render):
	  Add debugging category and return GstFlowReturn in the right places
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link):
	  Get something from the peer pad once we've checked if there is a peer pad.
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
	  (qtdemux_tree_get_child_by_type), (qtdemux_parse_trak),
	  (qtdemux_video_caps):
	  Couple of fixes

2005-12-19 15:06:27 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: Construct index for indexless files.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	  (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_parse_odml), (gst_avi_demux_peek_tag),
	  (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
	  (gst_avi_demux_stream_header), (gst_avi_demux_loop):
	  Construct index for indexless files.
	  Make sure pad/buffers are correctly reset to NULL once we don't need
	  them anymore, else we get lovely segfaults/assertions.
	  * gst/wavparse/gstwavparse.c:
	  Yes, you can have 96KHz audio and wma in wav :(

2005-12-18 15:14:44 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Check for optional dependency on zlib for id3demux
	  Original commit message from CVS:
	  * configure.ac:
	  Check for optional dependency on zlib for id3demux
	  * gst/id3demux/Makefile.am:
	  * gst/id3demux/gstid3demux.c: (gst_gst_id3demux_get_type),
	  (gst_id3demux_base_init), (gst_id3demux_class_init),
	  (gst_id3demux_reset), (gst_id3demux_init), (gst_id3demux_dispose),
	  (gst_id3demux_add_srcpad), (gst_id3demux_remove_srcpad),
	  (gst_id3demux_trim_buffer), (gst_id3demux_chain),
	  (gst_id3demux_set_property), (gst_id3demux_get_property),
	  (id3demux_get_upstream_size), (gst_id3demux_srcpad_event),
	  (gst_id3demux_read_id3v1), (gst_id3demux_read_id3v2),
	  (gst_id3demux_sink_activate), (gst_id3demux_src_activate_pull),
	  (gst_id3demux_src_checkgetrange), (gst_id3demux_read_range),
	  (gst_id3demux_src_getrange), (gst_id3demux_change_state),
	  (gst_id3demux_pad_query), (gst_id3demux_get_query_types),
	  (simple_find_peek), (simple_find_suggest),
	  (gst_id3demux_do_typefind), (gst_id3demux_send_tag_event),
	  (plugin_init):
	  * gst/id3demux/gstid3demux.h:
	  * gst/id3demux/id3tags.c: (read_synch_uint),
	  (id3demux_read_id3v1_tag), (id3demux_read_id3v2_tag),
	  (id3demux_id3v2_frame_hdr_size), (convert_fid_to_v240),
	  (id3demux_id3v2_frames_to_tag_list):
	  * gst/id3demux/id3tags.h:
	  * gst/id3demux/id3v2.4.0-frames.txt:
	  * gst/id3demux/id3v2.4.0-structure.txt:
	  * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	  (parse_comment_frame), (parse_text_identification_frame),
	  (id3v2_tag_to_taglist), (parse_split_strings):
	  All new LGPL id3 demuxer. Can use zlib for compressed frames,
	  otherwise it discards them. Works on my test files.
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_loop):
	  Don't send EOS to a non-existing srcpad
	  The debug category can be static

2005-12-17 17:48:38 +0000  Julien Moutte <julien@moutte.net>

	  docs/plugins/: Updates.
	  Original commit message from CVS:
	  2005-12-17  Julien MOUTTE  <julien@moutte.net>
	  * docs/plugins/gst-plugins-bad-plugins-decl.txt:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-undocumented.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-dfbvideosink.xml:
	  * docs/plugins/inspect/plugin-qtdemux.xml:
	  * docs/plugins/inspect/plugin-sdlvideosink.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-tta.xml: Updates.
	  * ext/directfb/dfbvideosink.c:
	  (gst_dfbvideosink_surface_create),
	  (gst_dfbvideosink_event_thread), (gst_dfbvideosink_enum_vmodes),
	  (gst_dfbvideosink_enum_devices), (gst_dfbvideosink_setup),
	  (gst_dfbvideosink_cleanup),
	  (gst_dfbvideosink_can_blit_from_format),
	  (gst_dfbvideosink_get_best_vmode), (gst_dfbvideosink_getcaps),
	  (gst_dfbvideosink_setcaps), (gst_dfbvideosink_show_frame),
	  (gst_dfbvideosink_buffer_alloc), (gst_dfbsurface_finalize),
	  (gst_dfbvideosink_interface_supported),
	  (gst_dfbvideosink_navigation_send_event),
	  (gst_dfbvideosink_update_colorbalance),
	  (gst_dfbvideosink_colorbalance_list_channels),
	  (gst_dfbvideosink_colorbalance_set_value),
	  (gst_dfbvideosink_colorbalance_get_value),
	  (gst_dfbvideosink_colorbalance_init),
	  (gst_dfbvideosink_set_property),
	  (gst_dfbvideosink_get_property),
	  (gst_dfbvideosink_init), (gst_dfbvideosink_class_init):
	  * ext/directfb/dfbvideosink.h: Implement vertical sync and
	  color balance interface.

2005-12-16 21:57:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  change some char* into char[]
	  Original commit message from CVS:
	  * ext/esd/esdmon.c: (gst_esdmon_open_audio):
	  * ext/esd/esdsink.c: (gst_esdsink_prepare):
	  * gst/multipart/multipartdemux.c:
	  change some char* into char[]

2005-12-16 19:32:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.*: Use GstSegment to implement more seeking features.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
	  (gst_wavparse_other), (gst_wavparse_perform_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data),
	  (gst_wavparse_loop), (gst_wavparse_pad_convert),
	  (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull):
	  * gst/wavparse/gstwavparse.h:
	  Use GstSegment to implement more seeking features.

2005-12-16 12:25:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/wavpack/gstwavpackdec.c: Oops, remove trailing comma from caps string.
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c:
	  Oops, remove trailing comma from caps string.

2005-12-16 10:12:49 +0000  Benjamin Pineau <ben.pineau@gmail.com>

	  gst/rtsp/rtspconnection.c: Add <netinet/in.h> include and move <arpa/inet.h> include to make things work on OpenBSD a...
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c:
	  Add <netinet/in.h> include and move <arpa/inet.h> include
	  to make things work on OpenBSD as well (fixes #323717;
	  patch by: Benjamin Pineau)

2005-12-16 09:59:21 +0000  gcocatre@gmail.com <gcocatre@gmail.com>

	  ext/wavpack/: Wavpack supports samplerates from 6-192kHz, fix pad template remove buffer-frames from caps, they are g...
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link):
	  * ext/wavpack/gstwavpackparse.c:
	  Wavpack supports samplerates from 6-192kHz, fix pad template
	  caps (fixes #322973; patch by: gcocatre@gmail.com). Also
	  remove buffer-frames from caps, they are gone in 0.10.

2005-12-14 20:05:45 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	  Set clock rate to be fixed in 8000. It fixes bug #324012.
	  Original commit message from CVS:
	  Set clock rate to be fixed in 8000. It fixes bug #324012.

2005-12-14 18:07:16 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-plugins-good/gst/rtp/: Fixed payload range in payloder caps. Removed payload range completly from depayloaders as...
	  Original commit message from CVS:
	  2005-12-14  Philippe Khalaf  <burger@speedy.org>
	  * gst-plugins-good/gst/rtp/gstasteriskh263.c:
	  * gst-plugins-good/gst/rtp/gstrtpamrdepay.c:
	  * gst-plugins-good/gst/rtp/gstrtpamrpay.c:
	  * gst-plugins-good/gst/rtp/gstrtpg711depay.c:
	  * gst-plugins-good/gst/rtp/gstrtpg711depay.c:
	  * gst-plugins-good/gst/rtp/gstrtpgsmdepay.c:
	  * gst-plugins-good/gst/rtp/gstrtph263pay.c:
	  * gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
	  * gst-plugins-good/gst/rtp/gstrtph263ppay.c:
	  * gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
	  * gst-plugins-good/gst/rtp/gstrtpmp4vpay.c:
	  * gst-plugins-good/gst/rtp/gstrtpmpadepay.c:
	  * gst-plugins-good/gst/rtp/gstrtpmpapay.c:
	  * gst-plugins-good/gst/rtp/README:
	  Fixed payload range in payloder caps. Removed payload range completly from
	  depayloaders as they don't require payload type in their caps. In effect,
	  there isn't any specific payload type for any given codec, only suggestions.
	  Fixes bug #324011.

2005-12-13 21:58:42 +0000  Julien Moutte <julien@moutte.net>

	  gst/videomixer/videomixer.c: Code cleanup and re-enabling queued time validity check for correct EOS handling.
	  Original commit message from CVS:
	  2005-12-13  Julien MOUTTE  <julien@moutte.net>
	  * gst/videomixer/videomixer.c: (gst_videomixer_init),
	  (gst_videomixer_fill_queues), (gst_videomixer_blend_buffers),
	  (gst_videomixer_collected): Code cleanup and re-enabling
	  queued time validity check for correct EOS handling.

2005-12-13 17:18:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/gstossmixerelement.c: Add 'device-name' property and fix state change function.
	  Original commit message from CVS:
	  * sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
	  (gst_oss_mixer_element_get_property),
	  (gst_oss_mixer_element_change_state):
	  Add 'device-name' property and fix state change function.

2005-12-13 10:45:04 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/flx/gstflxdec.c: If the speed of the file is null in the header, set the frame_time to the default setting of GST...
	  Original commit message from CVS:
	  * gst/flx/gstflxdec.c: (gst_flxdec_chain):
	  If the speed of the file is null in the header, set the frame_time to the default
	  setting of GST_SECOND / 70. Which is the default frame_delay for .fli files as
	  stated in this document : http://www.compuphase.com/flic.htm
	  Would be nice to have the time conversion done properly too
	  (duration = flxh->frames * flxdec->frame_time)

2005-12-12 22:29:34 +0000  Julien Moutte <julien@moutte.net>

	  Adding documentation for videomixer on my way with a funny sample pipeline.
	  Original commit message from CVS:
	  2005-12-12  Julien MOUTTE  <julien@moutte.net>
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * gst/videomixer/videomixer.c:
	  (gst_videomixer_pad_sink_setcaps),
	  (gst_videomixer_getcaps), (gst_videomixer_fill_queues),
	  (gst_videomixer_update_queues), (gst_videomixer_collected):
	  Adding
	  documentation for videomixer on my way with a funny sample
	  pipeline.

2005-12-12 21:43:00 +0000  Julien Moutte <julien@moutte.net>

	  gst/videomixer/videomixer.c: Fix caps negotiation. (#323896)
	  Original commit message from CVS:
	  2005-12-12  Julien MOUTTE  <julien@moutte.net>
	  * gst/videomixer/videomixer.c:
	  (gst_videomixer_pad_sink_setcaps),
	  (gst_videomixer_getcaps), (gst_videomixer_fill_queues),
	  (gst_videomixer_update_queues), (gst_videomixer_collected):
	  Fix caps negotiation. (#323896)

2005-12-12 18:14:58 +0000  Arwed v. Merkatz <v.merkatz@gmx.net>

	* ChangeLog:
	* gst/matroska/matroska-demux.c:
	  Set correct timestamps on audio laces, fixes playback of mp3 from matroska.
	  Original commit message from CVS:
	  Set correct timestamps on audio laces, fixes playback of mp3 from matroska.

2005-12-12 10:40:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/: GstObjects must be unref'ed with gst_object_unref() instead of g_object_unref(), otherwise things break for GLi...
	  Original commit message from CVS:
	  * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_loop):
	  * ext/libmms/gstmms.c: (gst_mms_src_query), (gst_mms_create):
	  * ext/musepack/gstmusepackdec.c: (gst_musepackdec_src_query),
	  (gst_musepackdec_loop):
	  * ext/swfdec/gstswfdec.c: (gst_swfdec_video_link),
	  (gst_swfdec_src_query):
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query):
	  GstObjects must be unref'ed with gst_object_unref() instead of
	  g_object_unref(), otherwise things break for GLib-2.6 users.

2005-12-12 10:30:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/auparse/gstauparse.*: Use gst_object_unref() for GstObjects instead of g_object_unref() and fix a mem leak in a d...
	  Original commit message from CVS:
	  * gst/auparse/gstauparse.c: (gst_au_parse_base_init),
	  (gst_au_parse_class_init), (gst_au_parse_init),
	  (gst_au_parse_dispose), (gst_au_parse_chain),
	  (gst_au_parse_change_state), (plugin_init):
	  * gst/auparse/gstauparse.h:
	  Use gst_object_unref() for GstObjects instead of
	  g_object_unref() and fix a mem leak in a debug
	  statement; while we're at it, also borgify, use
	  boilerplate macros and clean up a little bit.

2005-12-11 20:27:06 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/debug/efence.c: Added pull mode.
	  Original commit message from CVS:
	  * gst/debug/efence.c: (gst_efence_init), (gst_efence_getrange),
	  (gst_efence_checkgetrange), (gst_efence_activate_src_pull):
	  Added pull mode.

2005-12-11 19:25:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Use audiotestsrc instead of sinesrc (#323798).
	  Original commit message from CVS:
	  * gst/goom/gstgoom.c:
	  * gst/level/level-example.c: (main):
	  * gst/smoothwave/demo-osssrc.c: (main):
	  Use audiotestsrc instead of sinesrc (#323798).

2005-12-11 17:50:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/oss/gstosssink.c: more debug-func-ptr usage
	  Original commit message from CVS:
	  * sys/oss/gstosssink.c: (gst_oss_sink_class_init):
	  more debug-func-ptr usage

2005-12-11 16:43:42 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ChangeLog:
	* gst/flx/flx_color.c:
	* gst/flx/flx_color.h:
	* gst/flx/flx_fmt.h:
	* gst/flx/gstflxdec.c:
	* gst/flx/gstflxdec.h:
	  Now flxdec works on big-endian machines as well.
	  Original commit message from CVS:
	  Now flxdec works on big-endian machines as well.

2005-12-11 16:14:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/debug/efence.c: Make sure GST_BUFFER_DATA is set on fenced copied buffers; fix
	  Original commit message from CVS:
	  * gst/debug/efence.c: (gst_efence_init), (gst_efence_chain),
	  (gst_fenced_buffer_copy):
	  Make sure GST_BUFFER_DATA is set on fenced copied buffers; fix
	  GST_DEBUG crasher where GST_TIME_FORMAT was not used in
	  conjunction with GST_TIME_ARGS. Also, don't leak pad templates
	  and use GST_DEBUG_FUNCPTR for pad functions.

2005-12-10 20:26:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.*: Rewrite flacdec a bit, so that even seeking might work now. Most importantly, don't act upon a...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (gst_flac_dec_base_init),
	  (gst_flac_dec_class_init), (gst_flac_dec_init),
	  (gst_flac_dec_metadata_callback), (gst_flac_dec_error_callback),
	  (gst_flac_dec_eof), (gst_flac_dec_write), (gst_flac_dec_loop),
	  (gst_flac_dec_convert_src), (gst_flac_dec_get_src_query_types),
	  (gst_flac_dec_src_query), (gst_flac_dec_send_newsegment),
	  (gst_flac_dec_handle_seek_event), (gst_flac_dec_src_event),
	  (gst_flac_dec_change_state):
	  * ext/flac/gstflacdec.h:
	  Rewrite flacdec a bit, so that even seeking might work now. Most
	  importantly, don't act upon any flow return values we get, just tell
	  the decoder everything's dandy and act on the flow return values
	  later on in the loop function. We don't want to mess up the internal
	  decoder state for non-fatal things like flushing pads etc. Other
	  than that, use GstSegment (segment seeks don't work yet though, but
	  should be easy to add), use boilerplate macros, drop the superfluous
	  'flacdec:' from debug messages, use gst_util_uint64_scale_int, and
	  lots of other things.

2005-12-10 14:57:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Update comment in OSS includes check.
	  Original commit message from CVS:
	  * configure.ac:
	  Update comment in OSS includes check.
	  * sys/oss/gstossdmabuffer.c:
	  * sys/oss/gstosshelper.c:
	  * sys/oss/gstossmixer.c:
	  * sys/oss/gstossmixertrack.c:
	  * sys/oss/gstosssink.c:
	  * sys/oss/gstosssrc.c:
	  * sys/oss/oss_probe.c:
	  Don't assume the OSS soundcard.h include is always in
	  the sys/ directory. Instead, use the existing defines
	  from config.h to include the right file. Fixes
	  compilation on OpenBSD 3.8 (#323718).

2005-12-09 19:51:03 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* ext/flac/gstflac.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  borgify and fix up documentation
	  Original commit message from CVS:
	  borgify and fix up documentation

2005-12-09 15:30:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/faad/gstfaad.c: Assume that an unknown channel mapping with 2 channels is stereo and play it that way instead of ...
	  Original commit message from CVS:
	  * ext/faad/gstfaad.c: (gst_faad_chanpos_to_gst),
	  (gst_faad_update_caps):
	  Assume that an unknown channel mapping with 2 channels
	  is stereo and play it that way instead of erroring.
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
	  (gst_qtdemux_add_stream), (qtdemux_parse_trak):
	  Handle e.g. jpeg streams with 0 duration frames as having 0 framerate.
	  Debug fixes. Some 64 bit variable fixes

2005-12-09 11:12:48 +0000  Michael Smith <msmith@xiph.org>

	  ext/flac/gstflacdec.c: Accept a wider range of flac files, more closely matching flac sp
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (raw_caps_factory), (gst_flacdec_write):
	  Accept a wider range of flac files, more closely matching flac sp

2005-12-08 16:27:12 +0000  Julien Moutte <julien@moutte.net>

	  docs/plugins/Makefile.am: Add multipart elements.
	  Original commit message from CVS:
	  2005-12-08  Julien MOUTTE  <julien@moutte.net>
	  * docs/plugins/Makefile.am: Add multipart elements.
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt: Fix flac.
	  * docs/plugins/gst-plugins-good-plugins.hierarchy:
	  * gst/multipart/multipartdemux.c:
	  * gst/multipart/multipartmux.c: Add docs.

2005-12-07 11:46:15 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/qtdemux/qtdemux.c: Memleak fixes.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
	  (gst_qtdemux_add_stream):
	  Memleak fixes.
	  Send out EOS for valid reasons (couldn't pull_range() from upstream
	  for example).

2005-12-07 11:40:46 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: Memleak and crasher fixes.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event),
	  (gst_avi_demux_parse_stream), (gst_avi_demux_stream_header),
	  (gst_avi_demux_invert):
	  Memleak and crasher fixes.
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	  (gst_wavparse_create_sourcepad), (gst_wavparse_stream_headers):
	  Memleak fixes

2005-12-06 19:55:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/equalizer/gstiirequalizer.c:
	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	* sys/v4l2/gstv4l2colorbalance.h:
	* sys/v4l2/gstv4l2element.h:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/gstv4l2tuner.h:
	* sys/v4l2/gstv4l2xoverlay.h:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2_calls.h:
	* sys/v4l2/v4l2src_calls.c:
	* sys/v4l2/v4l2src_calls.h:
	  expand tabs
	  Original commit message from CVS:
	  expand tabs

2005-12-06 19:48:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.h:
	  expand tabs
	  Original commit message from CVS:
	  expand tabs

2005-12-06 19:44:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* ext/aalib/gstaasink.h:
	* ext/cairo/gsttextoverlay.h:
	* ext/dv/gstdvdec.h:
	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	* ext/esd/esdsink.h:
	* ext/flac/flac_compat.h:
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.h:
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.h:
	* ext/gconf/gstgconfvideosink.h:
	* ext/gdk_pixbuf/gstgdkanimation.h:
	* ext/jpeg/gstjpegdec.h:
	* ext/jpeg/smokecodec.h:
	* ext/jpeg/smokeformat.h:
	* ext/ladspa/gstsignalprocessor.h:
	* ext/ladspa/search.c:
	* ext/ladspa/utils.h:
	* ext/libmng/gstmngdec.h:
	* ext/libmng/gstmngenc.c:
	* ext/libmng/gstmngenc.h:
	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	* ext/shout2/gstshout2.h:
	* ext/speex/gstspeexdec.h:
	* ext/speex/gstspeexenc.c:
	* ext/speex/gstspeexenc.h:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.h:
	* gst/autodetect/gstautovideosink.h:
	* gst/avi/gstavidemux.h:
	* gst/cutter/gstcutter.h:
	* gst/debug/tests.c:
	* gst/debug/tests.h:
	* gst/effectv/gstwarp.c:
	* gst/flx/flx_fmt.h:
	* gst/flx/gstflxdec.h:
	* gst/goom/filters.c:
	* gst/goom/filters.h:
	* gst/goom/goom_tools.h:
	* gst/law/alaw-encode.c:
	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.h:
	* gst/monoscope/convolve.c:
	* gst/monoscope/convolve.h:
	* gst/multipart/multipartmux.c:
	* gst/oldcore/gstaggregator.c:
	* gst/oldcore/gstaggregator.h:
	* gst/oldcore/gstmd5sink.c:
	* gst/oldcore/gstmd5sink.h:
	* gst/oldcore/gstmultifilesrc.c:
	* gst/oldcore/gstmultifilesrc.h:
	* gst/oldcore/gstpipefilter.h:
	* gst/oldcore/gstshaper.h:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpdepay.h:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.h:
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/rtsptransport.h:
	* gst/rtsp/rtspurl.c:
	* gst/rtsp/rtspurl.h:
	* gst/rtsp/sdpmessage.c:
	* gst/rtsp/sdpmessage.h:
	* gst/smpte/barboxwipes.c:
	* gst/smpte/gstmask.h:
	* gst/smpte/gstsmpte.h:
	* gst/smpte/paint.c:
	* gst/smpte/paint.h:
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsink.h:
	* gst/udp/gstudpsrc.c:
	* gst/videomixer/videomixer.c:
	* gst/wavenc/riff.h:
	* gst/wavparse/gstwavparse.h:
	* sys/oss/gstossdmabuffer.h:
	* sys/oss/gstossmixer.h:
	* sys/oss/gstossmixerelement.h:
	* sys/oss/gstossmixertrack.h:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssink.h:
	* sys/oss/gstosssrc.c:
	* sys/oss/gstosssrc.h:
	* sys/osxaudio/gstosxaudioelement.h:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.h:
	  expand tabs
	  Original commit message from CVS:
	  expand tabs

2005-12-05 18:12:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.10.0 ===

2005-12-05 18:03:23 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  releasing 0.10.0
	  Original commit message from CVS:
	  releasing 0.10.0

2005-12-05 18:01:48 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-qtdemux.xml:
	  releasing 0.10.0
	  Original commit message from CVS:
	  releasing 0.10.0

2005-12-05 16:21:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2005-12-05 15:08:46 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/it.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  update translations
	  Original commit message from CVS:
	  update translations

2005-12-05 13:04:22 +0000  Andy Wingo <wingo@pobox.com>

	  Update for alloc_buffer changes.
	  Original commit message from CVS:
	  2005-12-05  Andy Wingo  <wingo@pobox.com>
	  * ext/faac/gstfaac.c: (gst_faac_sink_event), (gst_faac_chain):
	  * ext/faad/gstfaad.c: (gst_faad_chain):
	  * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_chain):
	  * ext/lcs/gstcolorspace.c: (gst_colorspace_chain):
	  * ext/xine/xineinput.c: (gst_xine_input_get):
	  * gst/colorspace/gstcolorspace.c: (gst_colorspace_chain):
	  * gst/speed/gstspeed.c: (speed_chain):
	  * gst/videocrop/gstvideocrop.c: (gst_video_crop_chain): Update for
	  alloc_buffer changes.

