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=== release 0.11.1 ===

2011-09-29  Wim Taymans <wim.taymans@collabora.co.uk>

	* configure.ac:
	  releasing 0.11.1, "A handful sometimes, A heartful always"

2011-09-29 13:46:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/adder.c:
	* tests/check/elements/audioconvert.c:
	* tests/check/elements/audiorate.c:
	* tests/check/elements/audioresample.c:
	* tests/check/elements/audiotestsrc.c:
	* tests/check/elements/decodebin2.c:
	* tests/check/elements/encodebin.c:
	* tests/check/elements/gdpdepay.c:
	* tests/check/elements/gdppay.c:
	* tests/check/elements/playbin-compressed.c:
	* tests/check/elements/videorate.c:
	* tests/check/elements/videotestsrc.c:
	* tests/check/elements/volume.c:
	* tests/check/libs/audio.c:
	* tests/check/libs/pbutils.c:
	* tests/check/libs/profile.c:
	* tests/check/pipelines/simple-launch-lines.c:
	* tests/check/pipelines/vorbisdec.c:
	* tests/check/pipelines/vorbisenc.c:
	  tests: update for new audio caps

2011-09-29 13:45:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* win32/common/libgstaudio.def:
	  defs: add new symbols

2011-09-28 16:08:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: fix refcounting error

2011-09-28 16:07:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstringbuffer.c:
	  ringbuffer: store info so we can debug it

2011-09-28 15:46:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-09-28 15:41:16 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: really push pending events

2011-09-28 15:35:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	  audiodecoder: add method to set output caps
	  Add a method to configure the output caps. Subclasses can't use
	  gst_pad_set_caps() anymore because then we won't see the caps.
	  Unbreak the padtemplate registration, the GTypeClass that is configured in the
	  object during _init is not the right one, we need to use the klass passed as the
	  argument to the init function..

2011-09-28 14:32:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: remove more tags from upstream tag events such as bitrate tags
	  We want to remove all codec specific tags.

2011-09-28 11:35:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst-libs/gst/audio/gstaudioencoder.c
	  gst/playback/gstplaybin2.c
	  gst/videotestsrc/videotestsrc.c

2011-09-28 01:56:42 +0300  Raimo Järvi <raimo.jarvi@gmail.com>

	* gst/videotestsrc/videotestsrc.c:
	  videotestsrc: Fix compiler warning on 64 bit mingw-w64
	  Fixes bug #660304.

2011-09-28 01:11:30 +0300  Raimo Järvi <raimo.jarvi@gmail.com>

	* gst/playback/gstplaybin2.c:
	  playbin2: Fix compiler warnings on 64 bit mingw-w64
	  Fixes bug #660301.

2011-09-27 16:18:05 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: only got_data if we really got some
	  ... which avoids going loopy with casual subclass.

2011-09-27 16:57:45 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: really push pending events

2011-09-27 16:16:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: send tag event after pending events
	  ... which probably includes a pending newsegment event.

2011-09-27 16:16:29 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: protect pending_events with proper lock

2011-09-27 15:31:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: clean up some documentation

2011-09-27 11:19:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstringbuffer.h:
	* gst-libs/gst/audio/multichannel.h:
	* gst-libs/gst/video/convertframe.c:
	* gst-libs/gst/video/video.h:
	  docs: improve docs

2011-09-27 00:32:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: minor docs fix

2011-09-26 21:11:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioenc: fix compilation

2011-09-26 19:22:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst-libs/gst/audio/gstaudiodecoder.c
	  gst-libs/gst/audio/gstaudioencoder.c
	  gst/encoding/gstencodebin.c

2011-09-26 16:36:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: Adjust for GstAudioEncoder API changes

2011-09-26 16:36:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* win32/common/libgstaudio.def:
	  win32: Adjust for GstAudioEncoder API changes

2011-09-26 16:35:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: Improve set_frame_sample_{min,max} documentation

2011-09-26 16:22:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	  audiodecoder: Fix thread safety issues if both pads have different streaming threads

2011-09-26 16:19:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: Delay sending of serialized events to finish_frame()

2011-09-26 16:02:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  Revert "audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code"
	  This reverts commit 11e375486e07cfa0686a97b5cf6110909b3a828c.
	  GST_BOILERPLATE() can't define an abstract type and
	  G_DEFINE_ABSTRACT_TYPE() does not pass the class struct to
	  the instance_init function and there's no way to get the
	  class struct of the current type in instance_init().