2005-12-05 13:03:00 +0000  Andy Wingo <wingo@pobox.com>

	  Update for alloc_buffer changes.
	  Original commit message from CVS:
	  2005-12-05  Andy Wingo  <wingo@pobox.com>
	  * ext/dv/gstdvdec.c: (gst_dvdec_chain):
	  * ext/flac/gstflacdec.c: (gst_flacdec_write):
	  * ext/flac/gstflacenc.c: (gst_flacenc_write_callback):
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain):
	  * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_chain):
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	  * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_process):
	  * ext/libpng/gstpngdec.c: (user_info_callback), (gst_pngdec_task):
	  * ext/speex/gstspeexdec.c: (speex_dec_chain):
	  * ext/speex/gstspeexenc.c: (gst_speexenc_chain):
	  * gst/auparse/gstauparse.c: (gst_auparse_chain):
	  * gst/flx/gstflxdec.c: (gst_flxdec_chain):
	  * gst/goom/gstgoom.c: (gst_goom_chain):
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_push_vorbis_codec_priv_data),
	  (gst_matroska_demux_add_wvpk_header):
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	  * gst/videomixer/videomixer.c: (gst_videomixer_collected):
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Update for
	  alloc_buffer changes.

2005-12-05 12:23:22 +0000  Michael Smith <msmith@xiph.org>

	  docs/plugins/gst-plugins-good-plugins.args: Remove args for plugins that aren't in -good.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-good-plugins.args:
	  Remove args for plugins that aren't in -good.

2005-12-04 22:26:07 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  remove pango plugin as its gone into base
	  Original commit message from CVS:
	  remove pango plugin as its gone into base

2005-12-03 18:51:48 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpg711pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpspeexpay.c:
	  fix element descriptions
	  Original commit message from CVS:
	  fix element descriptions

2005-12-03 18:50:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-fdsrc.xml:
	  remove fdsrc docs
	  Original commit message from CVS:
	  remove fdsrc docs

2005-12-01 19:18:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.9.7 ===

2005-12-01 19:14:26 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  releasing 0.9.7
	  Original commit message from CVS:
	  releasing 0.9.7

2005-12-01 19:13:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-qtdemux.xml:
	  releasing 0.9.7
	  Original commit message from CVS:
	  releasing 0.9.7

2005-12-01 17:53:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2005-12-01 15:34:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* docs/plugins/.gitignore:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	  add multipart plugin to docs
	  Original commit message from CVS:
	  add multipart plugin to docs

2005-12-01 15:22:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	* ext/Makefile.am:
	* ext/pango/Makefile.am:
	* ext/pango/gstclockoverlay.c:
	* ext/pango/gstclockoverlay.h:
	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	* ext/pango/gsttextrender.c:
	* ext/pango/gsttextrender.h:
	* ext/pango/gsttimeoverlay.c:
	* ext/pango/gsttimeoverlay.h:
	  move pango to base
	  Original commit message from CVS:
	  move pango to base

2005-12-01 14:39:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/rtp/: parsers are depayers
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtpL16depay.c:
	  * gst/rtp/gstrtpL16depay.h:
	  * gst/rtp/gstrtpL16parse.c:
	  * gst/rtp/gstrtpL16parse.h:
	  * gst/rtp/gstrtpgsmdepay.c:
	  * gst/rtp/gstrtpgsmdepay.h:
	  * gst/rtp/gstrtpgsmparse.c:
	  * gst/rtp/gstrtpgsmparse.h:
	  parsers are depayers

2005-12-01 14:30:01 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* common:
	* gst/rtp/Makefile.am:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16enc.c:
	* gst/rtp/gstrtpL16enc.h:
	* gst/rtp/gstrtpL16parse.c:
	* gst/rtp/gstrtpL16parse.h:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL16pay.h:
	* gst/rtp/gstrtpamrdec.c:
	* gst/rtp/gstrtpamrdec.h:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrenc.c:
	* gst/rtp/gstrtpamrenc.h:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpdec.c:
	* gst/rtp/gstrtpdec.h:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpdepay.h:
	* gst/rtp/gstrtpg711dec.c:
	* gst/rtp/gstrtpg711dec.h:
	* gst/rtp/gstrtpg711depay.c:
	* gst/rtp/gstrtpg711depay.h:
	* gst/rtp/gstrtpg711enc.c:
	* gst/rtp/gstrtpg711enc.h:
	* gst/rtp/gstrtpg711pay.c:
	* gst/rtp/gstrtpg711pay.h:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmenc.h:
	* gst/rtp/gstrtpgsmparse.c:
	* gst/rtp/gstrtpgsmparse.h:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtph263enc.c:
	* gst/rtp/gstrtph263enc.h:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	* gst/rtp/gstrtph263pdec.c:
	* gst/rtp/gstrtph263pdec.h:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph263penc.c:
	* gst/rtp/gstrtph263penc.h:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtpmp4vdec.c:
	* gst/rtp/gstrtpmp4vdec.h:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vdepay.h:
	* gst/rtp/gstrtpmp4venc.c:
	* gst/rtp/gstrtpmp4venc.h:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadec.c:
	* gst/rtp/gstrtpmpadec.h:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpadepay.h:
	* gst/rtp/gstrtpmpaenc.c:
	* gst/rtp/gstrtpmpaenc.h:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpapay.h:
	* gst/rtp/gstrtpspeexdec.c:
	* gst/rtp/gstrtpspeexdec.h:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexdepay.h:
	* gst/rtp/gstrtpspeexenc.c:
	* gst/rtp/gstrtpspeexenc.h:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpspeexpay.h:
	  Do burger's rename for rtp payloaders and depayloaders
	  Original commit message from CVS:
	  Do burger's rename for rtp payloaders and depayloaders

2005-11-30 19:02:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/dv/: Fix seeking in dvdemux again, add some more debug info.
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c: (gst_dvdec_chain):
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame):
	  * ext/dv/gstdvdemux.h:
	  Fix seeking in dvdemux again, add some more debug info.

2005-11-30 18:48:56 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	  fix tests
	  Original commit message from CVS:
	  fix tests

2005-11-30 18:40:19 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	  add tests subdir
	  Original commit message from CVS:
	  add tests subdir

2005-11-30 18:36:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/Makefile.am:
	  add Makefile.am
	  Original commit message from CVS:
	  add Makefile.am

2005-11-30 18:28:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  move
	  Original commit message from CVS:
	  * PORTED_09:
	  * docs/random/PORTED_09:
	  move
	  * tests/Makefile.am:
	  add
	  * win32/gst.sln:
	  remove

2005-11-30 18:24:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* Makefile.am:
	* check/.gitignore:
	* check/Makefile.am:
	* check/elements/.gitignore:
	* check/elements/level.c:
	* check/elements/matroskamux.c:
	* configure.ac:
	* examples/Makefile.am:
	* examples/capsfilter/Makefile.am:
	* examples/capsfilter/capsfilter1.c:
	* examples/gob/Makefile.am:
	* examples/gob/gst-identity2.gob:
	* examples/gstplay/.gitignore:
	* examples/gstplay/Makefile.am:
	* examples/gstplay/player.c:
	* examples/indexing/.gitignore:
	* examples/indexing/Makefile.am:
	* examples/indexing/indexmpeg.c:
	* examples/level/Makefile.am:
	* examples/level/README:
	* examples/level/demo.c:
	* examples/level/plot.c:
	* examples/stats/Makefile.am:
	* examples/stats/mp2ogg.c:
	* examples/switch/.gitignore:
	* examples/switch/Makefile.am:
	* examples/switch/switcher.c:
	  move under tests
	  Original commit message from CVS:
	  move under tests

2005-11-30 16:57:57 +0000  Christian Schaller <uraeus@gnome.org>

	* common:
	* gst-plugins-good.spec.in:
	  update for latest changes
	  Original commit message from CVS:
	  update for latest changes

2005-11-30 14:53:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextrender.*: Add missing files.
	  Original commit message from CVS:
	  * ext/pango/gsttextrender.c: (gst_text_render_base_init),
	  (gst_text_render_class_init), (resize_bitmap),
	  (gst_text_render_render_text), (gst_text_render_setcaps),
	  (gst_text_render_fixate_caps), (gst_text_renderer_bitmap_to_ayuv),
	  (gst_text_render_chain), (gst_text_render_finalize),
	  (gst_text_render_init), (gst_text_render_set_property):
	  * ext/pango/gsttextrender.h:
	  Add missing files.

2005-11-30 13:20:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Port pango-based textoverlay, timeoverlay and textrender to 0.9 and add background shading and text wrapping modes. M...
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/Makefile.am:
	  * ext/pango/Makefile.am:
	  * ext/pango/gstclockoverlay.c: (gst_clock_overlay_base_init),
	  (gst_clock_overlay_render_time), (gst_clock_overlay_get_text),
	  (gst_clock_overlay_class_init), (gst_clock_overlay_init):
	  * ext/pango/gstclockoverlay.h:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
	  (gst_text_overlay_get_text), (gst_text_overlay_class_init),
	  (gst_text_overlay_finalize), (gst_text_overlay_init),
	  (gst_text_overlay_update_wrap_mode), (gst_text_overlay_setcaps),
	  (gst_text_overlay_text_pad_linked),
	  (gst_text_overlay_text_pad_unlinked),
	  (gst_text_overlay_set_property), (gst_text_overlay_getcaps),
	  (gst_text_overlay_shade_y), (gst_text_overlay_blit_yuv420),
	  (gst_text_overlay_resize_bitmap), (gst_text_overlay_render_text),
	  (gst_text_overlay_push_frame), (gst_text_overlay_pop_video),
	  (gst_text_overlay_pop_text), (gst_text_overlay_collected),
	  (gst_text_overlay_change_state), (plugin_init):
	  * ext/pango/gsttextoverlay.h:
	  * ext/pango/gsttimeoverlay.c: (gst_time_overlay_base_init),
	  (gst_time_overlay_render_time), (gst_time_overlay_get_text),
	  (gst_time_overlay_class_init), (gst_time_overlay_init):
	  * ext/pango/gsttimeoverlay.h:
	  Port pango-based textoverlay, timeoverlay and textrender to 0.9
	  and add background shading and text wrapping modes. Make
	  timoverlay derive from textoverlay. Also add new clockoverlay
	  element.

2005-11-30 11:10:01 +0000  Julien Moutte <julien@moutte.net>

	  gst/udp/Makefile.am: Moved to netbuffer.
	  Original commit message from CVS:
	  2005-11-30  Julien MOUTTE  <julien@moutte.net>
	  * gst/udp/Makefile.am: Moved to netbuffer.

2005-11-30 10:18:42 +0000  Julien Moutte <julien@moutte.net>

	  Ported multipart mux/demux to 0.9.
	  Original commit message from CVS:
	  2005-11-30  Julien MOUTTE  <julien@moutte.net>
	  * configure.ac:
	  * PORTED_O9:
	  * gst/multipart/Makefile.am:
	  * gst/multipart/multipartdemux.c:
	  (gst_multipart_demux_base_init),
	  (gst_multipart_demux_class_init), (gst_multipart_demux_init),
	  (gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain),
	  (gst_multipart_demux_change_state),
	  (gst_multipart_demux_plugin_init):
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
	  (gst_multipart_mux_init), (gst_multipart_mux_finalize),
	  (gst_multipart_mux_sinkconnect),
	  (gst_multipart_mux_request_new_pad),
	  (gst_multipart_mux_handle_src_event),
	  (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
	  (gst_multipart_mux_change_state): Ported multipart mux/demux to
	  0.9.

2005-11-30 08:26:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/: update for symbols change
	  Original commit message from CVS:
	  * gst/debug/gstnavigationtest.c: (gst_navigationtest_get_type):
	  * gst/debug/gstnavigationtest.h:
	  * gst/effectv/gstaging.c: (gst_agingtv_get_type):
	  * gst/effectv/gstdice.c: (gst_dicetv_get_type):
	  * gst/effectv/gstedge.c: (gst_edgetv_get_type):
	  * gst/effectv/gstquark.c: (gst_quarktv_get_type):
	  * gst/effectv/gstrev.c: (gst_revtv_get_type):
	  * gst/effectv/gstshagadelic.c: (gst_shagadelictv_get_type):
	  * gst/effectv/gstvertigo.c: (gst_vertigotv_get_type):
	  * gst/effectv/gstwarp.c: (gst_warptv_get_type):
	  * gst/videofilter/gstvideoflip.c: (gst_video_flip_set_property),
	  (gst_video_flip_get_type):
	  * gst/videofilter/gstvideoflip.h:
	  update for symbols change

2005-11-29 17:46:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/udp/: the old gstnet lib was renamed gstnetbuffer (#322257)
	  Original commit message from CVS:
	  * gst/udp/gstdynudpsink.c:
	  * gst/udp/gstudpsrc.c:
	  the old gstnet lib was renamed gstnetbuffer (#322257)

2005-11-29 15:42:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cairo/gsttextoverlay.c: Actually render the text from the text pad.
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_render_text),
	  (gst_text_overlay_collected):
	  Actually render the text from the text pad.

2005-11-29 14:49:00 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/debug/: Update for GstBaseTransform event virtual method
	  Original commit message from CVS:
	  * gst/debug/gstnavseek.c: (gst_navseek_event):
	  * gst/debug/progressreport.c: (gst_progress_report_event):
	  Update for GstBaseTransform event virtual method

2005-11-29 10:55:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/cairo/Makefile.am: no need to link to videofilter
	  Original commit message from CVS:
	  2005-11-29  Thomas Vander Stichele  <thomas at apestaart dot org>
	  * ext/cairo/Makefile.am:
	  no need to link to videofilter

2005-11-29 10:46:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* gst/debug/Makefile.am:
	* gst/debug/gstnavigationtest.h:
	* gst/effectv/Makefile.am:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstvideofilter.c:
	* gst/videofilter/gstvideofilter.h:
	* gst/videofilter/gstvideoflip.h:
	  remove the videofilter library and link to the one in base
	  Original commit message from CVS:
	  remove the videofilter library and link to the one in base

2005-11-29 01:30:40 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideoflip.h:
	  borgify
	  Original commit message from CVS:
	  borgify

2005-11-28 17:31:44 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.c: Useless check now we're setting the current entry correctly.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
	  Useless check now we're setting the current entry correctly.

2005-11-28 16:54:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/jpeg/gstjpegenc.c: Don't leak input buffer in chain function (fixes #322667); make state change function thread-s...
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_resync), (gst_jpegenc_chain),
	  (gst_jpegenc_set_property), (gst_jpegenc_get_property),
	  (gst_jpegenc_change_state):
	  Don't leak input buffer in chain function (fixes #322667); make
	  state change function thread-safe; don't repeat the current function
	  name in GST_DEBUG statements; use GST_ROUND_UP_* macros; use
	  gst_pad_alloc_buffer(); misc. minor cleanups.

2005-11-28 15:43:29 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/faad/gstfaad.c: Handle gracefully the consequence of "Maximum number of scalefactor bands exceeded", which result...
	  Original commit message from CVS:
	  * ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
	  Handle gracefully the consequence of "Maximum number of scalefactor
	  bands exceeded", which results in 0 channels with samplerates of 0.
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state):
	  Do upward transitions, then call parent state_change, then do
	  downward transitions.

2005-11-28 15:13:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/matroska/matroska-mux.c: Look for pixel-aspect-ratio in caps, not pixel_width and pixel_height (Fixes: #322645)
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_video_pad_setcaps):
	  Look for pixel-aspect-ratio in caps, not pixel_width and
	  pixel_height (Fixes: #322645)

2005-11-28 12:59:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/matroska/matroska-mux.c: From Michal Benes: frame duration should be GST_SECOND / framerate, not
	  Original commit message from CVS:
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_video_pad_setcaps):
	  From Michal Benes:
	  frame duration should be GST_SECOND / framerate, not
	  GST_SECOND * framerate. (Fixes: #322643)

2005-11-27 17:02:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: fix up GST_PLUGIN_LDFLAGS
	  Original commit message from CVS:
	  * configure.ac:
	  fix up GST_PLUGIN_LDFLAGS
	  * gst/rtsp/rtspconnection.c:
	  fix includes (see #317043)
	  * gst/videofilter/Makefile.am:
	  stop installing this library

2005-11-27 15:30:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  no need for an AS_LIBTOOL call
	  Original commit message from CVS:
	  no need for an AS_LIBTOOL call

2005-11-27 14:33:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	* common:
	* gst-plugins-good.spec.in:
	  add ACLOCAL_AMFLAGS; remove old stuff from spec changelog
	  Original commit message from CVS:
	  add ACLOCAL_AMFLAGS; remove old stuff from spec changelog

2005-11-26 12:54:47 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/dv/gstdvdec.c: Handle the case where the incoming Video dv stream doesn't have a pixel aspect ratio set.
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c: (gst_dvdec_sink_setcaps):
	  Handle the case where the incoming Video dv stream doesn't have
	  a pixel aspect ratio set.

2005-11-25 22:14:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* ext/flac/gstflacdec.c:
	  document flacdec
	  Original commit message from CVS:
	  document flacdec

2005-11-25 21:36:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* ext/cairo/gstcairo.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttextoverlay.h:
	* ext/cairo/gsttimeoverlay.c:
	* ext/cairo/gsttimeoverlay.h:
	  do some name borgifying document
	  Original commit message from CVS:
	  do some name borgifying
	  document

2005-11-25 21:02:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  documenting auto*sink using strstr for the video sink lookup, class field is not ordered update other plugins
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_base_init):
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_base_init),
	  (gst_auto_video_sink_factory_filter):
	  documenting auto*sink
	  using strstr for the video sink lookup, class field is not ordered
	  update other plugins

2005-11-25 19:58:19 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ext/wavpack/Makefile.am:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackdec.h:
	* ext/wavpack/gstwavpackparse.c:
	* ext/wavpack/gstwavpackparse.h:
	  Wavpack ported to 0.9. No support for correction file yet.
	  Original commit message from CVS:
	  Wavpack ported to 0.9. No support for correction file yet.

2005-11-25 18:15:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/wavpack/: put back wavpack - still needs porting
	  Original commit message from CVS:
	  * ext/wavpack/gstwavpackcommon.h:
	  * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link),
	  (gst_wavpack_dec_wvclink), (gst_wavpack_dec_get_type),
	  (gst_wavpack_dec_base_init), (gst_wavpack_dec_dispose),
	  (gst_wavpack_dec_class_init), (gst_wavpack_dec_src_query),
	  (gst_wavpack_dec_init), (gst_wavpack_dec_setup_context),
	  (gst_wavpack_dec_format_samples), (gst_wavpack_dec_loop),
	  (gst_wavpack_dec_plugin_init):
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_get_type),
	  (gst_wavpack_parse_base_init), (gst_wavpack_parse_dispose),
	  (gst_wavpack_parse_class_init), (gst_wavpack_parse_src_query),
	  (gst_wavpack_parse_src_event), (find_header), (find_sample),
	  (gst_wavpack_parse_seek), (gst_wavpack_parse_init),
	  (gst_wavpack_parse_handle_event), (gst_wavpack_parse_loop),
	  (gst_wavpack_parse_change_state), (gst_wavpack_parse_plugin_init):
	  * ext/wavpack/gstwavpackparse.h:
	  put back wavpack - still needs porting

2005-11-25 18:03:24 +0000  Sebastien Cote <sebas642@yahoo.ca>

	  gst/udp/gstudpsrc.c: Patch from Sebastien Cote to close control sockets in udpsrc.
	  Original commit message from CVS:
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_stop):
	  Patch from Sebastien Cote to close control sockets in udpsrc.

2005-11-24 15:07:06 +0000  Julien Moutte <julien@moutte.net>

	  gst/effectv/gstquark.c: Flush the planes list on reverse caps negotiation. This was crashing because of differently s...
	  Original commit message from CVS:
	  2005-11-24  Julien MOUTTE  <julien@moutte.net>
	  * gst/effectv/gstquark.c: (gst_quarktv_set_caps),
	  (gst_quarktv_get_unit_size), (gst_quarktv_transform),
	  (gst_quarktv_planetable_clear), (gst_quarktv_change_state),
	  (gst_quarktv_base_init), (gst_quarktv_class_init),
	  (gst_quarktv_init): Flush the planes list on reverse caps
	  negotiation. This was crashing because of differently sized
	  buffers.

2005-11-24 12:50:28 +0000  Julien Moutte <julien@moutte.net>

	  gst/: Handle strides correctly, fix identity flipping, convert navigation event correctly again.
	  Original commit message from CVS:
	  2005-11-24  Julien MOUTTE  <julien@moutte.net>
	  * gst/debug/gstnavigationtest.c: (draw_box_planar411):
	  * gst/videofilter/gstvideoflip.c:
	  (gst_videoflip_method_get_type),
	  (gst_videoflip_set_caps), (gst_videoflip_transform_caps),
	  (gst_videoflip_get_unit_size), (gst_videoflip_flip),
	  (gst_videoflip_transform), (gst_videoflip_handle_src_event),
	  (gst_videoflip_set_property), (gst_videoflip_base_init),
	  (gst_videoflip_class_init), (gst_videoflip_init): Handle strides
	  correctly, fix identity flipping, convert navigation event
	  correctly again.

2005-11-24 11:16:53 +0000  Michael Smith <msmith@xiph.org>

	* README:
	  Fix #320288: wrong readme in plugins-good
	  Original commit message from CVS:
	  Fix #320288: wrong readme in plugins-good

2005-11-24 11:06:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	  fix torture target
	  Original commit message from CVS:
	  fix torture target

2005-11-23 21:25:56 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	  add a torture target
	  Original commit message from CVS:
	  add a torture target

2005-11-23 20:05:26 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.9.6 ===

2005-11-23 19:57:49 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-qtdemux.xml:
	  releasing 0.9.6
	  Original commit message from CVS:
	  releasing 0.9.6

2005-11-23 19:56:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  releasing 0.9.6
	  Original commit message from CVS:
	  releasing 0.9.6

2005-11-23 19:14:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/inspect/plugin-cutter.xml:
	  adding cutter
	  Original commit message from CVS:
	  adding cutter

2005-11-23 19:05:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2005-11-23 16:49:16 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/debug/gstnavigationtest.c: Oops, initialise the framerate GValue
	  Original commit message from CVS:
	  * gst/debug/gstnavigationtest.c: (gst_navigationtest_init):
	  Oops, initialise the framerate GValue

2005-11-23 15:50:51 +0000  Julien Moutte <julien@moutte.net>

	  VideoFilter inherits from
	  Original commit message from CVS:
	  2005-11-23  Julien MOUTTE  <julien@moutte.net>
	  * ext/cairo/gsttimeoverlay.c:
	  (gst_timeoverlay_update_font_height),
	  (gst_timeoverlay_set_caps), (gst_timeoverlay_get_unit_size),
	  (gst_timeoverlay_transform), (gst_timeoverlay_base_init),
	  (gst_timeoverlay_class_init), (gst_timeoverlay_init),
	  (gst_timeoverlay_get_type):
	  * ext/cairo/gsttimeoverlay.h:
	  * gst/debug/Makefile.am:
	  * gst/debug/gstnavigationtest.c:
	  (gst_navigationtest_handle_src_event),
	  (gst_navigationtest_get_unit_size),
	  (gst_navigationtest_set_caps),
	  (gst_navigationtest_transform),
	  (gst_navigationtest_change_state),
	  (gst_navigationtest_base_init), (gst_navigationtest_class_init),
	  (gst_navigationtest_init), (gst_navigationtest_get_type),
	  (plugin_init):
	  * gst/debug/gstnavigationtest.h:
	  * gst/effectv/Makefile.am:
	  * gst/effectv/gstaging.c: (gst_agingtv_set_caps),
	  (gst_agingtv_get_unit_size), (gst_agingtv_transform),
	  (gst_agingtv_base_init), (gst_agingtv_class_init),
	  (gst_agingtv_init), (gst_agingtv_get_type):
	  * gst/effectv/gstdice.c: (gst_dicetv_set_caps),
	  (gst_dicetv_get_unit_size), (gst_dicetv_transform),
	  (gst_dicetv_base_init), (gst_dicetv_class_init),
	  (gst_dicetv_init),
	  (gst_dicetv_get_type):
	  * gst/effectv/gstedge.c: (gst_edgetv_set_caps),
	  (gst_edgetv_get_unit_size), (gst_edgetv_transform),
	  (gst_edgetv_base_init), (gst_edgetv_class_init),
	  (gst_edgetv_init),
	  (gst_edgetv_get_type):
	  * gst/effectv/gsteffectv.c:
	  * gst/effectv/gsteffectv.h:
	  * gst/effectv/gstquark.c: (gst_quarktv_set_caps),
	  (gst_quarktv_get_unit_size), (fastrand),
	  (gst_quarktv_transform),
	  (gst_quarktv_change_state), (gst_quarktv_base_init),
	  (gst_quarktv_class_init), (gst_quarktv_init),
	  (gst_quarktv_get_type):
	  * gst/effectv/gstrev.c: (gst_revtv_set_caps),
	  (gst_revtv_get_unit_size), (gst_revtv_transform),
	  (gst_revtv_base_init), (gst_revtv_class_init), (gst_revtv_init),
	  (gst_revtv_get_type):
	  * gst/effectv/gstshagadelic.c: (gst_shagadelictv_set_caps),
	  (gst_shagadelictv_get_unit_size), (gst_shagadelictv_transform),
	  (gst_shagadelictv_base_init), (gst_shagadelictv_class_init),
	  (gst_shagadelictv_init), (gst_shagadelictv_get_type):
	  * gst/effectv/gstvertigo.c: (gst_vertigotv_set_caps),
	  (gst_vertigotv_get_unit_size), (gst_vertigotv_transform),
	  (gst_vertigotv_base_init), (gst_vertigotv_class_init),
	  (gst_vertigotv_init), (gst_vertigotv_get_type):
	  * gst/effectv/gstwarp.c: (gst_warptv_set_caps),
	  (gst_warptv_get_unit_size), (gst_warptv_transform),
	  (gst_warptv_base_init), (gst_warptv_class_init),
	  (gst_warptv_init),
	  (gst_warptv_get_type):
	  * gst/videofilter/Makefile.am:
	  * gst/videofilter/gstvideobalance.c:
	  * gst/videofilter/gstvideobalance.h:
	  * gst/videofilter/gstvideofilter.c: (gst_videofilter_get_type),
	  (gst_videofilter_class_init), (gst_videofilter_init):
	  * gst/videofilter/gstvideofilter.h:
	  * gst/videofilter/gstvideoflip.c: (gst_videoflip_set_caps),
	  (gst_videoflip_transform_caps), (gst_videoflip_get_unit_size),
	  (gst_videoflip_flip), (gst_videoflip_transform),
	  (gst_videoflip_handle_src_event), (gst_videoflip_set_property),
	  (gst_videoflip_base_init), (gst_videoflip_class_init),
	  (gst_videoflip_init), (plugin_init), (gst_videoflip_get_type):
	  * gst/videofilter/gstvideoflip.h: VideoFilter inherits from
	  BaseTransform, it's just a place holder for now and every video
	  effect plugin has been ported to use BaseTransform features
	  directly. QuarkTV was fixed too (was broken), navigationtest
	  works
	  and best for the end, videoflip converts navigation events
	  depending
	  on flip method ! Fixes #320953

2005-11-23 14:22:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Fixes for API changes
	  Original commit message from CVS:
	  * ext/aalib/gstaasink.c: (gst_aasink_fixate):
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_collected):
	  * gst/goom/gstgoom.c: (gst_goom_init), (gst_goom_src_setcaps),
	  (gst_goom_src_negotiate), (gst_goom_chain):
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_video_pad_setcaps):
	  * sys/osxvideo/osxvideosink.m:
	  Fixes for API changes

2005-11-23 12:19:06 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add cutter to spec in
	  Original commit message from CVS:
	  add cutter to spec in

2005-11-23 11:57:51 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/qtdemux/qtdemux.c: Convert to fractional framerates
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
	  (gst_qtdemux_add_stream), (qtdemux_dump_mvhd),
	  (qtdemux_parse_trak):
	  Convert to fractional framerates

2005-11-22 23:58:14 +0000  Michael Smith <msmith@xiph.org>

	  ext/jpeg/: JPEG fractiony goodness.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_setcaps),
	  (gst_jpeg_dec_chain), (gst_jpeg_dec_change_state):
	  * ext/jpeg/gstjpegdec.h:
	  * ext/jpeg/gstjpegenc.c: (gst_jpegenc_setcaps):
	  * ext/jpeg/gstjpegenc.h:
	  * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps),
	  (gst_smokeenc_resync):
	  * ext/jpeg/gstsmokeenc.h:
	  JPEG fractiony goodness.

2005-11-22 22:35:57 +0000  Michael Smith <msmith@xiph.org>

	* ChangeLog:
	* gst/goom/filters.c:
	* gst/goom/graphic.h:
	  Fix for #321430: unresolved symbols due to incorrect linkage on inline functions in goom.
	  Original commit message from CVS:
	  Fix for #321430: unresolved symbols due to incorrect linkage on inline functions
	  in goom.
	  Does not, however, fix the general crackheadedness of goom (global variables,
	  oh my!); this should be moved to -bad.

2005-11-22 22:21:37 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  More fractional framerate conversions
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_init),
	  (gst_text_overlay_setcaps), (gst_text_overlay_collected):
	  * ext/cairo/gsttextoverlay.h:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link):
	  * ext/gdk_pixbuf/gstgdkpixbuf.h:
	  * ext/libpng/gstpngdec.c: (gst_pngdec_init),
	  (gst_pngdec_caps_create_and_set):
	  * ext/libpng/gstpngdec.h:
	  * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps):
	  * gst/alpha/gstalphacolor.c: (gst_alpha_color_set_caps):
	  * gst/avi/gstavimux.c: (gst_avimux_init),
	  (gst_avimux_vidsinkconnect):
	  * gst/flx/gstflxdec.c: (gst_flxdec_chain):
	  * gst/goom/gstgoom.c: (gst_goom_init), (gst_goom_src_setcaps),
	  (gst_goom_src_negotiate), (gst_goom_chain):
	  * gst/goom/gstgoom.h:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_video_pad_setcaps):
	  * sys/osxvideo/osxvideosink.h:
	  * sys/osxvideo/osxvideosink.m:
	  More fractional framerate conversions

2005-11-22 20:07:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Convert to fractional framerates.
	  Original commit message from CVS:
	  * ext/aalib/gstaasink.c: (gst_aasink_fixate):
	  * gst/debug/gstnavigationtest.c:
	  (gst_navigationtest_handle_src_event):
	  * gst/videofilter/gstvideofilter.c:
	  (gst_videofilter_format_get_structure), (gst_videofilter_setcaps),
	  (gst_videofilter_init):
	  * gst/videofilter/gstvideofilter.h:
	  Convert to fractional framerates.

2005-11-22 18:11:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* ext/aalib/gstaasink.c:
	* ext/dv/gstdvdec.c:
	* ext/esd/esdmon.c:
	* ext/flac/gstflacenc.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/libcaca/gstcacasink.c:
	* ext/shout2/gstshout2.c:
	* gst/alpha/gstalpha.c:
	* gst/oldcore/gstaggregator.c:
	* gst/oldcore/gstshaper.c:
	* gst/smpte/barboxwipes.c:
	* gst/smpte/gstsmpte.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videomixer/videomixer.c:
	  fix up more enums
	  Original commit message from CVS:
	  fix up more enums

2005-11-22 17:39:11 +0000  Michael Smith <msmith@xiph.org>

	  gst/videomixer/videomixer.c: Fractional framerates, videomixer.
	  Original commit message from CVS:
	  * gst/videomixer/videomixer.c: (gst_videomixer_pad_sink_setcaps),
	  (gst_videomixer_getcaps), (gst_videomixer_fill_queues),
	  (gst_videomixer_update_queues):
	  Fractional framerates, videomixer.

2005-11-22 17:15:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  doh
	  Original commit message from CVS:
	  doh

2005-11-22 17:09:36 +0000  Michael Smith <msmith@xiph.org>

	  ext/dv/: Fractional framerates for DV.
	  Original commit message from CVS:
	  * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps):
	  * ext/dv/gstdvdec.h:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_init),
	  (gst_dvdemux_src_convert), (gst_dvdemux_sink_convert),
	  (gst_dvdemux_demux_video), (gst_dvdemux_demux_frame),
	  (gst_dvdemux_flush):
	  * ext/dv/gstdvdemux.h:
	  Fractional framerates for DV.

2005-11-22 17:04:38 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  fix up GValueEnum
	  Original commit message from CVS:
	  fix up GValueEnum

2005-11-22 14:44:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/autodetect/: Use gst_plugin_feature_list_free() to free feature list and in the case of autovideosink free the li...
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect):
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_find_best), (gst_auto_video_sink_detect):
	  Use gst_plugin_feature_list_free() to free feature list and
	  in the case of autovideosink free the list at all. Also
	  miscellaneous cosmetic fixes.

2005-11-22 13:13:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/cutter/gstcutter.c: copy calculation code from level; remove use of some audio functions
	  Original commit message from CVS:
	  * gst/cutter/gstcutter.c: (gst_cutter_chain),
	  (gst_cutter_set_property), (gst_cutter_get_caps):
	  copy calculation code from level; remove use of some audio
	  functions

2005-11-22 13:11:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/level/gstlevel.c:
	  various cosmetic fixes
	  Original commit message from CVS:
	  various cosmetic fixes

2005-11-22 12:48:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/level/gstlevel.c:
	  various cosmetic fixes
	  Original commit message from CVS:
	  various cosmetic fixes

2005-11-22 12:41:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/level/gstlevel.c:
	  various cosmetic fixes
	  Original commit message from CVS:
	  various cosmetic fixes

2005-11-22 12:39:29 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	  Update for gst_tag_setter API changes.
	  Original commit message from CVS:
	  2005-11-22  Andy Wingo  <wingo@pobox.com>
	  * Update for gst_tag_setter API changes.

2005-11-22 12:38:33 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/shout2/gstshout2.c:
	* ext/speex/gstspeexenc.c:
	* gst/avi/gstavimux.c:
	  Update for gst_tag_setter API changes.
	  Original commit message from CVS:
	  2005-11-22  Andy Wingo  <wingo@pobox.com>
	  * Update for gst_tag_setter API changes.

2005-11-22 11:57:51 +0000  Andy Wingo <wingo@pobox.com>

	* gst/qtdemux/qtdemux.c:
	  ext/faad/gstfaad.c (gst_faad_event) ext/ivorbis/vorbisfile.c (gst_ivorbisfile_loop) gst/qtdemux/qtdemux.c (gst_qtdemu...
	  Original commit message from CVS:
	  2005-11-22  Andy Wingo  <wingo@pobox.com>
	  * ext/faad/gstfaad.c (gst_faad_event)
	  * ext/ivorbis/vorbisfile.c (gst_ivorbisfile_loop)
	  * gst/qtdemux/qtdemux.c (gst_qtdemux_loop_header)
	  * gst/speed/gstspeed.c (speed_sink_event)
	  * gst/tta/gstttaparse.c (gst_tta_parse_src_event)
	  (gst_tta_parse_parse_header): Run update-funcnames.

2005-11-22 11:53:34 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/gconf/gstgconfaudiosink.c:
	* ext/gconf/gstgconfvideosink.c:
	* ext/libpng/gstpngdec.c:
	* ext/speex/gstspeexdec.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/avi/gstavidemux.c:
	* gst/goom/gstgoom.c:
	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-demux.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	  ext/dv/gstdvdemux.c (gst_dvdemux_handle_sink_event) (gst_dvdemux_demux_frame) ext/flac/gstflacdec.c (gst_flacdec_writ...
	  Original commit message from CVS:
	  2005-11-22  Andy Wingo  <wingo@pobox.com>
	  * ext/dv/gstdvdemux.c (gst_dvdemux_handle_sink_event)
	  (gst_dvdemux_demux_frame)
	  * ext/flac/gstflacdec.c (gst_flacdec_write)
	  * ext/flac/gstflacenc.c (gst_flacenc_seek_callback)
	  (gst_flacenc_sink_event)
	  * ext/gconf/gstgconfaudiosink.c (gst_gconf_audio_sink_init)
	  * ext/gconf/gstgconfvideosink.c (gst_gconf_video_sink_init)
	  * ext/libpng/gstpngdec.c (gst_pngdec_caps_create_and_set)
	  * ext/speex/gstspeexdec.c (speex_dec_event, speex_dec_chain)
	  * gst/auparse/gstauparse.c (gst_auparse_chain)
	  * gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_init)
	  * gst/autodetect/gstautovideosink.c (gst_auto_video_sink_init)
	  * gst/avi/gstavidemux.c (gst_avi_demux_stream_header)
	  (gst_avi_demux_handle_seek)
	  * gst/goom/gstgoom.c (gst_goom_event)
	  * gst/matroska/ebml-write.c (gst_ebml_write_seek)
	  * gst/matroska/matroska-demux.c
	  (gst_matroska_demux_handle_seek_event)
	  (gst_matroska_demux_loop_stream_parse_id)
	  * gst/wavenc/gstwavenc.c (gst_wavenc_stop_file)
	  * gst/wavparse/gstwavparse.c (gst_wavparse_handle_seek)
	  (gst_wavparse_stream_headers): Run update-funcnames.

2005-11-22 11:49:30 +0000  Edward Hervey <bilboed@bilboed.com>

	  URIHandler interface and element properties are now properly synchronized for DV1394src and UDPSrc
	  Original commit message from CVS:
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_class_init),
	  (gst_dv1394src_init), (gst_dv1394src_dispose),
	  (gst_dv1394src_set_property), (gst_dv1394src_discover_avc_node),
	  (gst_dv1394src_uri_set_uri):
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	  (gst_udpsrc_update_uri), (gst_udpsrc_set_uri),
	  (gst_udpsrc_set_property), (gst_udpsrc_uri_get_uri):
	  URIHandler interface and element properties are now properly
	  synchronized for DV1394src and UDPSrc

2005-11-22 11:36:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/: libgsttagedit has been renamed to libgsttag.
	  Original commit message from CVS:
	  * ext/flac/Makefile.am:
	  * ext/speex/Makefile.am:
	  libgsttagedit has been renamed to libgsttag.

2005-11-21 23:50:02 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/lame/gstlame.c: Don't take the stream lock
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_event):
	  Don't take the stream lock

2005-11-21 20:11:59 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/rtspconnection.c: Apply patch from Sebastien Cote to fix #319184.
	  Original commit message from CVS:
	  * gst/rtsp/rtspconnection.c: (read_body):
	  Apply patch from Sebastien Cote to fix #319184.

2005-11-21 19:50:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  port cutter
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/cutter/Makefile.am:
	  * gst/cutter/gstcutter.c: (gst_cutter_class_init),
	  (gst_cutter_init), (gst_cutter_message_new), (gst_cutter_chain),
	  (gst_cutter_set_property), (gst_cutter_get_property),
	  (plugin_init), (gst_cutter_get_caps):
	  port cutter
	  * gst/level/gstlevel.c:
	  fix up plugin details

2005-11-21 18:09:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Update for stream lock API changes: don't take stream log in sink event handlers any longer and change GST_STREAM_LOC...
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event):
	  * ext/flac/gstflacdec.c: (gst_flacdec_loop),
	  (gst_flacdec_src_event):
	  * ext/flac/gstflacenc.c: (gst_flacenc_sink_event):
	  * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_event),
	  (gst_signal_processor_getrange), (gst_signal_processor_chain):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
	  * gst/flx/gstflxdec.c: (gst_flxdec_src_event_handler),
	  (gst_flxdec_sink_event_handler):
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_handle_seek_event):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek):
	  Update for stream lock API changes: don't take stream log
	  in sink event handlers any longer and change GST_STREAM_LOCK
	  to GST_PAD_STREAM_LOCK. Don't leak references in flxdec event
	  functions.

2005-11-21 17:52:15 +0000  Michael Smith <msmith@xiph.org>

	* gst/auparse/Makefile.am:
	* gst/auparse/gstauparse.h:
	  Forgot to commit header file changes, Makefile.am changes. Oops.
	  Original commit message from CVS:
	  Forgot to commit header file changes, Makefile.am changes. Oops.

2005-11-21 17:49:21 +0000  Michael Smith <msmith@xiph.org>

	* ChangeLog:
	* gst/auparse/gstauparse.c:
	  gst_object_unref, not g_object_unref
	  Original commit message from CVS:
	  gst_object_unref, not g_object_unref

2005-11-21 17:37:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Fix for stream lock updates.
	  Original commit message from CVS:
	  * ext/faac/gstfaac.c: (gst_faac_sink_event):
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_event):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_src_event):
	  Fix for stream lock updates.

2005-11-21 17:23:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/wavparse/gstwavparse.c: Use GST_DEBUG_FUNCPTR; add debug message in pad activate function.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
	  (gst_wavparse_create_sourcepad), (gst_wavparse_sink_activate):
	  Use GST_DEBUG_FUNCPTR; add debug message in pad activate function.

2005-11-21 17:18:01 +0000  Michael Smith <msmith@xiph.org>

	  gst/auparse/: Partially fix #161712. playbin still doesn't work on these files, (on the bug report, Andy says we aren...
	  Original commit message from CVS:
	  * gst/auparse/Makefile.am:
	  * gst/auparse/gstauparse.c: (gst_auparse_class_init),
	  (gst_auparse_init), (gst_auparse_dispose), (gst_auparse_chain),
	  (gst_auparse_change_state):
	  * gst/auparse/gstauparse.h:
	  Partially fix #161712. playbin still doesn't work on these files,
	  (on the bug report, Andy says we aren't typefinding it for some
	  reason?) but at least auparse isn't totally busted like it was before.

2005-11-21 16:45:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: j@bootlab.org, #321903).
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	  Add DX50, DIVX and DIV3 fourccs (patch by
	  j@bootlab.org, #321903).

2005-11-21 16:36:05 +0000  Andy Wingo <wingo@pobox.com>

	  *.*: Ran scripts/update-macros. Oh yes.
	  Original commit message from CVS:
	  2005-11-21  Andy Wingo  <wingo@pobox.com>
	  * *.h:
	  * *.c: Ran scripts/update-macros. Oh yes.

2005-11-21 15:06:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: Filler events are gone for now, comment out section generating them.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams):
	  Filler events are gone for now, comment out section generating
	  them.

2005-11-21 14:39:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Update for GST_FOURCC_FORMAT API change.
	  Original commit message from CVS:
	  * ext/directfb/dfbvideosink.c:
	  (gst_dfbvideosink_get_format_from_caps):
	  * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_create):
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
	  (qtdemux_parse), (qtdemux_type_get), (qtdemux_node_dump_foreach),
	  (qtdemux_dump_hdlr), (qtdemux_dump_dref), (qtdemux_dump_stsd),
	  (qtdemux_dump_dcom), (qtdemux_parse_trak), (qtdemux_video_caps),
	  (qtdemux_audio_caps):
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps):
	  * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	  (gst_v4l2src_capture_init), (gst_v4l2src_get_size_limits):
	  Update for GST_FOURCC_FORMAT API change.

2005-11-21 14:33:11 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027)
	  Original commit message from CVS:
	  * ext/audioresample/gstaudioresample.c:
	  * ext/polyp/polypsink.c: (gst_polypsink_sink_fixate):
	  * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_fixate):
	  * gst/modplug/gstmodplug.cc:
	  * sys/glsink/glimagesink.c: (gst_glimagesink_fixate):
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_fixate):
	  Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
	  (#322027)

2005-11-21 14:31:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027)
	  Original commit message from CVS:
	  * ext/aalib/gstaasink.c: (gst_aasink_fixate):
	  * ext/mikmod/gstmikmod.c: (gst_mikmod_srcfixate):
	  * gst/goom/gstgoom.c: (gst_goom_src_negotiate):
	  * sys/osxvideo/osxvideosink.m:
	  Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
	  (#322027)

2005-11-21 13:38:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Fixes for GST_FOURCC_FORMAT API change.
	  Original commit message from CVS:
	  * ext/aalib/gstaasink.c: (gst_aasink_setcaps):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_file_header),
	  (gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_parse_odml), (gst_avi_demux_stream_index),
	  (gst_avi_demux_sync), (gst_avi_demux_stream_header),
	  (gst_avi_demux_stream_data):
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
	  * gst/wavenc/gstwavenc.c: (write_metadata):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_adtl),
	  (gst_wavparse_parse_file_header), (gst_wavparse_stream_headers):
	  Fixes for GST_FOURCC_FORMAT API change.