2011-09-26 15:59:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstaudioencoder.h:
	  audioencoder: Add support for requesting a minimum and maximum number of samples per frame
	  This extends the special case of a fixed number of samples per frame
	  that was supported before already.

2011-09-26 15:45:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstaudioencoder.h:
	  audioencoder: Fix thread safety issues if both pads have different streaming threads

2011-09-26 15:42:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: Delay sending of serialized events to finish_frame()
	  This makes sure that the caps are already set before any serialized
	  events are sent downstream.

2011-09-26 15:34:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code

2011-09-26 15:14:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstaudioencoder.h:
	  audioencoder: add some tag handling convenience help

2011-09-26 14:48:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: provide CODEC/AUDIO_CODEC handling

2011-09-26 13:42:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: filter AUDIO_CODEC/CODEC tags from passing tag events

2011-09-25 15:31:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c:
	  typefindfunctions: backport some const-ifications from 0.11 branch
	  To keep code identical as much as possible between the two branches,
	  for easier merging.

2011-09-25 15:24:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c:
	  typefindfunctions: fix indentation

2011-09-23 21:18:47 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Android.mk:
	* configure.ac:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/floatcast/Makefile.am:
	* gst-libs/gst/floatcast/floatcast.h:
	* gst-plugins-base.spec.in:
	* gst/audioconvert/audioconvert.c:
	* pkgconfig/Makefile.am:
	* pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
	* pkgconfig/gstreamer-floatcast.pc.in:
	* tests/check/elements/audioconvert.c:
	* tests/check/libs/gstlibscpp.cc:
	  libs: remove unused floatcast header-only library
	  There's no code whatsoever that uses these macros. If anyone
	  ever feels the need to resurrect them, we should add them to
	  gstutils.h in core or libgstaudio or so.

2011-09-23 18:27:11 +0200  Edward Hervey <bilboed@bilboed.com>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/ogg/gstoggdemux.c
	  ext/pango/gsttextoverlay.c
	  gst-libs/gst/audio/gstaudioencoder.c
	  gst-libs/gst/audio/gstbaseaudiosrc.c
	  gst/playback/gstsubtitleoverlay.c
	  gst/videorate/gstvideorate.c

2011-09-23 17:50:31 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/encoding/gstencodebin.c:
	  encodebin: Avoid unnecessary read only caps copy

2011-09-21 13:30:43 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	  gst-libs: Temporarily remove dependency of gstaudio on gstpbutils
	  Also re-order the SUBDIRS in the higher-level Makefile so it cleanly
	  installs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657675

2011-09-22 15:38:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: proxy some more optional downstream caps fields to upstream

2011-09-22 15:38:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: changed is verily the opposite of equal

2011-09-22 15:37:07 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: prevent crashing when comparing to a freshly inited GstAudioInfo

2011-09-22 15:36:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/audio.h:
	  audio: some more accessor macros for GstAudioInfo

2011-09-22 15:34:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: fix documentation typo

2011-09-21 13:54:27 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Update common to 0.11 branch

2011-09-21 13:31:35 +0200  Edward Hervey <bilboed@bilboed.com>

	* win32/common/libgstaudio.def:
	  win32: Update .def files

2011-09-19 18:32:26 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* tests/check/elements/videorate.c:
	  videorate: Add tests for the max-rate case

2011-09-19 18:31:07 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* tests/check/elements/videorate.c:
	  videorate: Print which caps didn't match up

2011-09-19 18:26:04 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	* gst/videorate/gstvideorate.h:
	  videorate: Add a max-rate property
	  In various use-case you want to dynamically change the framerate (e.g.
	  live streams where the available network bandwidth changes). Doing this
	  via capsfilters in the pipeline tends to be very cumbersome and racy,
	  using this property instead makes it very painless.