2005-11-21 12:13:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Fix for collect pads API change. Also fix textoverlay state change function.
	  Original commit message from CVS:
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_finalize),
	  (gst_text_overlay_init), (gst_text_overlay_text_pad_linked),
	  (gst_text_overlay_text_pad_unlinked), (gst_text_overlay_pop_video),
	  (gst_text_overlay_pop_text), (gst_text_overlay_collected),
	  (gst_text_overlay_change_state):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_init),
	  (gst_matroska_mux_reset), (gst_matroska_mux_request_new_pad),
	  (gst_matroska_mux_best_pad), (gst_matroska_mux_change_state):
	  * gst/smpte/gstsmpte.c: (gst_smpte_init), (gst_smpte_collected):
	  * gst/videomixer/videomixer.c: (gst_videomixer_init),
	  (gst_videomixer_request_new_pad), (gst_videomixer_fill_queues),
	  (gst_videomixer_change_state):
	  Fix for collect pads API change. Also fix textoverlay state
	  change function.

2005-11-20 17:04:55 +0000  Julien Moutte <julien@moutte.net>

	  gst/matroska/matroska-mux.c: Replace
	  Original commit message from CVS:
	  2005-11-20  Julien MOUTTE  <julien@moutte.net>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_start): Replace
	  GST_PAD_IS_USABLE by something approaching it.

2005-11-20 16:43:32 +0000  Julien Moutte <julien@moutte.net>

	  gst/matroska/matroska-mux.c: Fix for
	  Original commit message from CVS:
	  2005-11-20  Julien MOUTTE  <julien@moutte.net>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_start): Fix for
	  API changes.
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Fix for API
	  changes,
	  but also fix the code that was not checking return values from
	  pad_push neither using pad_alloc_buffer.

2005-11-18 18:19:21 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/libpng/gstpngenc.c: Added debug category
	  Original commit message from CVS:
	  * ext/libpng/gstpngenc.c: (gst_pngenc_class_init),
	  (gst_pngenc_chain):
	  Added debug category
	  Return GST_FLOW_UNEXPECTED when sending an EOS, so the whole pipeline
	  goes to EOS.

2005-11-17 18:23:23 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpg711dec.c:
	* gst/rtp/gstrtpg711depay.c:
	* gst/rtp/gstrtpg711enc.c:
	* gst/rtp/gstrtpg711enc.h:
	* gst/rtp/gstrtpg711pay.c:
	* gst/rtp/gstrtpg711pay.h:
	* gst/rtp/gstrtpspeexdec.c:
	* gst/rtp/gstrtpspeexdec.h:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexdepay.h:
	* gst/rtp/gstrtpspeexenc.c:
	* gst/rtp/gstrtpspeexenc.h:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpspeexpay.h:
	  Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size.
	  Original commit message from CVS:
	  Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size.

2005-11-16 19:08:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  check/elements/matroskamux.c: Fix leak in check.
	  Original commit message from CVS:
	  * check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad):
	  Fix leak in check.

2005-11-16 17:00:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/flx/gstflxdec.c: Fix state change.
	  Original commit message from CVS:
	  * gst/flx/gstflxdec.c: (gst_flxdec_change_state):
	  Fix state change.

2005-11-16 11:02:24 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* gst/udp/gstudpsrc.c:
	  Move comment.
	  Original commit message from CVS:
	  (gst_udpsrc_create): Move comment.

2005-11-16 10:43:44 +0000  Andy Wingo <wingo@pobox.com>

	  gst/udp/gstudpsrc.c: Clean up with the boilerplate macro.
	  Original commit message from CVS:
	  2005-11-16  Andy Wingo  <wingo@pobox.com>
	  * gst/udp/gstudpsrc.c: Clean up with the boilerplate macro.

2005-11-15 19:41:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: When seeking, seek to closest index entry at or before the requested seek position, no...
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
	  When seeking, seek to closest index entry at or before the requested
	  seek position, not just the closest one (#321001).

2005-11-15 12:16:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Invert DIB images again (see #132341).
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (swap_line), (gst_avi_demux_invert),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
	  Invert DIB images again (see #132341).

2005-11-14 02:13:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* common:
	* configure.ac:
	* ext/aalib/gstaasink.c:
	* ext/cairo/gstcairo.c:
	* ext/dv/gstdv.c:
	* ext/esd/gstesd.c:
	* ext/flac/gstflac.c:
	* ext/gconf/gstgconfelements.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/jpeg/gstjpeg.c:
	* ext/ladspa/gstladspa.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libmng/gstmng.c:
	* ext/libpng/gstpng.c:
	* ext/mikmod/gstmikmod.c:
	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttimeoverlay.c:
	* ext/raw1394/gst1394.c:
	* ext/speex/gstspeex.c:
	* gst/alpha/Makefile.am:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautodetect.c:
	* gst/avi/gstavi.c:
	* gst/cutter/gstcutter.c:
	* gst/debug/efence.c:
	* gst/debug/gstdebug.c:
	* gst/debug/gstnavigationtest.c:
	* gst/effectv/gsteffectv.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/law/alaw.c:
	* gst/law/mulaw.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska.c:
	* gst/median/gstmedian.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multipart/multipart.c:
	* gst/oldcore/gstelements.c:
	* gst/rtp/Makefile.am:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtp.c:
	* gst/rtsp/gstrtsp.c:
	* gst/smoothwave/gstsmoothwave.c:
	* gst/smpte/gstsmpte.c:
	* gst/udp/gstudp.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/videomixer/videomixer.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* sys/oss/gstossaudio.c:
	* sys/osxaudio/gstosxaudio.c:
	  rework configure.ac; make asterisk rtp stuff compile on mingw
	  Original commit message from CVS:
	  rework configure.ac; make asterisk rtp stuff compile on mingw

2005-11-12 13:31:56 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/jpeg/gstjpegdec.c: Only GST_DEBUG() information on the valid components.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
	  Only GST_DEBUG() information on the valid components.

2005-11-11 19:34:50 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	  back to head
	  Original commit message from CVS:
	  back to head

=== release 0.9.5 ===

2005-11-11 19:33:23 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  releasing 0.9.5
	  Original commit message from CVS:
	  releasing 0.9.5

2005-11-11 18:33:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2005-11-11 16:48:58 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/avi/gstavidemux.*: Yeah, implement proper seeking. Exact seeking and segment seeking.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	  (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_event),
	  (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	  (gst_avi_demux_loop):
	  * gst/avi/gstavidemux.h:
	  Yeah, implement proper seeking. Exact seeking and segment seeking.
	  Still need to do some checks for segment_stop.

2005-11-11 15:17:44 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  fix Cairo entry
	  Original commit message from CVS:
	  fix Cairo entry

2005-11-10 12:34:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Add support for custom genre tags.
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
	  Add support for custom genre tags.

2005-11-10 12:22:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-mux.c: Don't try to ready buffer duration from buffer that we don't own any  longer and that mi...
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_write_data):
	  Don't try to ready buffer duration from buffer that we don't
	  own any  longer and that might already have been unreffed.
	  (#321136)

2005-11-09 21:35:29 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ChangeLog:
	* gst/flx/gstflxdec.c:
	  Attempting to optimize the code for embedded systems.
	  Original commit message from CVS:
	  Attempting to optimize the code for embedded systems.

2005-11-08 08:54:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/gstosssink.c: Don't re-use already closed file descriptor. (#320920)
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * sys/oss/gstosssink.c: (gst_oss_sink_close):
	  Don't re-use already closed file descriptor. (#320920)

2005-11-07 17:35:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/gstosssink.*: Cache probed caps; fix debug output for SET_PARAM macros.
	  Original commit message from CVS:
	  * sys/oss/gstosssink.c: (gst_oss_sink_dispose),
	  (gst_oss_sink_set_property), (gst_oss_sink_getcaps),
	  (gst_oss_sink_prepare):
	  * sys/oss/gstosssink.h:
	  Cache probed caps; fix debug output for SET_PARAM macros.

2005-11-07 15:09:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cairo/: Port cairo textoverlay plugin to 0.9. Add 'shaded-background' property and redo position. Doesn't handle ...
	  Original commit message from CVS:
	  * ext/cairo/Makefile.am:
	  * ext/cairo/gstcairo.c: (plugin_init):
	  * ext/cairo/gsttextoverlay.c: (gst_text_overlay_base_init),
	  (gst_text_overlay_class_init), (gst_text_overlay_finalize),
	  (gst_text_overlay_init), (gst_text_overlay_font_init),
	  (gst_text_overlay_set_property), (gst_text_overlay_render_text),
	  (gst_text_overlay_getcaps), (gst_text_overlay_setcaps),
	  (gst_text_overlay_text_pad_linked),
	  (gst_text_overlay_text_pad_unlinked), (gst_text_overlay_shade_y),
	  (gst_text_overlay_blit_1), (gst_text_overlay_blit_sub2x2),
	  (gst_text_overlay_push_frame), (gst_text_overlay_pop_video),
	  (gst_text_overlay_pop_text), (gst_text_overlay_collected),
	  (gst_text_overlay_change_state):
	  * ext/cairo/gsttextoverlay.h:
	  Port cairo textoverlay plugin to 0.9. Add 'shaded-background'
	  property and redo position. Doesn't handle upstream renegotiation
	  yet though.

2005-11-07 10:31:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: No need to take the STREAM_LOCK in the loop function. Improve some debug messages. Don't leak ...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	  (gst_avi_demux_loop):
	  No need to take the STREAM_LOCK in the loop function. Improve
	  some debug messages. Don't leak pad names in debug messages.

2005-11-07 10:27:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: Don't error out when the source pad isn't linked.
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_push_vorbis_codec_priv_data),
	  (gst_matroska_demux_add_wvpk_header):
	  Don't error out when the source pad isn't linked.

2005-11-02 19:42:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gconf/: Fix state change functions here as well and set kid to NULL state before removing it.
	  Original commit message from CVS:
	  * ext/gconf/gstgconfaudiosink.c: (do_toggle_element),
	  (gst_gconf_audio_sink_change_state):
	  * ext/gconf/gstgconfvideosink.c: (do_toggle_element),
	  (gst_gconf_video_sink_change_state):
	  Fix state change functions here as well and set kid
	  to NULL state before removing it.

2005-11-02 16:48:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* check/elements/matroskamux.c:
	* common:
	* tests/check/elements/matroskamux.c:
	  sigh, static pad templates aren't refcounted properly
	  Original commit message from CVS:
	  sigh, static pad templates aren't refcounted properly

2005-11-01 16:14:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* check/elements/.gitignore:
	* gst/level/.gitignore:
	* tests/check/elements/.gitignore:
	  ignore more
	  Original commit message from CVS:
	  ignore more

2005-11-01 15:15:44 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/wavenc/gstwavenc.c: Added proper event handlind, made downstream newsegment event use GST_FORMAT_BYTES (otherwise...
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file),
	  (gst_wavenc_init), (gst_wavenc_event), (gst_wavenc_chain):
	  Added proper event handlind,
	  made downstream newsegment event use GST_FORMAT_BYTES (otherwise it's
	  ignored),
	  and don't set a duration of 0 for buffers otherwise they are discarded
	  by GstBaseSink.
	  GstWavEnc needs some serious loving, after going through the code I'm
	  really wondering how this can stay in -good ...

2005-11-01 15:11:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  Fix leaks and invalid memory access as reported by valgrind
	  Original commit message from CVS:
	  * check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad),
	  (setup_matroskamux), (check_buffer_data), (GST_START_TEST):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_finalize),
	  (gst_matroska_mux_reset), (gst_matroska_mux_audio_pad_setcaps),
	  (gst_matroska_mux_start), (gst_matroska_mux_write_data),
	  (gst_matroska_mux_collected):
	  Fix leaks and invalid memory access as reported by valgrind

2005-11-01 14:41:01 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* check/elements/matroskamux.c:
	* tests/check/elements/matroskamux.c:
	  ... and add the missing file
	  Original commit message from CVS:
	  ... and add the missing file

2005-11-01 14:36:02 +0000  Michal Benes <michal.benes@xeris.cz>

	  add a unit test for matroskamux fix the bugs that the unit test exposed
	  Original commit message from CVS:
	  Patch by: Michal Benes <michal.benes@xeris.cz>
	  * check/Makefile.am:
	  * gst/matroska/ebml-write.c: (gst_ebml_write_seek):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_handle_src_event),
	  (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start):
	  add a unit test for matroskamux
	  fix the bugs that the unit test exposed

2005-11-01 14:34:22 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/rtp/Makefile.am:
	  fix Makefile.am
	  Original commit message from CVS:
	  fix Makefile.am

2005-11-01 12:39:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/autodetect/: Fix state change function and use GST_DEBUG_FUNCPTR in class_init.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c:
	  (gst_auto_audio_sink_class_init),
	  (gst_auto_audio_sink_change_state):
	  * gst/autodetect/gstautovideosink.c:
	  (gst_auto_video_sink_class_init),
	  (gst_auto_video_sink_change_state):
	  Fix state change function and use GST_DEBUG_FUNCPTR in
	  class_init.

2005-11-01 12:35:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Set timestamps on outgoing ebml headers as well, so that the element after matroskamux can get the tim...
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst/matroska/ebml-write.c: (gst_ebml_write_new),
	  (gst_ebml_write_reset), (gst_ebml_write_element_new):
	  * gst/matroska/ebml-write.h:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_write_data):
	  Set timestamps on outgoing ebml headers as well, so that the
	  element after matroskamux can get the timestamp already when
	  reading the first ebml element and doesn't have to wait for
	  the actual data buffer for that (#320308).

2005-10-31 22:08:52 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* gst/videomixer/videomixer.c:
	  gst/videomixer/videomixer.c (gst_videomixer_pad_unlink)
	  Original commit message from CVS:
	  2005-10-31  Andy Wingo  <wingo@pobox.com>
	  * gst/videomixer/videomixer.c (gst_videomixer_pad_unlink)
	  (gst_videomixer_pad_link): Kill some memleaks.
	  (gst_videomixer_pad_get_property): Style fix.
	  (gst_videomixer_pad_set_property): Style fix.
	  (gst_videomixer_pad_init): Style fix.
	  (gst_videomixer_update_queues): Kill memleak.
	  (gst_videomixer_loop): Kill memleak.
	  (gst_videomixer_collected): Kill memleak.

2005-10-31 19:08:27 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* gst/auparse/gstauparse.c:
	  Just some cleanup.
	  Original commit message from CVS:
	  Just some cleanup.

2005-10-31 14:41:31 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* ext/speex/gstspeexenc.c:
	  Add checks to GST_FLOW_NOT_LINKED for values returned from gst_pad_push.
	  Original commit message from CVS:
	  Add checks to GST_FLOW_NOT_LINKED for values returned from gst_pad_push.

2005-10-31 12:00:10 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ChangeLog:
	* gst/rtp/gstrtpg711dec.c:
	* gst/rtp/gstrtpg711depay.c:
	  Payloader now sets some default caps on the srcpad if caps on the sinkpad are never set. This is important for the g7...
	  Original commit message from CVS:
	  Payloader now sets some default caps on the srcpad if caps on the sinkpad are never set. This is important for the g711 to work with burger's rtpbin element.

2005-10-28 19:19:40 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* common:
	* ext/speex/gstspeexenc.c:
	  Add checks for return values from gst_pad_push and gst_pad_alloc_buffer.
	  Original commit message from CVS:
	  Add checks for return values from gst_pad_push and gst_pad_alloc_buffer.

2005-10-28 15:32:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Add SimpleBlock support to matroska demuxer and muxer (part of
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_init_stream),
	  (gst_matroska_demux_parse_info),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_parse_cluster):
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
	  (gst_matroska_mux_init), (gst_matroska_mux_start),
	  (gst_matroska_mux_create_buffer_header),
	  (gst_matroska_mux_write_data), (gst_matroska_mux_set_property),
	  (gst_matroska_mux_get_property):
	  * gst/matroska/matroska-mux.h:
	  Add SimpleBlock support to matroska demuxer and muxer (part of
	  Matroska v2). (#319731)

2005-10-28 13:24:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/jpeg/gstjpegdec.*: Cleanups. Don't create caps for every chain.
	  Original commit message from CVS:
	  * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init), (gst_jpeg_dec_chain),
	  (gst_jpeg_dec_change_state):
	  * ext/jpeg/gstjpegdec.h:
	  Cleanups. Don't create caps for every chain.

2005-10-27 18:46:32 +0000  Flavio Oliveira <flavio.oliveira@indt.org.br>

	* ChangeLog:
	* gst/law/alaw-encode.c:
	* gst/law/alaw-encode.h:
	* gst/law/mulaw-encode.c:
	* gst/law/mulaw-encode.h:
	  Fix to set timestamp on buffer, it was tested with RTP G711 elements.
	  Original commit message from CVS:
	  Fix to set timestamp on buffer, it was tested with RTP G711 elements.

2005-10-27 11:27:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.h: Remove got_redirect from class structure as well.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.h:
	  Remove got_redirect from class structure as well.

2005-10-27 11:25:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/qtdemux/qtdemux.c: Remove 'got-redirect' signal and post element message on the bus instead.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
	  (qtdemux_parse_tree):
	  Remove 'got-redirect' signal and post element message
	  on the bus instead.

2005-10-27 11:00:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/oss/gstosssrc.c: Set correct format on oss instead of a silly value.
	  Original commit message from CVS:
	  * sys/oss/gstosssrc.c: (gst_oss_src_prepare):
	  Set correct format on oss instead of a silly value.

2005-10-27 09:52:08 +0000  Julien Moutte <julien@moutte.net>

	  gst/videobox/gstvideobox.c: Use liboil for
	  Original commit message from CVS:
	  2005-10-27  Julien MOUTTE  <julien@moutte.net>
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	  (gst_video_box_transform_caps), (gst_video_box_set_caps),
	  (gst_video_box_get_unit_size), (gst_video_box_copy_plane_i420),
	  (gst_video_box_i420), (gst_video_box_ayuv): Use liboil for
	  I420 rendering as well, doesn't bring much for my platform.
	  Might help on some other platforms.

2005-10-26 21:47:36 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ChangeLog:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmparse.c:
	* gst/rtp/gstrtpgsmpay.c:
	  Declaring the padtemplate correctly.
	  Original commit message from CVS:
	  Declaring the padtemplate correctly.

2005-10-26 20:28:32 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ChangeLog:
	* gst/rtp/gstrtpg711dec.c:
	* gst/rtp/gstrtpg711depay.c:
	* gst/rtp/gstrtpg711enc.c:
	* gst/rtp/gstrtpg711pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmparse.c:
	* gst/rtp/gstrtpgsmpay.c:
	  Setting the proper copyright notice.
	  Original commit message from CVS:
	  Setting the proper copyright notice.

2005-10-26 17:23:06 +0000  Julien Moutte <julien@moutte.net>

	  gst/videobox/Makefile.am: Use liboil.
	  Original commit message from CVS:
	  2005-10-26  Julien MOUTTE  <julien@moutte.net>
	  * gst/videobox/Makefile.am: Use liboil.
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	  (gst_video_box_set_property), (gst_video_box_transform_caps),
	  (gst_video_box_set_caps), (gst_video_box_get_unit_size),
	  (gst_video_box_ayuv): Lot of optimization in AYUV rendering
	  using liboil. Will dot the same to I420 border generation
	  tomorrow.

2005-10-26 16:36:01 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/rtp/Makefile.am:
	  fix automake warnings
	  Original commit message from CVS:
	  fix automake warnings

2005-10-26 14:50:59 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ChangeLog:
	* gst/rtp/gstrtpg711dec.c:
	* gst/rtp/gstrtpg711dec.h:
	* gst/rtp/gstrtpg711depay.c:
	* gst/rtp/gstrtpg711depay.h:
	* gst/rtp/gstrtpg711enc.c:
	* gst/rtp/gstrtpg711pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmparse.c:
	* gst/rtp/gstrtpgsmparse.h:
	* gst/rtp/gstrtpgsmpay.c:
	  Hacked the G711 (de)payloader to try to make things right. rtpg711dec now inherits from the basertpdepayloader.
	  Original commit message from CVS:
	  Hacked the G711 (de)payloader to try to make things right. rtpg711dec now inherits from the basertpdepayloader.

2005-10-26 14:23:45 +0000  Julien Moutte <julien@moutte.net>

	  gst/videobox/gstvideobox.c: Removing this forgotten debug.
	  Original commit message from CVS:
	  2005-10-26  Julien MOUTTE  <julien@moutte.net>
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	  (gst_video_box_transform_caps), (gst_video_box_get_unit_size),
	  (gst_video_box_ayuv): Removing this forgotten debug.

2005-10-26 14:08:49 +0000  Julien Moutte <julien@moutte.net>

	  gst/videobox/gstvideobox.c: Fix the stride issue when boxing to AYUV.
	  Original commit message from CVS:
	  2005-10-26  Julien MOUTTE  <julien@moutte.net>
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	  (gst_video_box_transform_caps), (gst_video_box_get_unit_size),
	  (gst_video_box_ayuv): Fix the stride issue when boxing to AYUV.

2005-10-26 11:12:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/: Actually use the 'oss' debug category we register.
	  Original commit message from CVS:
	  * sys/oss/gstossaudio.c:
	  * sys/oss/gstossdmabuffer.c:
	  * sys/oss/gstosshelper.c:
	  * sys/oss/gstossmixer.c:
	  * sys/oss/gstossmixerelement.c:
	  * sys/oss/gstossmixertrack.c:
	  * sys/oss/gstosssink.c:
	  * sys/oss/gstosssrc.c:
	  Actually use the 'oss' debug category we register.

2005-10-26 10:38:18 +0000  Julien Moutte <julien@moutte.net>

	  gst/videomixer/videomixer.c: Use gst_pad_get_parent and drop the ref that was added through that call.
	  Original commit message from CVS:
	  2005-10-26  Julien MOUTTE  <julien@moutte.net>
	  * gst/videomixer/videomixer.c:
	  (gst_videomixer_pad_set_property),
	  (gst_videomixer_pad_sink_setcaps), (gst_videomixer_getcaps):
	  Use gst_pad_get_parent and drop the ref that was added through
	  that call.

2005-10-26 10:03:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmpay.c:
	  fix compilation
	  Original commit message from CVS:
	  fix compilation

2005-10-25 21:09:36 +0000  Flavio Oliveira <flavio.oliveira@indt.org.br>

	* ChangeLog:
	* gst/rtp/gstrtpg711dec.c:
	* gst/rtp/gstrtpg711depay.c:
	  Just removed a couple of lines of weird code used during development/test time.
	  Original commit message from CVS:
	  Just removed a couple of lines of weird code used during development/test time.

2005-10-25 19:19:38 +0000  Flavio Oliveira <flavio.oliveira@indt.org.br>

	* ChangeLog:
	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpg711dec.c:
	* gst/rtp/gstrtpg711dec.h:
	* gst/rtp/gstrtpg711depay.c:
	* gst/rtp/gstrtpg711depay.h:
	* gst/rtp/gstrtpg711enc.c:
	* gst/rtp/gstrtpg711enc.h:
	* gst/rtp/gstrtpg711pay.c:
	* gst/rtp/gstrtpg711pay.h:
	  G711 payloader and depayloader created by Edgard Lima (it supports mulaw and alaw (dec)encoders)
	  Original commit message from CVS:
	  G711 payloader and depayloader created by Edgard Lima (it supports
	  mulaw and alaw (dec)encoders)

2005-10-25 17:55:19 +0000  Julien Moutte <julien@moutte.net>

	  gst/videobox/gstvideobox.c: Doh ! I introduced wingo's bug again ! Sorry...
	  Original commit message from CVS:
	  2005-10-25  Julien MOUTTE  <julien@moutte.net>
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	  (gst_video_box_transform_caps), (gst_video_box_get_unit_size):
	  Doh ! I introduced wingo's bug again ! Sorry...

2005-10-25 16:02:38 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	* gst/rtp/Makefile.am:
	  add missing header files for disting
	  Original commit message from CVS:
	  add missing header files for disting

2005-10-25 15:07:02 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ChangeLog:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmenc.h:
	* gst/rtp/gstrtpgsmparse.c:
	* gst/rtp/gstrtpgsmparse.h:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgsmpay.h:
	  Getting the GSM (de)payloader working and compatible with our plans for RTP.
	  Original commit message from CVS:
	  Getting the GSM (de)payloader working and compatible with our plans for RTP.

2005-10-25 13:03:04 +0000  Christian Schaller <uraeus@gnome.org>

	* gst/rtp/gstrtp.c:
	  fix mistaken claim on GPL, its LGPL
	  Original commit message from CVS:
	  fix mistaken claim on GPL, its LGPL

2005-10-25 10:47:09 +0000  Julien Moutte <julien@moutte.net>

	  ext/libpng/gstpngdec.c: Push a newsegment event, move some redundant code in a single place.
	  Original commit message from CVS:
	  2005-10-25  Julien MOUTTE  <julien@moutte.net>
	  * ext/libpng/gstpngdec.c: (user_info_callback),
	  (gst_pngdec_caps_create_and_set), (gst_pngdec_task): Push
	  a newsegment event, move some redundant code in a single place.

2005-10-25 10:23:26 +0000  Julien Moutte <julien@moutte.net>

	  ext/libpng/gstpngdec.c: Temporary hack to get correct colors order when we have a png image with alpha channel.
	  Original commit message from CVS:
	  2005-10-25  Julien MOUTTE  <julien@moutte.net>
	  * ext/libpng/gstpngdec.c: (user_info_callback),
	  (gst_pngdec_caps_create_and_set), (gst_pngdec_task): Temporary
	  hack to get correct colors order when we have a png image with
	  alpha channel.

2005-10-24 17:29:02 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/dv/gstdvdemux.c: Call gst_element_no_more_pads when there will be no more pads.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads):
	  Call gst_element_no_more_pads when there will be no more pads.

2005-10-24 16:39:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added two new payloaders, an RFC 2190 payloader for h263 and a payload convertor for an asterisk server.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_get_type),
	  (gst_asteriskh263_base_init), (gst_asteriskh263_class_init),
	  (gst_asteriskh263_init), (gst_asteriskh263_finalize),
	  (gst_asteriskh263_chain), (gst_asteriskh263_set_property),
	  (gst_asteriskh263_get_property), (gst_asteriskh263_change_state),
	  (gst_asteriskh263_plugin_init):
	  * gst/rtp/gstasteriskh263.h:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtph263enc.c: (gst_rtph263enc_get_type),
	  (gst_rtph263enc_base_init), (gst_rtph263enc_class_init),
	  (gst_rtph263enc_init), (gst_rtph263enc_finalize),
	  (gst_rtph263enc_setcaps), (gst_rtph263enc_gobfiner),
	  (gst_rtph263enc_flush), (gst_rtph263enc_handle_buffer),
	  (gst_rtph263enc_plugin_init):
	  * gst/rtp/gstrtph263enc.h:
	  Added two new payloaders, an RFC 2190 payloader for h263 and
	  a payload convertor for an asterisk server.

2005-10-24 15:57:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/gstosssrc.c: Set bytes_per_sample correctly (is not always 4, but depends on width and number of channels).
	  Original commit message from CVS:
	  * sys/oss/gstosssrc.c: (gst_oss_src_prepare):
	  Set bytes_per_sample correctly (is not always 4, but
	  depends on width and number of channels).

2005-10-24 15:50:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacenc.*: Fix seeking, so that flacenc can rewrite the header with the correct duration and amount of sa...
	  Original commit message from CVS:
	  * ext/flac/gstflacenc.c: (gst_flacenc_base_init),
	  (gst_flacenc_init), (gst_flacenc_sink_setcaps),
	  (gst_flacenc_seek_callback), (gst_flacenc_write_callback),
	  (gst_flacenc_sink_event), (gst_flacenc_chain),
	  (gst_flacenc_set_property), (gst_flacenc_get_property),
	  (gst_flacenc_change_state):
	  * ext/flac/gstflacenc.h:
	  Fix seeking, so that flacenc can rewrite the header with the
	  correct duration and amount of samples and all that at EOS;
	  also set timestamps and granulepos on outgoing buffers; add
	  debug category; fix state change function.

2005-10-24 13:46:09 +0000  Julien Moutte <julien@moutte.net>

	  gst/videomixer/videomixer.c: Don't restrict video geometry from 16 to 4096.
	  Original commit message from CVS:
	  2005-10-24  Julien MOUTTE  <julien@moutte.net>
	  * gst/videomixer/videomixer.c: Don't restrict video geometry
	  from 16 to 4096.

2005-10-24 13:22:14 +0000  Julien Moutte <julien@moutte.net>

	  gst/videobox/gstvideobox.c: Fix caps negotiation correctly, add debugging category.
	  Original commit message from CVS:
	  2005-10-24  Julien MOUTTE  <julien@moutte.net>
	  * gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	  (gst_video_box_transform_caps), (gst_video_box_get_unit_size):
	  Fix caps negotiation correctly, add debugging category.

2005-10-24 13:02:47 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	* configure.ac:
	  port over plugin listing from base
	  Original commit message from CVS:
	  port over plugin listing from base

2005-10-24 08:59:24 +0000  Julien Moutte <julien@moutte.net>

	  ext/libpng/gstpngdec.c: Don't use fixed caps on a sink pad.
	  Original commit message from CVS:
	  2005-10-24  Julien MOUTTE  <julien@moutte.net>
	  * ext/libpng/gstpngdec.c: (gst_pngdec_init): Don't use fixed
	  caps on
	  a sink pad.

2005-10-23 23:05:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	* docs/upload.mak:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.9.4 ===

2005-10-23 22:43:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  releasing 0.9.4
	  Original commit message from CVS:
	  releasing 0.9.4

2005-10-23 11:07:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/libpng/gstpngdec.c:
	* gst/wavparse/gstwavparse.c:
	* po/POTFILES.in:
	  STOPPED->FAILED
	  Original commit message from CVS:
	  STOPPED->FAILED

2005-10-21 17:00:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/speex/gstspeexenc.c: Add position and duration query, fix query type function.
	  Original commit message from CVS:
	  * ext/speex/gstspeexenc.c: (gst_speexenc_get_query_types),
	  (gst_speexenc_src_query):
	  Add position and duration query, fix query type function.
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	  (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
	  Let's not set non-fixed caps on source pads.

2005-10-21 16:15:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Set correct stream_time in newsegment event. avi can also handle a duration query now.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_get_src_query_types),
	  (gst_avi_demux_handle_seek):
	  Set correct stream_time in newsegment event.
	  avi can also handle a duration query now.

2005-10-21 10:06:40 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  update for latest additions
	  Original commit message from CVS:
	  update for latest additions

2005-10-20 19:14:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-demux.c: Fix duration query; fix basetime in newsegment event after seek; fix duration in initi...
	  Original commit message from CVS:
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_handle_src_query),
	  (gst_matroska_demux_handle_seek_event),
	  (gst_matroska_demux_loop_stream_parse_id):
	  Fix duration query; fix basetime in newsegment event after
	  seek; fix duration in initial newsegment event.
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start):
	  Extract number of channels and samplerate from vorbis headers;
	  add some debug messages when querying the durations of the
	  input streams.

2005-10-20 11:50:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Set stream time correctly in newsegment.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data),
	  (gst_wavparse_pad_convert), (gst_wavparse_srcpad_event):
	  Set stream time correctly in newsegment.

2005-10-20 11:39:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/avi/gstavidemux.c: Correctly fill in the stream time.
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
	  Correctly fill in the stream time.

2005-10-19 20:48:24 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* check/elements/level.c:
	* gst/level/gstlevel.c:
	* gst/level/level-example.c:
	* tests/check/elements/level.c:
	  use ELEMENT messages instead
	  Original commit message from CVS:
	  use ELEMENT messages instead

2005-10-19 15:58:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/: API change fix.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types),
	  (gst_qtdemux_handle_src_query):
	  * gst/speed/gstspeed.c: (speed_get_query_types), (speed_src_query):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_src_event),
	  (gst_tta_parse_get_query_types), (gst_tta_parse_query):
	  API change fix.

2005-10-19 15:57:04 +0000  Wim Taymans <wim.taymans@gmail.com>

	  API change fix.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_get_src_query_types),
	  (gst_dvdemux_src_query):
	  * ext/flac/gstflacdec.c: (gst_flacdec_length),
	  (gst_flacdec_src_query):
	  * ext/raw1394/gstdv1394src.c: (gst_dv1394src_query):
	  * ext/speex/gstspeexdec.c: (speex_dec_src_query):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
	  * gst/debug/gstnavseek.c: (gst_navseek_seek):
	  * gst/debug/progressreport.c: (gst_progress_report_report):
	  * gst/matroska/ebml-read.c: (gst_ebml_read_get_length):
	  * gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_handle_src_query):
	  * gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data),
	  (gst_wavparse_pad_convert), (gst_wavparse_pad_query),
	  (gst_wavparse_srcpad_event):
	  API change fix.

2005-10-19 10:57:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/goom/: Make inline functions either 'static inline' or 'extern inline', otherwise the Forte compiler apparently w...
	  Original commit message from CVS:
	  * gst/goom/filters.c:
	  * gst/goom/graphic.h:
	  * gst/goom/lines.c:
	  Make inline functions either 'static inline' or 'extern inline',
	  otherwise the Forte compiler apparently won't inline them (#317300).

2005-10-18 22:50:11 +0000  Julien Moutte <julien@moutte.net>

	  ext/libpng/gstpngdec.c: forgot the buffer unref in pull.
	  Original commit message from CVS:
	  2005-10-19  Julien MOUTTE  <julien@moutte.net>
	  * ext/libpng/gstpngdec.c: forgot the buffer unref in pull.

2005-10-18 22:44:11 +0000  Julien Moutte <julien@moutte.net>

	  ext/libpng/gstpngdec.*: Complete rewrite of pngdec. It's now very nice and handle push/pull based model. if you have ...
	  Original commit message from CVS:
	  2005-10-19  Julien MOUTTE  <julien@moutte.net>
	  * ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
	  (gst_pngdec_init), (user_error_fn), (user_warning_fn),
	  (user_info_callback), (user_endrow_callback),
	  (user_end_callback),
	  (user_read_data), (gst_pngdec_caps_create_and_set),
	  (gst_pngdec_task), (gst_pngdec_chain), (gst_pngdec_sink_event),
	  (gst_pngdec_libpng_clear), (gst_pngdec_libpng_init),
	  (gst_pngdec_change_state), (gst_pngdec_sink_activate_push),
	  (gst_pngdec_sink_activate_pull), (gst_pngdec_sink_activate):
	  * ext/libpng/gstpngdec.h: Complete rewrite of pngdec. It's now
	  very nice and handle push/pull based model. if you have filesrc
	  connected to it, it will do random access to load the png file.
	  If you have a network source that can't do _getrange, it does
	  progressive loading through the chain function.
	  * gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps),
	  (transform_rgb), (transform_bgr): Fix caps negotiation correctly
	  thanks to Master Wim Taymans ;-)

2005-10-18 18:12:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/: Ported matroska demuxer to 0.9.
	  Original commit message from CVS:
	  * gst/matroska/Makefile.am:
	  * gst/matroska/ebml-read.c:
	  * gst/matroska/ebml-read.h:
	  * gst/matroska/matroska-demux.c:
	  * gst/matroska/matroska-demux.h:
	  * gst/matroska/matroska.c: (plugin_init):
	  Ported matroska demuxer to 0.9.

2005-10-18 18:06:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/matroska/matroska-mux.c: Fix mpeg4 input handling (#318847); also, while we're at it, fix media type for Motion-J...
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst/matroska/matroska-mux.c:
	  (gst_matroska_mux_video_pad_setcaps),
	  (gst_matroska_mux_audio_pad_setcaps):
	  Fix mpeg4 input handling (#318847); also, while we're at it,
	  fix media type for Motion-JPEG: should be image/jpeg.

2005-10-18 13:21:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.c: Fix for segment-start/stop API change.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data),
	  (gst_wavparse_pad_convert), (gst_wavparse_srcpad_event):
	  Fix for segment-start/stop API change.

2005-10-17 17:18:56 +0000  Julien Moutte <julien@moutte.net>

	  gst/alpha/gstalphacolor.c: Handle caps negotiation in a better way.
	  Original commit message from CVS:
	  2005-10-17  Julien MOUTTE  <julien@moutte.net>
	  * gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps),
	  (transform_rgb), (transform_bgr): Handle caps negotiation in a
	  better
	  way.

2005-10-17 16:59:20 +0000  Julien Moutte <julien@moutte.net>

	  gst/videobox/gstvideobox.c: Fix caps nego some more to get
	  Original commit message from CVS:
	  2005-10-17  Julien MOUTTE  <julien@moutte.net>
	  * gst/videobox/gstvideobox.c: (gst_video_box_transform_caps),
	  (gst_video_box_get_unit_size): Fix caps nego some more to get
	  AYUV
	  output declared in transform_caps.

2005-10-17 15:23:24 +0000  Julien Moutte <julien@moutte.net>

	  ext/libpng/gstpngdec.c: We use fixed caps.
	  Original commit message from CVS:
	  2005-10-17  Julien MOUTTE  <julien@moutte.net>
	  * ext/libpng/gstpngdec.c: (gst_pngdec_init): We use fixed caps.

2005-10-17 15:14:29 +0000  Julien Moutte <julien@moutte.net>

	  gst/videobox/gstvideobox.c: Fix wrong size calculations and implement get_unit_size correctly.
	  Original commit message from CVS:
	  2005-10-17  Julien MOUTTE  <julien@moutte.net>
	  * gst/videobox/gstvideobox.c: (gst_video_box_transform_caps),
	  (gst_video_box_get_unit_size): Fix wrong size calculations and
	  implement get_unit_size correctly.

2005-10-17 14:56:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Enable flx plugin.
	  Original commit message from CVS:
	  * configure.ac:
	  Enable flx plugin.
	  * gst/flx/gstflxdec.c: (flx_decode_chunks):
	  Fix gcc4 signedness issue.

2005-10-17 08:46:30 +0000  Julien Moutte <julien@moutte.net>

	  configure.ac: Adding videomixer.
	  Original commit message from CVS:
	  2005-10-17  Julien MOUTTE  <julien@moutte.net>
	  * configure.ac: Adding videomixer.
	  * ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
	  (user_read_data), (gst_pngdec_chain): More debugging.
	  * gst/alpha/Makefile.am: Adding alphacolor
	  * gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
	  (gst_alpha_color_class_init), (gst_alpha_color_init),
	  (gst_alpha_color_transform_caps), (gst_alpha_color_set_caps),
	  (transform_rgb), (transform_bgr),
	  (gst_alpha_color_transform_ip),
	  (plugin_init): Ported to 0.9 using in place base tranform.
	  * gst/videomixer/Makefile.am:
	  * gst/videomixer/videomixer.c: (gst_videomixer_pad_get_type),
	  (gst_videomixer_pad_class_init),
	  (gst_videomixer_pad_sink_setcaps),
	  (gst_videomixer_pad_link), (gst_videomixer_pad_unlink),
	  (gst_videomixer_pad_init), (gst_videomixer_class_init),
	  (gst_videomixer_init), (gst_videomixer_getcaps),
	  (gst_videomixer_request_new_pad), (gst_videomixer_fill_queues),
	  (gst_videomixer_blend_buffers), (gst_videomixer_update_queues),
	  (gst_videomixer_collected), (gst_videomixer_change_state):
	  Ported
	  to 0.9 using collectpads.

2005-10-16 21:19:44 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ChangeLog:
	* common:
	* configure.ac:
	* gst/flx/Makefile.am:
	* gst/flx/gstflxdec.c:
	* gst/flx/gstflxdec.h:
	  flx plugin ported to 0.9
	  Original commit message from CVS:
	  flx plugin ported to 0.9

2005-10-16 14:33:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* ext/shout2/gstshout2.c:
	  use gst_version_string
	  Original commit message from CVS:
	  use gst_version_string

2005-10-16 13:17:11 +0000  Andy Wingo <wingo@pobox.com>

	  configure.ac: GLIB_CHECK.
	  Original commit message from CVS:
	  2005-10-16  Andy Wingo  <wingo@pobox.com>
	  * configure.ac: GLIB_CHECK.

2005-10-15 16:48:55 +0000  Julien Moutte <julien@moutte.net>

	  ext/libpng/: Ported pngdec to 0.9
	  Original commit message from CVS:
	  2005-10-15  Julien MOUTTE  <julien@moutte.net>
	  * ext/libpng/Makefile.am:
	  * ext/libpng/gstpng.c: (plugin_init):
	  * ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
	  (gst_pngdec_init), (user_read_data), (gst_pngdec_chain):
	  * ext/libpng/gstpngdec.h: Ported pngdec to 0.9

2005-10-14 12:43:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Port matroska muxer to 0.9 (#318847).
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * configure.ac:
	  * gst/matroska/Makefile.am:
	  * gst/matroska/ebml-ids.h:
	  * gst/matroska/ebml-write.c:
	  * gst/matroska/ebml-write.h:
	  * gst/matroska/matroska-ids.h:
	  * gst/matroska/matroska-mux.c:
	  * gst/matroska/matroska-mux.h:
	  * gst/matroska/matroska.c: (plugin_init):
	  Port matroska muxer to 0.9 (#318847).

2005-10-13 18:59:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/speex/gstspeexenc.c: Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE; use GST_READ_UINT32_LE() and fr...
	  Original commit message from CVS:
	  * ext/speex/gstspeexenc.c: (gst_speexenc_get_tag_value),
	  (comment_init), (comment_add):
	  Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE;
	  use GST_READ_UINT32_LE() and friends rather than the private
	  implementation of those same macros.