2011-09-01 17:05:23 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* tests/check/elements/videorate.c:
	  videorate: Add test for caps negotiation

2011-09-01 16:47:49 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: Add more strict caps negotiation
	  When in drop-only mode we can never provide a framerate that is higher
	  then the input, so let the caps negotiation reflect this.

2011-09-20 13:35:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: don't unref event we don't own
	  http://bugzilla.gnome.org/show_bug.cgi?id=659562

2011-09-20 14:04:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Only check if this is a discarded type if we have fixed caps
	  For unfixed caps we will get here again later when the caps are fixed.

2011-09-20 14:03:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Only call autoplug-continue with fixed caps
	  With unfixed caps we can't reliably decide if the final caps
	  are going to be "raw" (e.g. supported by a sink) or not.
	  We will get here again later when the caps are fixed.

2011-09-20 13:45:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/decodebin2.c:
	  decodebin2: Fix unit test by strictly implementing parser behaviour instead of relying on basetransform

2011-01-13 15:35:30 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggstream.c:
	  oggstream: only use information from skeleton if we have nothing better
	  The codec setup headers are a lot more likely to have correct information,
	  especially as it's easy to remux a skeleton in a file where streams don't
	  have the same parameters (I've even seen a file with two skeletons).
	  Still, this is useful in the case we have a codec we can't decode, so we
	  can at least (theoretically) convert granpos to time, so we discard this
	  information if the codec setup has already provided it.
	  This fixes playback on (at lesat) the original archive.org encoding of
	  "The Night of the Living Dead" (now replaced by another encoding).
	  https://bugzilla.gnome.org/show_bug.cgi?id=612443

2011-09-19 14:16:19 +0200  Age Bosma <agebosma@gmail.com>

	* gst-libs/gst/pbutils/gstdiscoverer.h:
	  discoverer: Don't use gtk-doc /* < ... > */ style comments for signals
	  The /*< ... >*/ style is only used for public|protected|private,
	  signal comments use /* signals */. This prevents the some code
	  parsers/binding generators to be confused by the comment.

2011-09-19 14:02:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: Get the target of the video sinkpad, not the target sinkpad in the video setcaps handler

2011-08-18 15:13:23 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Initialize variable correctly
	  If subdrained isn't initialized to FALSE then a chain might think
	  that its group is drained when in fact it's not and this can cause
	  a switch too early or even cause a deadlock.

2011-07-28 16:44:33 +0000  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Rewrite EOS-handling code
	  This is now really threadsafe and improves switching
	  between different groups.

2011-09-19 11:53:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Fix non-prerolling pipelines and not-linked errors if a parser is available but no decoder
	  Fixes bug #658846.

2011-08-01 07:54:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspdefs.c:
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	  rtspdefs: add RTCP-Interval header

2011-09-19 11:24:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: Implement support for switching between raw and non-raw video streams

2011-09-19 09:34:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: Protect against accessing the NULL parent of the pads during shutdown
	  Fixes bug #658901.

2011-09-16 20:14:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: remove superfluous check in newsegment event handler
	  If we get a newsegment event from upstream, we can be quite
	  sure we're not operating pull-based.

2011-09-16 20:11:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: minor printf format fix

2011-09-14 12:23:19 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: fix wedge when seeking twice quickly in push mode
	  This could happen when testing with navseek, and pressing
	  right and left at roughly the same time. The current chain
	  is temporarily moved away, and this caused the flush events
	  not to be sent to the source pads, which would cause the
	  data queues downstream to reject incoming data after the
	  seek, and shut down, wedging the pipeline.
	  Now, I can't really decide whether this is a nasty steaming
	  hack or a good fix, but it certainly does fix the issue, and
	  does not seem to break anything else so far.
	  https://bugzilla.gnome.org/show_bug.cgi?id=621897

2011-08-13 14:18:56 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggdemux.h:
	  oggdemux: implement push mode seeking
	  This patch implements seeking in push mode (eg, over the net)
	  in Ogg, using the double bisection method.
	  As a side effect, it also fixes duration determination of network
	  streams, by seeking to the end to check the actual duration.
	  Known issues:
	  - Getting an EOS while seeking stops the streaming task, I can't
	  find a way to prevent this (eg, by issuing a seek in the event
	  handler).
	  - Seeking twice in a VERY short succession with playbin2 fails
	  for streams with subtitles, we end up pushing in a dataqueue
	  which is flushing. Rare in normal use AFAICT.
	  - Seeking is slow on slow links - byte ranges guesses could be
	  made better, decreasing the number of required requests
	  - If no granule position is found in the last 64 KB of a stream,
	  duration will be left unknown (should be pretty rare)
	  https://bugzilla.gnome.org/show_bug.cgi?id=621897

2011-09-15 22:04:56 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/playback/gstplaybin2.c:
	  playbin2: fix compiler warning
	  Remove a check for gchar >= 128

2011-09-15 16:47:26 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/adder/gstadder.c:
	  adder: don't access the event after pushing
	  Fixes valgrind warnings.

2011-09-15 14:27:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  Revert "playbin2: autoplug sink if stream is incompatible to the configured one"
	  This reverts commit b0b4e286c8cde2e79a959a444a2c68e99c3f29c6.
	  We agreed that the previous (pre-.35) behaviour is broken and a bug and the
	  current behaviour is correct, deterministic and allows the application to
	  handle stuff properly while the old behaviour can't be handled properly by
	  applications and just worked in some applications by luck.
	  The solution to the problem that was solved by relying on the old, broken
	  behaviour would be, to make decodebin2/playbin2 more aware of decoders and
	  improve the autoplugging of decoders by considering the caps supported by the
	  sink instead of just using something with the highest rank.
	  See bug #656923.

2011-09-15 09:23:54 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/playback/gstplaybin2.c:
	  playbin2: autoplug sink if stream is incompatible to the configured one
	  Fixes regression since 0.10.33 where sinks that can cope with non raw
	  caps or custom caps are not autoplugged if there's a sink configured
	  with the properties video-sink and audio-sink which cannot handle
	  the stream. This change checks for compatibility on the configured one
	  and use it if success. Otherwhise it tries with the found factories.

2011-08-13 14:14:19 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: do not propagate discontinuities in sparse streams
	  The first packet of a sparse stream may arrive after an initial
	  delay in the stream. If ogg_stream_packetout reports a discontinuity
	  in a sparse stream, do not propagate it to other streams in the
	  chain unnecessarily.
	  https://bugzilla.gnome.org/show_bug.cgi?id=621897

2011-09-12 15:48:59 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/playback/gstplaysink.c:
	  Revert "playsink: only add text overlay if vido sink also accepts raw caps"
	  This reverts commit a22faad18a73a27a2a0c903748c1a355df4d8c13. Instead
	  of disabling subtitles completelly when video stream have custom caps,
	  just let the sutbtileoverlay cope with them as now it's able to.

2011-09-12 15:46:46 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: gracefully handle non raw video streams
	  Implement handling of non raw video streams by avoiding colorspace
	  elements and autoplugging a compatible renderer if available. Fallback
	  to passthrough if no compatible renderer is found.

2011-09-12 15:10:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: try to catch malformed URIs
	  Only log in debug log for now, since the check is a bit
	  half-hearted, its purpose is mostly to make sure people
	  use gst_filename_to_uri() or g_filename_to_uri().
	  https://bugzilla.gnome.org/show_bug.cgi?id=654673

2011-09-12 19:53:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/tag.h:
	  docs: minor addition to GST_TAG_ID3V2_HEADER_SIZE docs

2011-09-11 14:22:59 -0400  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: Fix descriptions of properties

2011-09-10 18:30:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	  baseaudiosrc: don't try to fixate "width" field for alaw/mulaw
	  Fixes warning when trying to fixate e.g. pulsesrc ! audio/x-alaw ! fakesink.