2005-10-13 16:01:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/cairo/Makefile.am:
	  fix dist
	  Original commit message from CVS:
	  fix dist

2005-10-13 15:28:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  examples/stats/mp2ogg.c: more typo fixes
	  Original commit message from CVS:
	  * examples/stats/mp2ogg.c:
	  more typo fixes

2005-10-12 14:30:36 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition
	  Original commit message from CVS:
	  * examples/indexing/indexmpeg.c: (main):
	  * ext/a52dec/gsta52dec.c: (gst_a52dec_init):
	  * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_is_open),
	  (dvdnavsrc_set_property), (dvdnavsrc_open), (dvdnavsrc_close),
	  (dvdnavsrc_event), (dvdnavsrc_convert), (dvdnavsrc_query):
	  * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_set_property),
	  (dvdreadsrc_srcpad_query), (dvdreadsrc_get),
	  (dvdreadsrc_open_file), (dvdreadsrc_close_file):
	  * ext/dvdread/dvdreadsrc.h:
	  * ext/lame/gstlame.h:
	  * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init):
	  * gst/asfdemux/gstasfmux.c: (gst_asfmux_init):
	  * gst/iec958/ac3iec.h:
	  * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_init):
	  * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_init):
	  * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init):
	  * gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_init):
	  * gst/synaesthesia/gstsynaesthesia.c: (gst_synaesthesia_init):
	  renamed GST_FLAGS macros to GST_OBJECT_FLAGS
	  moved bitshift from macro to enum definition

2005-10-12 14:29:55 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition
	  Original commit message from CVS:
	  * examples/indexing/indexmpeg.c: (main):
	  * ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio),
	  (gst_artsdsink_close_audio), (gst_artsdsink_change_state):
	  * ext/artsd/gstartsdsink.h:
	  * ext/audiofile/gstafparse.c: (gst_afparse_open_file),
	  (gst_afparse_close_file):
	  * ext/audiofile/gstafparse.h:
	  * ext/audiofile/gstafsink.c: (gst_afsink_open_file),
	  (gst_afsink_close_file), (gst_afsink_chain),
	  (gst_afsink_change_state):
	  * ext/audiofile/gstafsink.h:
	  * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
	  (gst_afsrc_close_file), (gst_afsrc_change_state):
	  * ext/audiofile/gstafsrc.h:
	  * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init):
	  * ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init):
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_init):
	  * ext/jack/gstjack.h:
	  * ext/jack/gstjackbin.c: (gst_jack_bin_init),
	  (gst_jack_bin_change_state):
	  * ext/musepack/gstmusepackdec.c: (gst_musepackdec_init):
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init):
	  * ext/nas/nassink.c: (gst_nassink_open_audio),
	  (gst_nassink_close_audio), (gst_nassink_change_state):
	  * ext/nas/nassink.h:
	  * ext/polyp/polypsink.c: (gst_polypsink_init):
	  * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state):
	  * ext/sdl/sdlvideosink.h:
	  * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init):
	  * ext/sndfile/gstsf.c: (gst_sf_set_property),
	  (gst_sf_change_state), (gst_sf_release_request_pad),
	  (gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop):
	  * ext/sndfile/gstsf.h:
	  * ext/swfdec/gstswfdec.c: (gst_swfdec_init):
	  * ext/tarkin/gsttarkindec.c: (gst_tarkindec_init):
	  * gst/apetag/apedemux.c: (gst_ape_demux_init):
	  * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init):
	  * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init):
	  * gst/festival/gstfestival.c: (gst_festival_change_state):
	  * gst/festival/gstfestival.h:
	  * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
	  * gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init),
	  (gst_multifilesink_set_location), (gst_multifilesink_open_file),
	  (gst_multifilesink_close_file), (gst_multifilesink_next_file),
	  (gst_multifilesink_pad_query), (gst_multifilesink_handle_event),
	  (gst_multifilesink_chain), (gst_multifilesink_change_state):
	  * gst/multifilesink/gstmultifilesink.h:
	  * gst/videodrop/gstvideodrop.c: (gst_videodrop_init):
	  * sys/cdrom/gstcdplayer.c: (cdplayer_init):
	  * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init),
	  (dxr3audiosink_open), (dxr3audiosink_close),
	  (dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3),
	  (dxr3audiosink_change_state):
	  * sys/dxr3/dxr3audiosink.h:
	  * sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open),
	  (dxr3spusink_close), (dxr3spusink_chain),
	  (dxr3spusink_change_state):
	  * sys/dxr3/dxr3spusink.h:
	  * sys/dxr3/dxr3videosink.c: (dxr3videosink_init),
	  (dxr3videosink_open), (dxr3videosink_close),
	  (dxr3videosink_write_data), (dxr3videosink_change_state):
	  * sys/dxr3/dxr3videosink.h:
	  * sys/glsink/glimagesink.c: (gst_glimagesink_init):
	  * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state),
	  (gst_qcamsrc_open), (gst_qcamsrc_close):
	  * sys/qcam/gstqcamsrc.h:
	  * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  * sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get),
	  (gst_vcdsrc_open_file), (gst_vcdsrc_close_file),
	  (gst_vcdsrc_change_state), (gst_vcdsrc_recalculate):
	  * sys/vcd/vcdsrc.h:
	  renamed GST_FLAGS macros to GST_OBJECT_FLAGS
	  moved bitshift from macro to enum definition

2005-10-12 14:29:43 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition
	  Original commit message from CVS:
	  * examples/indexing/indexmpeg.c: (main):
	  * ext/esd/esdmon.c: (gst_esdmon_open_audio),
	  (gst_esdmon_close_audio), (gst_esdmon_change_state):
	  * ext/esd/esdmon.h:
	  * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init):
	  * ext/pango/gsttextoverlay.c: (gst_textoverlay_init):
	  * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_init):
	  * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_init):
	  * gst/avi/gstavimux.c: (gst_avimux_init):
	  * gst/matroska/matroska-demux.c: (gst_matroska_demux_init):
	  * gst/multipart/multipartdemux.c: (gst_multipart_demux_init):
	  * gst/multipart/multipartmux.c: (gst_multipart_mux_init):
	  * gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init),
	  (gst_multifilesrc_get), (gst_multifilesrc_open_file),
	  (gst_multifilesrc_close_file), (gst_multifilesrc_change_state):
	  * gst/oldcore/gstmultifilesrc.h:
	  * gst/oldcore/gstpipefilter.c: (gst_pipefilter_init),
	  (gst_pipefilter_open_file), (gst_pipefilter_close_file),
	  (gst_pipefilter_change_state):
	  * gst/oldcore/gstpipefilter.h:
	  * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_init):
	  * gst/videomixer/videomixer.c: (gst_videomixer_init):
	  * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_init):
	  * sys/osxaudio/gstosxaudiosink.h:
	  * sys/osxaudio/gstosxaudiosrc.h:
	  renamed GST_FLAGS macros to GST_OBJECT_FLAGS
	  moved bitshift from macro to enum definition

2005-10-12 03:14:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/Makefile.am:
	  dist cairo
	  Original commit message from CVS:
	  dist cairo

2005-10-12 03:12:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/: update of cairo-based timeoverlay to 1.0 Cairo API doesn't work yet for resizing of output sink
	  Original commit message from CVS:
	  * ext/Makefile.am:
	  * ext/cairo/Makefile.am:
	  * ext/cairo/gstcairo.c: (plugin_init):
	  * ext/cairo/gsttextoverlay.c: (gst_textoverlay_change_state):
	  * ext/cairo/gsttimeoverlay.c: (gst_timeoverlay_update_font_height),
	  (gst_timeoverlay_setup), (gst_timeoverlay_planar411):
	  * ext/cairo/gsttimeoverlay.h:
	  update of cairo-based timeoverlay to 1.0 Cairo API
	  doesn't work yet for resizing of output sink

2005-10-12 03:07:26 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  don't build checks if we don't have check
	  Original commit message from CVS:
	  don't build checks if we don't have check

2005-10-12 03:03:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	* common:
	  don't build checks if we don't have gstcheck
	  Original commit message from CVS:
	  don't build checks if we don't have gstcheck

2005-10-11 17:38:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/speex/gstspeexdec.c: newsegment API fix.
	  Original commit message from CVS:
	  * ext/speex/gstspeexdec.c: (speex_dec_event), (speex_dec_chain):
	  newsegment API fix.

2005-10-11 16:34:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/: newsegment API update.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
	  * gst/tta/gstttaparse.c: (gst_tta_parse_src_event),
	  (gst_tta_parse_parse_header):
	  newsegment API update.

2005-10-11 16:33:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  newsegment API update.
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event),
	  (gst_dvdemux_demux_frame):
	  * ext/flac/gstflacdec.c: (gst_flacdec_write):
	  * gst/auparse/gstauparse.c: (gst_auparse_chain):
	  * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header),
	  (gst_avi_demux_handle_seek):
	  * gst/goom/gstgoom.c: (gst_goom_event):
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file):
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data),
	  (gst_wavparse_loop), (gst_wavparse_pad_convert),
	  (gst_wavparse_srcpad_event):
	  newsegment API update.

2005-10-11 10:07:35 +0000  Andy Wingo <wingo@pobox.com>

	  ext/speex/gstspeexenc.c: Signedness cleanups.
	  Original commit message from CVS:
	  2005-10-11  Andy Wingo  <wingo@pobox.com>
	  * ext/speex/gstspeexenc.c: Signedness cleanups.

2005-10-10 19:57:40 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* PORTED_09:
	* ext/speex/Makefile.am:
	* ext/speex/gstspeex.c:
	* ext/speex/gstspeexenc.c:
	  Speexenc ported to 0.9.
	  Original commit message from CVS:
	  Speexenc ported to 0.9.

2005-10-10 14:16:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/oss/: Cleanups, make device configurable in the sink, handle and report errors.
	  Original commit message from CVS:
	  * sys/oss/gstosssink.c: (gst_oss_sink_class_init),
	  (gst_oss_sink_init), (gst_oss_sink_set_property),
	  (gst_oss_sink_get_property), (gst_oss_sink_open),
	  (gst_oss_sink_prepare), (gst_oss_sink_reset):
	  * sys/oss/gstosssink.h:
	  * sys/oss/gstosssrc.c: (gst_oss_src_class_init),
	  (gst_oss_src_set_property), (gst_oss_src_init), (gst_oss_src_open),
	  (gst_oss_src_prepare):
	  Cleanups, make device configurable in the sink, handle and report
	  errors.

2005-10-10 12:31:07 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/gconf/: Make sure element is NULL before removing from the bin.
	  Original commit message from CVS:
	  * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset):
	  * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset):
	  Make sure element is NULL before removing from the bin.

2005-10-07 16:28:24 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/raw1394/gstdv1394src.c:
	  Don't unref the message.
	  Original commit message from CVS:
	  (gst_dv1394src_bus_reset): Don't unref the message.

2005-10-07 16:22:59 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/raw1394/gstdv1394src.c:
	  Post a message when the cable is unplugged.
	  Original commit message from CVS:
	  (gst_dv1394src_bus_reset): Post a message when the cable is
	  unplugged.
	  (gst_dv1394src_create, gst_dv1394src_unlock): Remove some prints.

2005-10-07 15:24:24 +0000  Andy Wingo <wingo@pobox.com>

	  ext/raw1394/gstdv1394src.c: Make interruptible, so it won't block forever in a read().
	  Original commit message from CVS:
	  2005-10-07  Andy Wingo  <wingo@pobox.com>
	  * ext/raw1394/gstdv1394src.c: Make interruptible, so it won't
	  block forever in a read().

2005-10-07 13:17:53 +0000  Andy Wingo <wingo@pobox.com>

	  ext/raw1394/gstdv1394src.c: Clean up for style before doing some hacking. The only change should be that the state ch...
	  Original commit message from CVS:
	  2005-10-07  Andy Wingo  <wingo@pobox.com>
	  * ext/raw1394/gstdv1394src.c: Clean up for style before doing some
	  hacking. The only change should be that the state change stuff was
	  put into basesrc's start() and stop() routines, which coalesces
	  some steps.

2005-10-07 11:30:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Add check for mmap
	  Original commit message from CVS:
	  * configure.ac:
	  Add check for mmap
	  * gst/debug/Makefile.am:
	  Only compile efence plugin on systems that have mmap.

2005-10-05 16:36:57 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add latest files
	  Original commit message from CVS:
	  add latest files

2005-10-05 11:38:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/debug/: Port progressreport, navseek, navigationtest, testsink and breakmydata.
	  Original commit message from CVS:
	  * gst/debug/Makefile.am:
	  * gst/debug/breakmydata.c:
	  * gst/debug/gstdebug.c:
	  * gst/debug/gstnavigationtest.c:
	  * gst/debug/gstnavseek.c:
	  * gst/debug/gstnavseek.h:
	  * gst/debug/progressreport.c:
	  * gst/debug/testplugin.c:
	  Port progressreport, navseek, navigationtest, testsink and
	  breakmydata.

2005-10-05 11:15:23 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/dv/gstdvdemux.c: Fixes for better conversion
	  Original commit message from CVS:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_src_convert),
	  (gst_dvdemux_src_query):
	  Fixes for better conversion

2005-10-04 17:58:40 +0000  Michael Smith <msmith@xiph.org>

	  gst/autodetect/: Set state of elements to NULL before removing from bins.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset),
	  (gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect):
	  * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
	  (gst_auto_video_sink_find_best), (gst_auto_video_sink_detect):
	  Set state of elements to NULL before removing from bins.
	  Set state of test element to NULL if we failed to move it to READY

2005-10-04 17:44:43 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/dv/: Added DEFAULT <==> BYTES, TIME conversions on srcpad,
	  Original commit message from CVS:
	  * ext/dv/Makefile.am:
	  * ext/dv/gstdvdemux.c: (gst_dvdemux_src_query), (gst_dvdemux_src_conver):
	  Added DEFAULT <==> BYTES, TIME conversions on srcpad,
	  Corrected the query function for position so it doesn't forget what
	  format was asked, and calls the conversion functions on the correct pad.

2005-10-03 17:59:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	  back to head
	  Original commit message from CVS:
	  back to head

=== release 0.9.3 ===

2005-10-03 17:48:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* README:
	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  release time
	  Original commit message from CVS:
	  release time

2005-10-02 23:08:35 +0000  Andy Wingo <wingo@pobox.com>

	  ext/flac/gstflacdec.c (gst_flacdec_write): Deal with pad_alloc error returns.
	  Original commit message from CVS:
	  2005-10-03  Andy Wingo  <wingo@pobox.com>
	  * ext/flac/gstflacdec.c (gst_flacdec_write): Deal with pad_alloc
	  error returns.

2005-10-02 15:33:14 +0000  Andy Wingo <wingo@pobox.com>

	  configure.ac (GST_PLUGIN_LDFLAGS): Change to be like -base.
	  Original commit message from CVS:
	  2005-10-02  Andy Wingo  <wingo@pobox.com>
	  * configure.ac (GST_PLUGIN_LDFLAGS): Change to be like -base.
	  * ext/flac/gstflacenc.c: Ported to 0.9.
	  * ext/flac/gstflacdec.c (gst_flacdec_loop): Handle errors better.
	  * ext/flac/Makefile.am: Add the GST_PLUGINS_BASE cflags and libs,
	  and link to gsttagedit. Enable flacenc.
	  * ext/flac/gstflacdec.c: Re-enable tag reading.

2005-09-30 16:36:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Various class and caps fixes from Andre Magalhaes (andrunko)
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps):
	  * gst/rtp/gstrtpgsmparse.c:
	  * gst/rtp/gstrtph263penc.c:
	  * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
	  (gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
	  (gst_rtpmp4venc_set_property):
	  * gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
	  Various class and caps fixes from Andre Magalhaes (andrunko)

2005-09-29 13:08:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/level/level-example.c: Update for new bus API.
	  Original commit message from CVS:
	  * gst/level/level-example.c: (main):
	  Update for new bus API.

2005-09-28 13:38:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/qtdemux/qtdemux.c: No need to take stream lock here.
	  Original commit message from CVS:
	  * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
	  No need to take stream lock here.

2005-09-28 09:45:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Fix unexpanded autoconf macro GST_DOC, which has been renamed to GST_DOCBOOK_CHECK (see common/m4/gst-d...
	  Original commit message from CVS:
	  * configure.ac:
	  Fix unexpanded autoconf macro GST_DOC, which has been renamed
	  to GST_DOCBOOK_CHECK (see common/m4/gst-doc.m4) (#316202).

2005-09-27 15:12:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/oss/gstosssink.c: Fix playback of mono streams (bytes_per_sample should be set from the sample width and the numb...
	  Original commit message from CVS:
	  * sys/oss/gstosssink.c: (gst_oss_sink_prepare):
	  Fix playback of mono streams (bytes_per_sample should be set
	  from the sample width and the number of channels negotiated,
	  and not just be set to 4) (#317338)

2005-09-26 14:59:10 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  add auparse to plugins list
	  Original commit message from CVS:
	  add auparse to plugins list

2005-09-26 14:42:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmpaenc.c: Set buffer duration correctly.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_flush),
	  (gst_rtpmpaenc_handle_buffer):
	  Set buffer duration correctly.

2005-09-26 13:06:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/avi/gstavidemux.c: Don't crash when encountering a stream with an unknown fourcc or codec id. Instead, create a p...
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
	  (gst_avi_demux_class_init), (gst_avi_demux_parse_stream),
	  (gst_avi_demux_change_state):
	  Don't crash when encountering a stream with an unknown fourcc or
	  codec id. Instead, create a pad of type video/x-avi-unknown or
	  audio/x-avi-unknown, which as a side-effect also results in less
	  confusing error messages in players ('no decoder' vs. 'no streams');
	  minor fixes to state change function and class_init function.

2005-09-24 13:34:46 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* check/Makefile.am:
	* tests/check/Makefile.am:
	  set up plugin paths properly
	  Original commit message from CVS:
	  set up plugin paths properly

2005-09-24 13:10:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/autodetect/: These are sinks.
	  Original commit message from CVS:
	  * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_init):
	  * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_init):
	  These are sinks.

2005-09-24 12:10:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  check/elements/level.c: fix test for new GstClockTime use
	  Original commit message from CVS:
	  * check/elements/level.c: (GST_START_TEST):
	  fix test for new GstClockTime use
	  * gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
	  (gst_level_transform_ip):
	  * gst/level/gstlevel.h:
	  fix up the decay peak, ensuring the decay peak is never lower
	  than the peak for that interval

2005-09-23 18:23:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* gst/level/gstlevel.c:
	  updating docs
	  Original commit message from CVS:
	  updating docs

2005-09-23 18:15:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* Makefile.am:
	* check/elements/level.c:
	* common:
	* gst/level/Makefile.am:
	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	* gst/level/level-example.c:
	* tests/check/elements/level.c:
	  convert to using GstClockTime for all time values, finally.
	  Original commit message from CVS:
	  convert to using GstClockTime for all time values, finally.

2005-09-23 15:01:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/goom/Makefile.am:
	  fix build of goom
	  Original commit message from CVS:
	  fix build of goom

2005-09-23 14:20:01 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* gst/level/gstlevel.c:
	  we handle more than two channels
	  Original commit message from CVS:
	  we handle more than two channels

2005-09-23 04:23:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	* ext/cairo/Makefile.am:
	* ext/dv/Makefile.am:
	* ext/esd/Makefile.am:
	* ext/flac/Makefile.am:
	* ext/gconf/Makefile.am:
	* ext/gdk_pixbuf/Makefile.am:
	* ext/jpeg/Makefile.am:
	* ext/ladspa/Makefile.am:
	* ext/libcaca/Makefile.am:
	* ext/libmng/Makefile.am:
	* ext/libpng/Makefile.am:
	* ext/mikmod/Makefile.am:
	* ext/pango/Makefile.am:
	* ext/raw1394/Makefile.am:
	* ext/shout2/Makefile.am:
	* ext/speex/Makefile.am:
	* gst/alpha/Makefile.am:
	* gst/auparse/Makefile.am:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/Makefile.am:
	* gst/avi/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debug/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/flx/Makefile.am:
	* gst/goom/Makefile.am:
	* gst/law/Makefile.am:
	* gst/matroska/Makefile.am:
	* gst/median/Makefile.am:
	* gst/monoscope/Makefile.am:
	* gst/multipart/Makefile.am:
	* gst/oldcore/Makefile.am:
	* gst/rtp/Makefile.am:
	* gst/rtsp/Makefile.am:
	* gst/smoothwave/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/videobox/Makefile.am:
	* gst/videofilter/Makefile.am:
	* gst/videomixer/Makefile.am:
	* gst/wavenc/Makefile.am:
	* gst/wavparse/Makefile.am:
	* sys/oss/Makefile.am:
	* sys/osxaudio/Makefile.am:
	  fix build and use of GST_LIBS
	  Original commit message from CVS:
	  fix build and use of GST_LIBS

2005-09-22 22:38:48 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* PORTED_09:
	* configure.ac:
	* gst/auparse/gstauparse.c:
	* gst/auparse/gstauparse.h:
	  Auparse ported to 0.9. Tested with filesrc ! auparse ! osssink and alsasink
	  Original commit message from CVS:
	  Auparse ported to 0.9. Tested with filesrc ! auparse ! osssink and alsasink

2005-09-22 14:13:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Use is_filled to both check MTU and max-ptime of base class.
	  Original commit message from CVS:
	  * gst/rtp/TODO:
	  * gst/rtp/gstrtpdec.c: (gst_rtpdec_getcaps):
	  * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
	  (gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
	  (gst_rtpmp4venc_set_property):
	  * gst/rtp/gstrtpmp4venc.h:
	  * gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
	  * gst/rtp/gstrtpmpaenc.h:
	  Use is_filled to both check MTU and max-ptime of base class.

2005-09-22 11:28:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpmp4venc.c: Don't fragment packets with multiple frames.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
	  (gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
	  (gst_rtpmp4venc_set_property):
	  Don't fragment packets with multiple frames.

2005-09-22 10:39:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Remove g_print.
	  Original commit message from CVS:
	  * gst/rtp/TODO:
	  * gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps):
	  * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
	  (gst_rtpmp4venc_init), (gst_rtpmp4venc_parse_data),
	  (gst_rtpmp4venc_handle_buffer), (gst_rtpmp4venc_set_property),
	  (gst_rtpmp4venc_get_property):
	  * gst/rtp/gstrtpmp4venc.h:
	  Remove g_print.
	  Update TODO
	  Make payload encoder a bit smarter and more correct with
	  timestamps.
	  Added option in payloader to include config string in-band.

2005-09-21 19:41:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: Strip spaces for key/value pairs.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
	  (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	  (gst_rtspsrc_send):
	  Strip spaces for key/value pairs.

2005-09-21 17:53:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/gstrtspsrc.c: More SDP parsing and caps setting.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
	  (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	  (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
	  (gst_rtspsrc_change_state):
	  More SDP parsing and caps setting.
	  Do NO_PREROLL differently.
	  add pads only after negotiated.
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	  (gst_udpsrc_getcaps):
	  Implement the getcaps function.

2005-09-21 17:50:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtpamrdec.c: Handle multiple AMr packets per payload. Handle CRC and parse ILL/ILP.
	  Original commit message from CVS:
	  * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps),
	  (gst_rtpamrdec_chain):
	  Handle multiple AMr packets per payload. Handle CRC and
	  parse ILL/ILP.
	  * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps):
	  Make caps params strings for easy SDP mapping.
	  * gst/rtp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
	  Handle capsnego better.
	  * gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps):
	  * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_new_caps):
	  Generate and parse config string in the caps.

2005-09-21 12:19:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/README: Update README
	  Original commit message from CVS:
	  * gst/rtp/README:
	  Update README
	  * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps):
	  Make extra params as strings.
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
	  (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send):
	  Make state change return NO_PREROLL as this is a live
	  source.
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
	  Don't unref old caps when NULL.

2005-09-20 17:35:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtsp/: Add URI handler.
	  Original commit message from CVS:
	  * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
	  (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	  (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send),
	  (gst_rtspsrc_open), (gst_rtspsrc_uri_get_type),
	  (gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_get_uri),
	  (gst_rtspsrc_uri_set_uri), (gst_rtspsrc_uri_handler_init):
	  * gst/rtsp/sdpmessage.c: (sdp_media_get_format):
	  * gst/rtsp/sdpmessage.h:
	  Add URI handler.
	  Parse SDP and create caps.

2005-09-20 17:19:43 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	  more spec file fixoring
	  Original commit message from CVS:
	  more spec file fixoring

2005-09-20 17:04:33 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-good.spec.in:
	* gst-plugins.spec.in:
	  fix spec files
	  Original commit message from CVS:
	  fix spec files

2005-09-20 10:51:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/rtp/README:
	* gst/rtp/gstrtpamrdec.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrenc.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263pdec.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263penc.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpmp4vdec.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4venc.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadec.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpaenc.c:
	* gst/rtp/gstrtpmpapay.c:
	  don't use underscores
	  Original commit message from CVS:
	  don't use underscores

2005-09-20 07:30:31 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/alpha/gstalpha.c: fix element description
	  Original commit message from CVS:
	  * gst/alpha/gstalpha.c:
	  fix element description

2005-09-19 17:57:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/gst-plugins-good-plugins.prerequisites:
	  prereqs as well
	  Original commit message from CVS:
	  prereqs as well

2005-09-19 17:53:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/.gitignore:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.signals:
	  commit result of scanobj step
	  Original commit message from CVS:
	  commit result of scanobj step

2005-09-19 17:03:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/gstrtph263pdec.c: Don't check payload for now.
	  Original commit message from CVS:
	  * gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_chain):
	  Don't check payload for now.

2005-09-19 16:43:56 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	  add check-valgrind target
	  Original commit message from CVS:
	  add check-valgrind target

2005-09-19 16:26:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/wavparse/gstwavparse.*: Fix wavparse some more.
	  Original commit message from CVS:
	  * gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
	  (gst_wavparse_init), (gst_wavparse_parse_file_header),
	  (gst_wavparse_stream_init), (gst_wavparse_handle_seek),
	  (gst_wavparse_stream_headers), (gst_wavparse_stream_data),
	  (gst_wavparse_loop), (gst_wavparse_pad_convert),
	  (gst_wavparse_pad_query), (gst_wavparse_srcpad_event),
	  (gst_wavparse_change_state):
	  * gst/wavparse/gstwavparse.h:
	  Fix wavparse some more.

2005-09-19 11:48:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  check/elements/level.c: Fix for bus API change.
	  Original commit message from CVS:
	  * check/elements/level.c: (GST_START_TEST):
	  Fix for bus API change.

2005-09-19 11:38:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/level/level-example.c: Fix for new bus API.
	  Original commit message from CVS:
	  * gst/level/level-example.c: (main):
	  Fix for new bus API.
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
	  Set caps on pads.

2005-09-19 11:07:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/lame/gstlame.c: Set caps on outgoing buffers.
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_chain):
	  Set caps on outgoing buffers.

2005-09-19 11:06:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/debug/Makefile.am:
	  disable flags for unbuilt plugins
	  Original commit message from CVS:
	  disable flags for unbuilt plugins

2005-09-19 08:21:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* docs/plugins/scanobj-build.stamp:
	  normal builds shouldn't scan gobjects
	  Original commit message from CVS:
	  normal builds shouldn't scan gobjects

2005-09-16 16:04:28 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  clean up further so we don't try to set up five times for a simple pipeline
	  Original commit message from CVS:
	  clean up further so we don't try to set up five times for
	  a simple pipeline

2005-09-16 00:38:50 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* check/Makefile.am:
	* common:
	* tests/check/Makefile.am:
	  remove gst-register
	  Original commit message from CVS:
	  remove gst-register

2005-09-15 13:57:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ChangeLog:
	* common:
	* gst/rtp/Makefile.am:
	* gst/rtp/README:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpamrdec.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrenc.c:
	* gst/rtp/gstrtpamrenc.h:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpamrpay.h:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmdepay.h:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmenc.h:
	* gst/rtp/gstrtpgsmparse.c:
	* gst/rtp/gstrtpgsmparse.h:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgsmpay.h:
	* gst/rtp/gstrtph263pdec.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263penc.c:
	* gst/rtp/gstrtph263penc.h:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph263ppay.h:
	* gst/rtp/gstrtpmp4vdec.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4venc.c:
	* gst/rtp/gstrtpmp4venc.h:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmp4vpay.h:
	* gst/rtp/gstrtpmpadec.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpaenc.c:
	* gst/rtp/gstrtpmpaenc.h:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpapay.h:
	  Updates to payloader/depayloaders, make payloaders use the base classes.
	  Original commit message from CVS:
	  Updates to payloader/depayloaders, make payloaders use
	  the base classes.
	  Updated README with suggested RTP caps and how to convert
	  to/from SDP.
	  Added config descriptor in mp4v payloader.

2005-09-15 10:47:58 +0000  Andy Wingo <wingo@pobox.com>

	  gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_find_best): gst/autodetect/gstautovideosink.c
	  Original commit message from CVS:
	  2005-09-15  Andy Wingo  <wingo@pobox.com>
	  * gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_find_best):
	  * gst/autodetect/gstautovideosink.c
	  (gst_auto_video_sink_find_best): Update for new registry API.

2005-09-14 20:51:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  common/: a simple py script to generate valid xml from a C example probably also need to strip an MIT license when we...
	  Original commit message from CVS:
	  * common/c-to-xml.py:
	  * common/gtk-doc-plugins.mak:
	  a simple py script to generate valid xml from a C example
	  probably also need to strip an MIT license when we decide
	  * docs/plugins/Makefile.am:
	  * gst/level/Makefile.am:
	  * gst/level/gstlevel.c: (gst_level_init):
	  * gst/level/level-example.c: (message_handler), (main):
	  add an example to level that will show up in the docs
	  * gst/rtp/TODO:
	  add a note for the future

2005-09-14 11:44:11 +0000  Michael Smith <msmith@xiph.org>

	  gst/wavenc/gstwavenc.c: Actually define the debug object being used in wavenc. Fixes #316205
	  Original commit message from CVS:
	  * gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
	  Actually define the debug object being used in wavenc. Fixes #316205

2005-09-14 11:23:44 +0000  Michael Smith <msmith@xiph.org>

	* ChangeLog:
	* gst/smpte/Makefile.am:
	  Link smpte plugin against GST_BASE_LIBS, to get libgstbase; needed to build on win32 as this plugin uses collectpads ...
	  Original commit message from CVS:
	  Link smpte plugin against GST_BASE_LIBS, to get libgstbase; needed to
	  build on win32 as this plugin uses collectpads (bug 316204)

2005-09-12 16:37:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Fix up bogus ChangeLog entry
	  Original commit message from CVS:
	  Fix up bogus ChangeLog entry

2005-09-12 16:14:48 +0000  Andy Wingo <wingo@pobox.com>

	  autogen.sh (package): Now type 'make' to build gst-plugins-good.
	  Original commit message from CVS:
	  2005-09-12  Andy Wingo  <wingo@pobox.com>
	  * autogen.sh (package): Now type 'make' to build gst-plugins-good.

2005-09-11 17:52:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-fdsrc.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  add source module to docs; reinspect
	  Original commit message from CVS:
	  add source module to docs; reinspect

2005-09-09 17:56:43 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Move fdsrc back into gstreamer core elements.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/fdsrc/Makefile.am:
	  * gst/fdsrc/gstfdsrc.c:
	  * gst/fdsrc/gstfdsrc.h:
	  Move fdsrc back into gstreamer core elements.
	  * gst/level/gstlevel.c: (gst_level_class_init),
	  (gst_level_transform_ip):
	  * gst/videobox/gstvideobox.c: (gst_video_box_set_property):
	  Basetransform changes.

2005-09-09 16:11:48 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* ext/jpeg/gstsmokeenc.c:
	* ext/jpeg/smokecodec.c:
	  fix compiler warnings
	  Original commit message from CVS:
	  fix compiler warnings

2005-09-09 11:09:49 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst-plugins-good.spec.in: spec file fixes
	  Original commit message from CVS:
	  * gst-plugins-good.spec.in:
	  spec file fixes
	  * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	  (gst_multiudpsink_render), (gst_multiudpsink_add),
	  (gst_multiudpsink_clear):
	  it actually helps to actually stream if we hook up the
	  add signal to an actual implementation
	  * gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  some debugging

2005-09-08 16:58:40 +0000  Flavio Oliveira <flavio.oliveira@indt.org.br>

	* ext/jpeg/Makefile.am:
	* ext/jpeg/gstjpeg.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokeenc.c:
	  jpgenc ported to GSTreamer 0.9
	  Original commit message from CVS:
	  jpgenc ported to GSTreamer 0.9

2005-09-08 16:26:17 +0000  Flavio Oliveira <flavio.oliveira@indt.org.br>

	* ChangeLog:
	  jpegenc ported to GStreamer 0.9
	  Original commit message from CVS:
	  jpegenc ported to GStreamer 0.9

2005-09-07 13:49:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/: gsttaginterface.h -> gsttagsetter.h
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c:
	  * ext/flac/gstflacenc.c:
	  * ext/flac/gstflactag.c:
	  * ext/speex/gstspeexenc.c:
	  gsttaginterface.h -> gsttagsetter.h

2005-09-06 23:30:03 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Port to 0.9 and re-enable efence plugin.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/debug/Makefile.am:
	  * gst/debug/efence.c: (gst_efence_class_init), (gst_efence_init),
	  (gst_efence_chain), (gst_efence_buffer_alloc), (plugin_init),
	  (gst_fenced_buffer_finalize), (gst_fenced_buffer_copy),
	  (gst_fenced_buffer_alloc), (gst_fenced_buffer_class_init),
	  (gst_fenced_buffer_init), (gst_fenced_buffer_get_type):
	  Port to 0.9 and re-enable efence plugin.

2005-09-06 21:31:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/flac/gstflacdec.*: Add support for flac files with 24/32 bits per sample; and misc. minor clean-ups. Seeking is s...
	  Original commit message from CVS:
	  * ext/flac/gstflacdec.c: (flac_caps_factory), (raw_caps_factory),
	  (gst_flacdec_write), (gst_flacdec_convert_src):
	  * ext/flac/gstflacdec.h:
	  Add support for flac files with 24/32 bits per sample; and misc.
	  minor clean-ups. Seeking is still partly broken (for me at least).

2005-09-06 15:50:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/rtp/: Added mpeg4 video payload encoder/decoder.
	  Original commit message from CVS:
	  * gst/rtp/Makefile.am:
	  * gst/rtp/gstrtp.c: (plugin_init):
	  * gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_get_type),
	  (gst_rtpmp4vdec_base_init), (gst_rtpmp4vdec_class_init),
	  (gst_rtpmp4vdec_init), (gst_rtpmp4vdec_setcaps),
	  (gst_rtpmp4vdec_chain), (gst_rtpmp4vdec_set_property),
	  (gst_rtpmp4vdec_get_property), (gst_rtpmp4vdec_change_state),
	  (gst_rtpmp4vdec_plugin_init):
	  * gst/rtp/gstrtpmp4vdec.h:
	  * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_get_type),
	  (gst_rtpmp4venc_base_init), (gst_rtpmp4venc_class_init),
	  (gst_rtpmp4venc_init), (gst_rtpmp4venc_setcaps),
	  (gst_rtpmp4venc_flush), (gst_rtpmp4venc_chain),
	  (gst_rtpmp4venc_set_property), (gst_rtpmp4venc_get_property),
	  (gst_rtpmp4venc_change_state), (gst_rtpmp4venc_plugin_init):
	  * gst/rtp/gstrtpmp4venc.h:
	  * gst/rtp/gstrtpmpadec.c: (gst_rtpmpadec_chain):
	  * gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_flush):
	  Added mpeg4 video payload encoder/decoder.
	  Added some docs in mpa payloader.

2005-09-06 14:06:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.9.1 ===

2005-09-06 14:05:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* README:
	* RELEASE:
	* autogen.sh:
	* common:
	* configure.ac:
	  releasing 0.9.2
	  Original commit message from CVS:
	  releasing 0.9.2

2005-09-05 17:20:28 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* gst/videocrop/gstvideocrop.c:
	* sys/v4l2/gstv4l2element.c:
	* sys/v4l2/gstv4l2src.c:
	  Fix up all the state change functions.
	  Original commit message from CVS:
	  Fix up all the state change functions.

2005-09-05 16:28:16 +0000  Andy Wingo <wingo@pobox.com>

	  ext/dv/gstdvdemux.c (gst_dvdemux_chain): Move the pad adding here from the state change handler, so we fire signals w...
	  Original commit message from CVS:
	  2005-09-05  Andy Wingo  <wingo@pobox.com>
	  * ext/dv/gstdvdemux.c (gst_dvdemux_chain): Move the pad adding
	  here from the state change handler, so we fire signals without
	  holding the state lock.

2005-09-05 15:10:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/qtdemux/qtdemux.c:
	  cleaning up bad
	  Original commit message from CVS:
	  cleaning up bad

2005-09-05 13:18:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/.gitignore:
	* docs/plugins/.gitignore:
	  maintenance commits
	  Original commit message from CVS:
	  maintenance commits

2005-09-04 15:09:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/inspect-build.stamp:
	* docs/plugins/inspect.stamp:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-fdsrc.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	  distcheck fixes
	  Original commit message from CVS:
	  distcheck fixes

2005-09-04 11:50:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	* autogen.sh:
	* common:
	* docs/plugins/Makefile.am:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  fix distcheck
	  Original commit message from CVS:
	  fix distcheck

2005-09-02 15:56:52 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst-plugins-good.spec.in:
	  various spec fixes
	  Original commit message from CVS:
	  various spec fixes

2005-09-02 15:44:50 +0000  Andy Wingo <wingo@pobox.com>

	* check/elements/level.c:
	* examples/gstplay/player.c:
	* examples/stats/mp2ogg.c:
	* ext/aalib/gstaasink.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/esd/esdmon.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/gconf/gstgconfaudiosink.c:
	* ext/gconf/gstgconfvideosink.c:
	* ext/gdk_pixbuf/gstgdkanimation.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/ladspa/gstsignalprocessor.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libmng/gstmngdec.c:
	* ext/mikmod/gstmikmod.c:
	* ext/pango/gsttextoverlay.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/shout2/gstshout2.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* gst/alpha/gstalpha.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/debug/breakmydata.c:
	* gst/debug/gstnavigationtest.c:
	* gst/effectv/gstquark.c:
	* gst/fdsrc/gstfdsrc.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-write.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/oldcore/gstmd5sink.c:
	* gst/oldcore/gstmultifilesrc.c:
	* gst/oldcore/gstpipefilter.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16enc.c:
	* gst/rtp/gstrtpL16parse.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpamrdec.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrenc.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpdec.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmenc.c:
	* gst/rtp/gstrtpgsmparse.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263pdec.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263penc.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpmpadec.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpaenc.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/smoothwave/gstsmoothwave.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/videomixer/videomixer.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* sys/oss/gstossmixerelement.c:
	* sys/osxaudio/gstosxaudioelement.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* tests/check/elements/level.c:
	  All plugins updated for element state changes.
	  Original commit message from CVS:
	  2005-09-02  Andy Wingo  <wingo@pobox.com>
	  * All plugins updated for element state changes.

2005-09-02 15:43:54 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	  All plugins updated for element state changes.
	  Original commit message from CVS:
	  2005-09-02  Andy Wingo  <wingo@pobox.com>
	  * All plugins updated for element state changes.

2005-09-01 21:24:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/aalib/Makefile.am:
	  fix build after cleaning up my vomit
	  Original commit message from CVS:
	  fix build after cleaning up my vomit

2005-09-01 21:23:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/aalib/Makefile.am:
	  fix build after cleaning up my vomit
	  Original commit message from CVS:
	  fix build after cleaning up my vomit

2005-09-01 21:20:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/smpte/Makefile.am:
	  fix build after cleaning up my vomit
	  Original commit message from CVS:
	  fix build after cleaning up my vomit

2005-09-01 21:15:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/smpte/Makefile.am:
	  fix build after cleaning up my vomit
	  Original commit message from CVS:
	  fix build after cleaning up my vomit

2005-09-01 20:23:22 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* Makefile.am:
	* check/.gitignore:
	* check/Makefile.am:
	* check/elements/.gitignore:
	* check/elements/level.c:
	* common:
	* configure.ac:
	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	* tests/check/.gitignore:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/level.c:
	  Andrewio Patrickoforus Wingonymus - 5 additional tests for your sins
	  Original commit message from CVS:
	  Andrewio Patrickoforus Wingonymus - 5 additional tests for your sins
	  Add a regression test for level and fix a casting bug that made the additional
	  channels turn out wrong

2005-09-01 17:55:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  add docs to build
	  Original commit message from CVS:
	  * Makefile.am:
	  * configure.ac:
	  add docs to build
	  * common/plugins.xsl:
	  wrap Description into a refsect2
	  * docs/Makefile.am:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-good-plugins-sections.txt:
	  * gst/goom/Makefile.am:
	  * gst/goom/gstgoom.c: (gst_goom_get_type), (gst_goom_base_init),
	  (gst_goom_class_init), (gst_goom_init), (gst_goom_dispose),
	  (gst_goom_sink_setcaps), (gst_goom_src_setcaps),
	  (gst_goom_src_negotiate), (gst_goom_event), (gst_goom_chain),
	  (gst_goom_change_state):
	  * gst/goom/gstgoom.h:
	  GstGOOM -> GstGoom
	  add an example launch line
	  * gst/level/gstlevel.h:
	  * gst/monoscope/gstmonoscope.c:
	  cleanups

2005-08-31 16:28:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/dvdlpcmdec/.gitignore:
	* gst/dvdlpcmdec/Makefile.am:
	* gst/dvdlpcmdec/gstdvdlpcmdec.c:
	* gst/dvdlpcmdec/gstdvdlpcmdec.h:
	  remove dvdlpcmdec, it's dvd stuff
	  Original commit message from CVS:
	  remove dvdlpcmdec, it's dvd stuff

2005-08-30 19:41:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	* gst-libs/gst/gettext.h:
	* gst-libs/gst/gst-i18n-plugin.h:
	  add some i18n headers
	  Original commit message from CVS:
	  add some i18n headers

2005-08-30 19:24:37 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/.gitignore:
	  ignore more
	  Original commit message from CVS:
	  ignore more

2005-08-30 19:24:03 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/Makefile.am:
	  Makefile.am
	  Original commit message from CVS:
	  Makefile.am

2005-08-30 19:20:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/upload.mak:
	* docs/version.entities.in:
	  commit new stuff
	  Original commit message from CVS:
	  commit new stuff

2005-08-30 19:01:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* common:
	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.types:
	  document elements and plugins.  Shazam !
	  Original commit message from CVS:
	  document elements and plugins.  Shazam !

2005-08-30 17:37:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* .gitignore:
	* COPYING:
	* RELEASE:
	* gst-plugins-good.spec.in:
	  add some files
	  Original commit message from CVS:
	  add some files

2005-08-17 19:05:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  configure.ac: Added mpegaudioparse
	  Original commit message from CVS:
	  * configure.ac:
	  Added mpegaudioparse
	  * ext/lame/gstlame.c: (gst_lame_src_getcaps),
	  (gst_lame_src_setcaps), (gst_lame_sink_setcaps),
	  (gst_lame_sink_event), (gst_lame_chain):
	  Some cleanups.
	  Fix memleak.
	  * gst/mpegaudioparse/gstmpegaudioparse.c:
	  (gst_mp3parse_class_init), (gst_mp3parse_init),
	  (gst_mp3parse_chain), (gst_mp3parse_change_state):
	  * gst/mpegaudioparse/gstmpegaudioparse.h:
	  Ported mpegaudioparse

2005-08-16 16:12:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Fix compile warning.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/amrnb/amrnbparse.c: (gst_amrnbparse_read_header):
	  Fix compile warning.
	  * ext/lame/gstlame.c: (gst_lame_class_init),
	  (gst_lame_src_getcaps), (gst_lame_src_setcaps),
	  (gst_lame_sink_setcaps), (gst_lame_init), (gst_lame_sink_event),
	  (gst_lame_chain), (gst_lame_change_state):
	  * ext/lame/gstlame.h:
	  Port lame plugin

2005-07-05 10:51:49 +0000  Andy Wingo <wingo@pobox.com>

	  Way, way, way too many files: Remove crack comment from the 2000 era.
	  Original commit message from CVS:
	  2005-07-05  Andy Wingo  <wingo@pobox.com>
	  * Way, way, way too many files:
	  Remove crack comment from the 2000 era.

2005-07-05 10:51:41 +0000  Andy Wingo <wingo@pobox.com>

	  Way, way, way too many files: Remove crack comment from the 2000 era.
	  Original commit message from CVS:
	  2005-07-05  Andy Wingo  <wingo@pobox.com>
	  * Way, way, way too many files:
	  Remove crack comment from the 2000 era.

2004-10-26 11:36:52 +0000  Iain Holmes <iain@prettypeople.org>

	* ext/lame/gstlame.c:
	  Memory leak fixes
	  Original commit message from CVS:
	  Memory leak fixes
	  Allow level to take mono or stereo audio

2004-08-26 00:32:00 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.*: Added new media support to lame
	  Original commit message from CVS:
	  2004-08-26  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain):
	  * ext/lame/gstlame.h:
	  Added new media support to lame

2004-08-19 22:44:50 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  Only enable lame presets if version of lame has presets in API
	  Original commit message from CVS:
	  2004-08-19  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * configure.ac:
	  * ext/lame/Makefile.am:
	  * ext/lame/gstlame.c: (gst_lame_class_init),
	  (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup):
	  Only enable lame presets if version of lame has presets in API

2004-08-15 13:47:00 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.c: describe the enum values for vbr mode and presets more verbosely
	  Original commit message from CVS:
	  2004-08-15  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_vbrmode_get_type),
	  (gst_lame_preset_get_type), (gst_lame_class_init):
	  describe the enum values for vbr mode and presets more verbosely

2004-08-13 15:22:49 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.*: add preset property to lame so it can use lame presets
	  Original commit message from CVS:
	  2004-08-13  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_mode_get_type),
	  (gst_lame_quality_get_type), (gst_lame_padding_get_type),
	  (gst_lame_preset_get_type), (gst_lame_class_init), (gst_lame_init),
	  (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup):
	  * ext/lame/gstlame.h:
	  add preset property to lame so it can use lame presets

2004-08-13 14:55:27 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.c: whoops forgot break, thanks teuf
	  Original commit message from CVS:
	  2004-08-13  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_get_property):
	  whoops forgot break, thanks teuf

2004-08-13 14:41:02 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.*: fix lame's broken vbr stuff, allow it to resample if need be, and also make xing header optional
	  Original commit message from CVS:
	  2004-08-13  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_vbrmode_get_type),
	  (gst_lame_class_init), (gst_lame_src_getcaps),
	  (gst_lame_sink_link), (gst_lame_init), (gst_lame_set_property),
	  (gst_lame_get_property), (gst_lame_setup):
	  * ext/lame/gstlame.h:
	  fix lame's broken vbr stuff, allow it to resample if need be, and also
	  make xing header optional

2004-08-12 17:22:30 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.c: added getcaps function so samplerate doesntget fixated to silly values
	  Original commit message from CVS:
	  2004-08-12  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_src_getcaps), (gst_lame_init):
	  added getcaps function so samplerate doesntget fixated to silly values

2004-08-12 16:44:14 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.c: revert previous fix
	  Original commit message from CVS:
	  2004-08-12  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_src_link):
	  revert previous fix

2004-08-12 16:12:00 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.c: made source pad link function check if sinkpad is ok..fixes the problem where core fixates the ou...
	  Original commit message from CVS:
	  2004-08-12  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_src_link):
	  made source pad link function check if sinkpad is ok..fixes the problem
	  where core fixates the output rate of lame stupidly

2004-08-12 15:48:50 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.c: set default compression ratio paramter to 0.0 so bitrate parameter works :)
	  Original commit message from CVS:
	  2004-08-12  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init):
	  set default compression ratio paramter to 0.0 so bitrate parameter
	  works :)

2004-08-09 09:22:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  fix add debugging
	  Original commit message from CVS:
	  fix add debugging

2004-08-02 11:39:17 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  gearing up for release
	  Original commit message from CVS:
	  gearing up for release