2011-09-09 13:10:13 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/design/design-decodebin.txt:
	  docs: fix some typos in the decodebin design document

2011-09-09 13:07:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/interfaces/colorbalance.c:
	  colorbalance: add some guards to interface methods
	  https://bugzilla.gnome.org/show_bug.cgi?id=658584

2011-09-09 12:07:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: recognize Asylum modules
	  Note that there is already a AMF detection for a different
	  magic, I'm not sure if that's a different format with the
	  same initials or not. AMF is used for a few different formats
	  (including video), so...
	  This fixes playbin2 playing Asylum modules.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658514

2011-08-31 20:51:17 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/subparse/gstsubparse.c:
	  subparse: Improve subrip type check regex
	  This patch prevents timestamp like "1 1:00:00", which would have been seen
	  as hour 101 by our parser, and allow single digit hour, minute and seconds
	  as it's already supported by the parser, and also by other implementation
	  like in mplayer. This fixes bug 657872.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657872

2011-09-08 14:46:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/design/design-decodebin.txt:
	  decodebin: Update design documentation about how Parser/Converter are handled

2011-09-08 14:42:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  Revert "Revert "decodebin2: Do a subset check before actually using a factory""
	  This reverts commit 5f5d832a3bcff0828758f164fcb13c4258aefb36.

2011-09-08 14:42:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  docs/libs/Makefile.am
	  tests/check/elements/decodebin2.c

2011-09-08 13:25:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  Revert "decodebin2: Do a subset check before actually using a factory"
	  This reverts commit 50a88396ae6d54a83a10e7d2efd551d39033148e.
	  See bug #658541.

2011-09-07 16:44:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/decodebin2.c:
	  decodebin2: Don't use bufferalloc in the test elements
	  This will cause not-linked errors that usually don't happen
	  because normal decoders/parsers will set srcpad caps before
	  allocating buffers from downstream.

2011-09-07 16:43:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Make sure to fixate Parser/Converter caps before continuing autoplugging

2011-09-07 16:04:43 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst/playback/gstplaysink.c:
	  playsink: only add text overlay if vido sink also accepts raw caps
	  Fixes regression, pipeline fails with not negotiated, on media
	  containing subtitles when decoder/sink with custom caps is used.

2011-09-07 14:19:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Intersect the factory caps with the current caps for the capsfilter
	  Otherwise we'll include many incompatible caps in the capsfilter that
	  will only slow down negotiation.

2011-09-07 14:07:00 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/libs/Makefile.am:
	* 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.

2011-09-07 14:04:10 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiotestsrc/gstaudiotestsrc.h:
	  docs: add two mising enum docs

2011-09-07 14:10:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/audiorate.c:
	  audiorate: Use complete audio caps, including the endianness field

2011-09-07 12:32:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: fix element factory refcounting
	  g_value_get_object() does not give us our own ref.
	  Fixes "Trying to dispose object "flacparse", but it still has a parent "registry0".
	  You need to let the parent manage the object instead of unreffing the object directly."
	  and similar warnings.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658416

2011-09-07 11:06:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: do not automatically override quality when using target bitrate
	  If both quality and bitrate are set, libtheora will try to meet
	  both constraints, causing it to prefer emitting a smaller number
	  of good frames, to emitting the full number of frames that would
	  not meet the requested quality. This causes a slideshow effect
	  when the bitrate is low and the quality is high. And the default
	  theoraenc is high (48/63).
	  So only set quality when it is requested, and leave it unset
	  otherwise.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658443

2011-09-06 21:24:33 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From a39eb83 to 11f0cd5

2011-09-06 19:18:27 +0100  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>

	* gst-plugins-base.spec.in:
	  Add latest files to spec file

2011-09-06 20:13:30 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/libs/Makefile.am:
	  docs: activate overrides file to fix make distcheck

2011-09-06 16:42:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Merge remote-tracking branch 'origin/master' into 0.11
	  Merge in doc updates for audio enums from 0.10, and get rid
	  of the #if #else in the enum list, since that confuses gtk-doc.
	  Conflicts:
	  gst-libs/gst/audio/audio.c
	  gst-libs/gst/audio/audio.h

2011-09-06 16:46:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.h:
	  audio: rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN

2011-09-06 16:46:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	  audio/video add descriptions
	  Add a description to the audio and video format info in case we want to use this
	  later.

2011-09-06 15:46:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	  audio: update internal silent sample defines as well to match 0.11

2011-09-06 16:46:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.h:
	* gst/audioconvert/audioconvert.c:
	  rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN

2011-09-06 15:16:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	  audio: update audio format enums to match changes in 0.11
	  And add new audio format info stuff to docs.

2011-09-06 15:40:02 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 605cd9a to a39eb83

2011-09-06 15:31:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst/playback/gstsubtitleoverlay.c
	  tests/check/elements/decodebin2.c

2011-09-06 15:24:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst-libs/gst/audio/audio.h
	  gst-libs/gst/audio/gstaudiodecoder.c
	  gst-libs/gst/audio/gstaudiodecoder.h
	  gst-libs/gst/audio/gstaudioencoder.c
	  gst-libs/gst/audio/gstbaseaudioencoder.h
	  gst/playback/Makefile.am
	  gst/playback/gstplaybin.c
	  gst/playback/gstplaysink.c
	  gst/playback/gstplaysinkvideoconvert.c
	  gst/playback/gstsubtitleoverlay.c
	  gst/videorate/gstvideorate.c
	  gst/videoscale/gstvideoscale.c
	  win32/common/libgstaudio.def

2011-09-06 14:16:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Do a subset check before actually using a factory
	  This prevents autoplugging if the caps have a non-empty intersection
	  but are not accepted by the next element's pad.

2011-09-06 14:04:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: Use subset check instead of non-empty-intersection check to check if pads are compatible

2011-09-06 14:03:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: Use subset check instead of non-empty-intersection check to check if pads are compatible

2011-09-06 13:06:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Fix memory leak

2011-09-06 12:14:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/decodebin2.c:
	  decodebin2: Add unit test for correct parser/converter negotiation

2011-06-26 15:40:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Correctly negotiate format for parsers that can convert different stream formats
	  This is done by adding a capsfilter after every parser/converter that contains
	  all possible caps supported by downstream elements. A capsfilter is necessary
	  here because the decoder is only selected after the parser selected a format
	  and the parser can't know what downstream would support otherwise.

2011-09-05 15:19:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: If a audio/video sink was already selected don't check caps of all other possible sinks

2011-09-06 08:25:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/decodebin2.c:
	  decodebin2: Add Tim as author for the parser test

2011-09-06 12:06:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasrc.c:
	* ext/ogg/gstoggstream.c:
	* ext/vorbis/gstvorbisdeclib.h:
	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/cdda/gstcddabasesrc.c:
	* gst-libs/gst/riff/riff-media.c:
	* gst/adder/gstadder.c:
	* gst/audiorate/gstaudiorate.c:
	* gst/audioresample/gstaudioresample.c:
	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/volume/gstvolume.c:
	  audio: change audio format syntax a little
	  Remove the _ in front of the endianness prefix.
	  Remove the _3 postfix for the 24 bits formats.
	  Add a _32 postfix after the formats that occupy extra space beyond their
	  natural size.
	  The result is that the GST_AUDIO_NE() macro can simply append the endianness
	  after all formats and that we only specify a different sample width when it is
	  different from the natural size of the sample. This makes things more consistent
	  and follows the pulseaudio conventions instead of the alsa ones.

2011-09-06 10:07:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	* gst-libs/gst/audio/gstaudioencoder.h:
	  docs: more docs clean-ups

2011-09-05 23:00:30 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: don't take the object lock twice in {set,get}_property
	  https://bugzilla.gnome.org/show_bug.cgi?id=658294

2011-09-05 22:51:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/audio.h:
	  audio: fix GST_AUDIO_FORMAT_INFO_IS_*() macros to return a boolean

2011-09-05 21:40:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	* gst-libs/gst/audio/gstaudioencoder.h:
	  docs: some docs love