2004-08-02 09:16:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  add link function. fixes @148986
	  Original commit message from CVS:
	  add link function. fixes @148986

2004-07-28 20:26:31 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  ext/lame/gstlame.c: send tag events downstream
	  Original commit message from CVS:
	  2004-07-28  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * ext/lame/gstlame.c: (gst_lame_chain): send tag events downstream
	  * ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type),
	  (gst_shout2send_get_type), (gst_shout2send_set_clock),
	  (gst_shout2send_class_init), (gst_shout2send_init),
	  (set_shout_metadata), (gst_shout2send_set_metadata),
	  (gst_shout2send_chain), (gst_shout2send_set_property),
	  (gst_shout2send_get_property), (gst_shout2send_connect),
	  (gst_shout2send_change_state):
	  * ext/shout2/gstshout2.h:
	  - fix for sending mp3 audio to icecast2 server, if pad link function not
	  called before PAUSED state
	  - added option to use GStreamer clock sync (as opposed to libshout's own sync)
	  - added tagging support for mp3 audio broadcasted
	  * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
	  debug info

2004-07-27 21:51:30 +0000  Steve Lhomme <steve.lhomme@free.fr>

	* gst/audiofx/gststereo.c:
	  fix local includes and 64 bits constants
	  Original commit message from CVS:
	  fix local includes and 64 bits constants

2004-07-26 15:42:18 +0000  Benjamin Otte <otte@gnome.org>

	  ext/lame/gstlame.c: add debugging category, add error checks like checking return values of setup calls, make sure it...
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_sink_link), (gst_lame_init),
	  (gst_lame_chain), (gst_lame_setup), (gst_lame_change_state),
	  (plugin_init):
	  add debugging category, add error checks like checking return values
	  of setup calls, make sure it still works after
	  PLAYING=>NULL=>PLAYING, fix encoding of mono streams

2004-06-14 10:58:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  sync mp3 caps
	  Original commit message from CVS:
	  sync mp3 caps

2004-06-14 10:52:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  add comment
	  Original commit message from CVS:
	  add comment

2004-05-21 23:28:57 +0000  Stéphane Loeuillet <gstreamer@leroutier.net>

	* ext/lame/gstlame.c:
	  second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in ...
	  Original commit message from CVS:
	  second batch :
	  remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
	  (in gst-plugins/ext/ this time)

2004-05-09 14:37:15 +0000  Benjamin Otte <otte@gnome.org>

	  ext/: \1/Codec, (fixes #142193)
	  Original commit message from CVS:
	  reviewed by Benjamin Otte  <otte@gnome.org>
	  * ext/a52dec/gsta52dec.c:
	  * ext/divx/gstdivxdec.c:
	  * ext/divx/gstdivxenc.c:
	  * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
	  * ext/faac/gstfaac.c: (gst_faac_base_init):
	  * ext/faad/gstfaad.c: (gst_faad_base_init):
	  * ext/ivorbis/vorbisfile.c:
	  * ext/lame/gstlame.c:
	  * ext/libfame/gstlibfame.c:
	  * ext/mpeg2enc/gstmpeg2enc.cc:
	  * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
	  * ext/sidplay/gstsiddec.cc:
	  * ext/speex/gstspeexdec.c:
	  * ext/speex/gstspeexenc.c:
	  * ext/xvid/gstxviddec.c:
	  * ext/xvid/gstxvidenc.c:
	  correct klasses. Mostly s,Codec/(Audio|Video),\1/Codec,
	  (fixes #142193)

2004-05-07 00:43:50 +0000  Benjamin Otte <otte@gnome.org>

	  ext/lame/gstlame.c: simplify
	  Original commit message from CVS:
	  * ext/lame/gstlame.c: (gst_lame_chain):
	  simplify
	  * ext/mad/gstmad.c: (gst_mad_handle_event):
	  fix event leak
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
	  be able to detect mp3 files < 4096 bytes

2004-05-03 16:46:10 +0000  Stéphane Loeuillet <gstreamer@leroutier.net>

	* ext/lame/gstlame.c:
	  don't trust lame_init to set good values as defaults
	  Original commit message from CVS:
	  don't trust lame_init to set good values as defaults

2004-03-15 19:32:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/audiofx/gststereo.c:
	  don't mix tabs and spaces
	  Original commit message from CVS:
	  don't mix tabs and spaces

2004-03-15 19:32:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  don't mix tabs and spaces
	  Original commit message from CVS:
	  don't mix tabs and spaces

2004-03-15 16:32:54 +0000  Johan Dahlin <johan@gnome.org>

	  *.h: Revert indenting
	  Original commit message from CVS:
	  * *.h: Revert indenting

2004-03-15 16:32:53 +0000  Johan Dahlin <johan@gnome.org>

	  *.h: Revert indenting
	  Original commit message from CVS:
	  * *.h: Revert indenting

2004-03-14 22:34:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/audiofx/gststereo.c:
	* gst/audiofx/gststereo.h:
	  gst-indent
	  Original commit message from CVS:
	  gst-indent

2004-03-14 22:34:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	* ext/lame/test-lame.c:
	  gst-indent
	  Original commit message from CVS:
	  gst-indent

2004-02-22 15:14:24 +0000  Benjamin Otte <otte@gnome.org>

	  configure.ac: export [_]*{gst,Gst,GST}.* symbols from plugins
	  Original commit message from CVS:
	  2004-02-22  Benjamin Otte  <otte@gnome.org>
	  * configure.ac:
	  export [_]*{gst,Gst,GST}.* symbols from plugins
	  2004-02-22  Christophe Fergeau <teuf@gnome.org>
	  reviewed by: Benjamin Otte  <otte@gnome.org>
	  * ext/lame/gstlame.c: (add_one_tag):
	  * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value),
	  (gst_vorbisenc_metadata_set1):
	  * gst/tags/gstid3tag.c:
	  * gst/tags/gstvorbistag.c: (gst_vorbis_tag_add):
	  apply fixes from bugs #135042 (lame can't write tags) and #133817
	  (add GST_ALBUM_VOLUME_{COUNT,NUMBER} tags)

2004-02-19 22:19:55 +0000  Benjamin Otte <otte@gnome.org>

	  ext/: use gst_tag_list_insert when you want to insert tags
	  Original commit message from CVS:
	  2004-02-19  Benjamin Otte  <otte@gnome.org>
	  * ext/lame/gstlame.c: (gst_lame_chain):
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
	  use gst_tag_list_insert when you want to insert tags

2004-02-02 17:23:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  change NULL to (NULL) for GST_ELEMENT_ERROR
	  Original commit message from CVS:
	  change NULL to (NULL) for GST_ELEMENT_ERROR
	  Make sure errors end with "."

2004-01-29 23:20:44 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  GST_ELEMENT_ERROR
	  Original commit message from CVS:
	  GST_ELEMENT_ERROR

2004-01-18 21:46:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  use new error signal and classification
	  Original commit message from CVS:
	  use new error signal and classification

2003-12-22 01:47:08 +0000  David Schleef <ds@schleef.org>

	* ext/lame/gstlame.c:
	  Merge CAPS branch
	  Original commit message from CVS:
	  Merge CAPS branch

2003-12-07 14:47:09 +0000  Christophe Fergeau <teuf@gnome.org>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  Uses new tagging framework
	  Original commit message from CVS:
	  Uses new tagging framework

2003-12-04 10:37:38 +0000  Andy Wingo <wingo@pobox.com>

	* gst/audiofx/gststereo.c:
	  remove copyright field from plugins
	  Original commit message from CVS:
	  remove copyright field from plugins

2003-12-04 10:37:35 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	  remove copyright field from plugins
	  Original commit message from CVS:
	  remove copyright field from plugins

2003-12-02 02:28:12 +0000  David Schleef <ds@schleef.org>

	* ext/lame/test-lame.c:
	  change _connect to _link
	  Original commit message from CVS:
	  change _connect to _link

2003-11-16 22:02:23 +0000  Leif Johnson <leif@ambient.2y.net>

	* gst/audiofx/gststereo.c:
	  + checking in plugin category changes
	  Original commit message from CVS:
	  + checking in plugin category changes

2003-11-07 12:47:02 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* gst/audiofx/gststereo.h:
	  Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro...
	  Original commit message from CVS:
	  Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files

2003-11-07 12:46:51 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/gstlame.h:
	  Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro...
	  Original commit message from CVS:
	  Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files

2003-11-02 22:34:11 +0000  Benjamin Otte <otte@gnome.org>

	* gst/audiofx/gststereo.c:
	  fix for new plugin system
	  Original commit message from CVS:
	  fix for new plugin system

2003-11-02 00:13:26 +0000  Iain Holmes <iain@prettypeople.org>

	* ext/lame/gstlame.c:
	  Fixed lame too
	  Original commit message from CVS:
	  Fixed lame too

2003-10-09 09:04:23 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/gstlame.c:
	  Fix typo in Andy's commit
	  Original commit message from CVS:
	  Fix typo in Andy's commit

2003-10-08 16:08:19 +0000  Andy Wingo <wingo@pobox.com>

	* gst/audiofx/gststereo.c:
	  /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
	  Original commit message from CVS:
	  /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.

2003-10-08 16:08:10 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	  /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
	  Original commit message from CVS:
	  /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.

2003-09-30 19:48:39 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/gstlame.c:
	  Input and output samplerate are *not* necessarily the same in lame. This fixes the output caps
	  Original commit message from CVS:
	  Input and output samplerate are *not* necessarily the same in lame. This fixes the output caps

2003-09-16 10:00:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  reverting error patch before making a branch.
	  Original commit message from CVS:
	  reverting error patch before making a branch.

2003-09-15 01:08:38 +0000  Benjamin Otte <otte@gnome.org>

	* ext/lame/gstlame.c:
	  converted gst_element_error to new format in ext/ - gettext pending
	  Original commit message from CVS:
	  converted gst_element_error to new format in ext/ - gettext pending

2003-09-12 11:35:23 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/gstlame.c:
	  Fix tiny caps error in lame caps - mpegversion(1) was missing
	  Original commit message from CVS:
	  Fix tiny caps error in lame caps - mpegversion(1) was missing

2003-08-10 00:01:58 +0000  David Schleef <ds@schleef.org>

	* ext/lame/Makefile.am:
	  Remove redundant plugindir definition
	  Original commit message from CVS:
	  Remove redundant plugindir definition

2003-07-10 15:39:11 +0000  Christian Schaller <uraeus@gnome.org>

	* ext/lame/README:
	* ext/lame/gstlame.c:
	  fix license field of lame plugin to say LGPL, lame is LGPL. Add Readme with info
	  Original commit message from CVS:
	  fix license field of lame plugin to say LGPL, lame is LGPL. Add Readme with info

2003-07-06 20:49:50 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/gstlame.c:
	  New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri...
	  Original commit message from CVS:
	  New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs

2003-07-05 22:48:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  patch from hadess, modified
	  Original commit message from CVS:
	  patch from hadess, modified

2003-06-29 19:46:13 +0000  Benjamin Otte <otte@gnome.org>

	* gst/audiofx/gststereo.c:
	  compatibility fix for new GST_DEBUG stuff.
	  Original commit message from CVS:
	  compatibility fix for new GST_DEBUG stuff.
	  Includes fixes for missing includes for config.h and unistd.h
	  I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.

2003-06-29 19:46:09 +0000  Benjamin Otte <otte@gnome.org>

	* ext/lame/gstlame.c:
	  compatibility fix for new GST_DEBUG stuff.
	  Original commit message from CVS:
	  compatibility fix for new GST_DEBUG stuff.
	  Includes fixes for missing includes for config.h and unistd.h
	  I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.

2003-06-07 00:34:51 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  Another duration patch from Joshua (slightly modified by me)
	  Original commit message from CVS:
	  Another duration patch from Joshua (slightly modified by me)

2003-05-29 19:32:39 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/gstlame.h:
	  Fix build prob
	  Original commit message from CVS:
	  Fix build prob

2003-05-29 12:41:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	  - copy offset from input buffer
	  Original commit message from CVS:
	  - copy offset from input buffer

2003-05-13 12:28:16 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  Get timestamping somewhat better
	  Original commit message from CVS:
	  Get timestamping somewhat better

2003-05-12 20:08:17 +0000  Zeeshan Ali <zeenix@gmail.com>

	* ext/lame/gstlame.c:
	  Hacked lame to make it copy the timestamp on the source buffer to the sink buffer
	  Original commit message from CVS:
	  Hacked lame to make it copy the timestamp on the source buffer to the sink buffer

2003-01-10 13:38:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  PadConnect -> PadLink
	  Original commit message from CVS:
	  PadConnect -> PadLink

2003-01-10 10:22:24 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t...
	  Original commit message from CVS:
	  another batch of connect->link fixes
	  please let me know about issues
	  and please refrain of making them yourself, so that I don't spend double
	  the time resolving conflicts

2002-12-08 17:20:44 +0000  Iain Holmes <iain@prettypeople.org>

	* ext/lame/gstlame.c:
	  Replace audio/mp3 with audio/x-mp3 and audio/x-flac with application/x-flac
	  Original commit message from CVS:
	  Replace audio/mp3 with audio/x-mp3 and audio/x-flac with application/x-flac

2002-12-08 14:50:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/Makefile.am:
	  parallel install fixes
	  Original commit message from CVS:
	  parallel install fixes

2002-12-08 02:44:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	  cleanups
	  Original commit message from CVS:
	  cleanups

2002-11-20 21:02:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	  Remove redundant properties.
	  Original commit message from CVS:
	  Remove redundant properties.

2002-11-02 05:39:21 +0000  David I. Lehn <dlehn@users.sourceforge.net>

	* ext/lame/Makefile.am:
	  use AM_CFLAGS instead of CFLAGS
	  Original commit message from CVS:
	  use AM_CFLAGS instead of CFLAGS

2002-10-02 08:04:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  api change
	  Original commit message from CVS:
	  api change

2002-09-18 19:02:52 +0000  Christian Schaller <uraeus@gnome.org>

	* gst/audiofx/gststereo.c:
	  plugins part of license field patch
	  Original commit message from CVS:
	  plugins part of license field patch

2002-09-18 19:02:46 +0000  Christian Schaller <uraeus@gnome.org>

	* ext/lame/gstlame.c:
	  plugins part of license field patch
	  Original commit message from CVS:
	  plugins part of license field patch

2002-09-10 09:31:38 +0000  Ronald S. Bultje <rbultje@ronald.bitfreak.net>

	* ext/lame/test-lame.c:
	  This updates all plugins to the new API for gst_pad_try_set_caps
	  Original commit message from CVS:
	  This updates all plugins to the new API for gst_pad_try_set_caps

2002-09-01 15:40:39 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  small updates
	  Original commit message from CVS:
	  small updates

2002-07-08 19:32:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	  unref event
	  Original commit message from CVS:
	  unref event

2002-07-07 14:17:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	  Don't free uninitialized pointers
	  Original commit message from CVS:
	  Don't free uninitialized pointers

2002-07-07 14:06:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	  Lame should accept events even when not negotiated yet.
	  Original commit message from CVS:
	  Lame should accept events even when not negotiated yet.

2002-06-08 09:26:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  reorder
	  Original commit message from CVS:
	  reorder

2002-04-20 21:42:51 +0000  Andy Wingo <wingo@pobox.com>

	* gst/audiofx/gststereo.c:
	  a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br...
	  Original commit message from CVS:
	  * a hack to work around intltool's brokenness
	  * a current check for mpeg2dec
	  * details->klass reorganizations
	  * an element browser that uses details->klass
	  * separated cdxa parse out from the avi directory

2002-04-11 20:42:26 +0000  Andy Wingo <wingo@pobox.com>

	* gst/audiofx/gststereo.c:
	  GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind.
	  Original commit message from CVS:
	  GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE
	  same with *factory and typefind.
	  also, some -Werror fixes.

2002-04-11 20:42:25 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	* ext/lame/test-lame.c:
	  GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind.
	  Original commit message from CVS:
	  GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE
	  same with *factory and typefind.
	  also, some -Werror fixes.

2002-03-30 17:06:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	* ext/lame/test-lame.c:
	  Changed to the new props API
	  Original commit message from CVS:
	  Changed to the new props API
	  Other small tuff.

2002-03-27 04:02:38 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	  update g_value stuff to match property types
	  Original commit message from CVS:
	  update g_value stuff to match property types

2002-03-24 22:07:03 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	  filter newlines out of GST_DEBUG statements to reflect new core behavior fixes to adder's caps, again
	  Original commit message from CVS:
	  * filter newlines out of GST_DEBUG statements to reflect new core behavior
	  * fixes to adder's caps, again

2002-03-20 21:45:04 +0000  Andy Wingo <wingo@pobox.com>

	* gst/audiofx/gststereo.c:
	* gst/audiofx/gststereo.h:
	  s/Gnome-Streamer/GStreamer/
	  Original commit message from CVS:
	  s/Gnome-Streamer/GStreamer/

2002-03-20 21:45:03 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  s/Gnome-Streamer/GStreamer/
	  Original commit message from CVS:
	  s/Gnome-Streamer/GStreamer/

2002-03-19 17:14:57 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	  fix compile error (untested)
	  Original commit message from CVS:
	  fix compile error (untested)

2002-03-19 04:10:06 +0000  Andy Wingo <wingo@pobox.com>

	* gst/audiofx/gststereo.c:
	  removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with...
	  Original commit message from CVS:
	  * removal of //-style comments
	  * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
	  and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.

2002-03-19 04:10:05 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/Makefile.am:
	* ext/lame/gstlame.c:
	  removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with...
	  Original commit message from CVS:
	  * removal of //-style comments
	  * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
	  and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.

2002-03-19 01:39:42 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/Makefile.am:
	  s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi...
	  Original commit message from CVS:
	  s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/
	  @-substitued variables variables are defined as make variables automagically,
	  and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag

2002-03-03 00:53:24 +0000  Andy Wingo <wingo@pobox.com>

	* ext/lame/gstlame.c:
	  get up-to-date with the gst_caps_debug api improved capsnego in mad improved capsnego in adder improved capsnego in i...
	  Original commit message from CVS:
	  * get up-to-date with the gst_caps_debug api
	  * improved capsnego in mad
	  * improved capsnego in adder
	  * improved capsnego in intfloat plugins
	  * unbroke capsnego in stereomono plugins
	  * fix cothread stack allocation within the main thread in new cothreads

2002-02-21 17:33:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/Makefile.am:
	  uncomment lame test until we can get the register to work
	  Original commit message from CVS:
	  uncomment lame test until we can get the register to work

2002-02-21 17:20:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  use gst-debuginfo.m4 macro so plugins are actually compiled with debug info some more debug output for lame
	  Original commit message from CVS:
	  * use gst-debuginfo.m4 macro so plugins are actually compiled with
	  debug info
	  * some more debug output for lame

2002-02-21 14:04:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  on sink connect, check if the current pad is compatible with the given caps cleaned up debug output change pad templa...
	  Original commit message from CVS:
	  * on sink connect, check if the current pad is compatible with the given
	  caps
	  * cleaned up debug output
	  * change pad template to only accept allowed sample rates
	  if these changes are considered ok by others then the same should be
	  applied to other encoding plugins (notably the compatibility check)

2002-02-19 20:49:52 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/test-lame.c:
	  ok, this works
	  Original commit message from CVS:
	  ok, this works

2002-02-19 20:35:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/test-lame.c:
	  Always bring the elements to READY before trying to do capsnego. fix the caps as lame doesn't accept law==1
	  Original commit message from CVS:
	  Always bring the elements to READY before trying to do capsnego.
	  fix the caps as lame doesn't accept law==1

2002-02-19 20:19:36 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/test-lame.c:
	  still does not work ;(
	  Original commit message from CVS:
	  still does not work ;(

2002-02-19 18:28:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/Makefile.am:
	* ext/lame/test-lame.c:
	  adding a test for lame stuff
	  Original commit message from CVS:
	  adding a test for lame stuff

2002-02-19 17:29:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	  Added event handling.
	  Original commit message from CVS:
	  Added event handling.
	  Fix flush
	  Fix state change.
	  Convert to gobject deep_notify

2002-02-19 12:55:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.c:
	  somebody help me fix lame ;)
	  Original commit message from CVS:
	  somebody help me fix lame ;)
	  I commented out the state change function because it is called before lame has the right caps.
	  Is the state change function still necessary ?
	  in any case, at least now lame actually listens to osssrc re: rate and channels

2002-01-31 17:08:46 +0000  David I. Lehn <dlehn@users.sourceforge.net>

	* ext/lame/gstlame.h:
	  Revert lame include dir change.  Upstream uses $prefix/include/lame/lame.h.
	  Original commit message from CVS:
	  Revert lame include dir change.  Upstream uses $prefix/include/lame/lame.h.

2002-01-30 11:25:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/gstlame.h:
	  I checked lame packages and source code and they seem to want lame.h in prefix/include/lame.h so I fixed stuff accord...
	  Original commit message from CVS:
	  I checked lame packages and source code and they seem to want lame.h in
	  prefix/include/lame.h
	  so I fixed stuff accordingly.
	  Do any systems have lame in include/lame/lame.h ?
	  If so, mail me and we'll work it out.

2002-01-18 02:05:25 +0000  Wrobell <wrobell@ite.pl>

	* ext/lame/Makefile.am:
	  - plugins are built without versioning info
	  Original commit message from CVS:
	  - plugins are built without versioning info

2002-01-13 22:27:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	  Bring the plugins in sync with the new core capsnego system.
	  Original commit message from CVS:
	  Bring the plugins in sync with the new core capsnego system.
	  Added some features, enhancements...

2002-01-12 03:34:26 +0000  David I. Lehn <dlehn@users.sourceforge.net>

	* ext/lame/Makefile.am:
	  s/filter/plugin/ link plugins to GST_LIBS rearrange rules to a common format
	  Original commit message from CVS:
	  * s/filter/plugin/
	  * link plugins to GST_LIBS
	  * rearrange rules to a common format

2001-12-22 23:26:33 +0000  Andy Wingo <wingo@pobox.com>

	* gst/audiofx/gststereo.c:
	* gst/audiofx/gststereo.h:
	  Initial revision
	  Original commit message from CVS:
	  Initial revision

2001-12-21 12:47:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  Lame cleanup
	  Original commit message from CVS:
	  Lame cleanup
	  Added EOS, flush, error reporting etc.

2001-12-20 23:48:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/lame/Makefile.am:
	* ext/lame/gstlame.c:
	* ext/lame/gstlame.h:
	  adding lame
	  Original commit message from CVS:
	  adding lame

2001-12-17 18:37:01 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  building up speed
	  Original commit message from CVS:
	  building up speed