2011-09-05 20:45:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* docs/libs/gst-plugins-base-libs.types:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstaudioencoder.h:
	  docs: add GstAudioDecoder and GstAudioEncoder to documentation

2011-09-05 15:01:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstaudioencoder.h:
	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	* win32/common/libgstaudio.def:
	  audio: rename GstBaseAudioDecoder/Encoder to GstAudioDecoder/Encoder
	  API: gst_gst_audio_decoder_finish_frame()
	  API: gst_gst_audio_decoder_get_audio_info()
	  API: gst_gst_audio_decoder_get_byte_time()
	  API: gst_gst_audio_decoder_get_delay()
	  API: gst_gst_audio_decoder_get_latency()
	  API: gst_gst_audio_decoder_get_max_errors()
	  API: gst_gst_audio_decoder_get_min_latenc()y
	  API: gst_gst_audio_decoder_get_parse_state()
	  API: gst_gst_audio_decoder_get_plc()
	  API: gst_gst_audio_decoder_get_plc_aware()
	  API: gst_gst_audio_decoder_get_tolerance()
	  API: gst_gst_audio_decoder_get_type()
	  API: gst_gst_audio_decoder_set_byte_time()
	  API: gst_gst_audio_decoder_set_latency()
	  API: gst_gst_audio_decoder_set_max_errors()
	  API: gst_gst_audio_decoder_set_min_latency()
	  API: gst_gst_audio_decoder_set_plc()
	  API: gst_gst_audio_decoder_set_plc_aware()
	  API: gst_gst_audio_decoder_set_tolerance()
	  API: gst_gst_audio_encoder_finish_frame()
	  API: gst_gst_audio_encoder_get_audio_info()
	  API: gst_gst_audio_encoder_get_frame_max()
	  API: gst_gst_audio_encoder_get_frame_samples()
	  API: gst_gst_audio_encoder_get_hard_resync()
	  API: gst_gst_audio_encoder_get_latency()
	  API: gst_gst_audio_encoder_get_lookahead()
	  API: gst_gst_audio_encoder_get_mark_granule()
	  API: gst_gst_audio_encoder_get_perfect_timestamp()
	  API: gst_gst_audio_encoder_get_tolerance()
	  API: gst_gst_audio_encoder_get_type()
	  API: gst_gst_audio_encoder_proxy_getcaps()
	  API: gst_gst_audio_encoder_set_frame_max()
	  API: gst_gst_audio_encoder_set_frame_samples()
	  API: gst_gst_audio_encoder_set_hard_resync()
	  API: gst_gst_audio_encoder_set_latency()
	  API: gst_gst_audio_encoder_set_lookahead()
	  API: gst_gst_audio_encoder_set_mark_granule()
	  API: gst_gst_audio_encoder_set_perfect_timestamp()
	  API: gst_gst_audio_encoder_set_tolerance()
	  https://bugzilla.gnome.org/show_bug.cgi?id=642690

2011-08-03 13:31:59 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/encoding/gstencodebin.c:
	  encodebin: Select muxer further
	  Sort muxers based on their caps and ranking before iterating to
	  find one that fits the profile.
	  Sorting is done by putting the elements that have a pad template
	  that can produce the exact caps that is on the profile. For example:
	  when asking for "video/quicktime, variant=iso", muxers that
	  have this exact caps on their pad templates will be put first on
	  the list than ones that have only "video/quicktime".
	  https://bugzilla.gnome.org/show_bug.cgi?id=651496

2011-09-05 20:31:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Actually iterate over the factories instead of only taking the first one

2011-09-05 15:51:25 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/libs/profile.c:
	* tests/check/libs/tag.c:
	* tests/check/libs/video.c:
	  tests: supress ERROR log output for some tests
	  Be nice when we tests for correct error handling and don't spam stdout.

2011-09-05 14:40:24 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  Revert "playsink: Try include 'pitch', if no other sink is provided"
	  This reverts commit 105814e2c78f9867c61531b9e8166e4ae994296f.
	  The general consensus seems to be that we should revert this for
	  now. If such behaviour is desired, we should probably enable it
	  via a flag. And maybe use the scaletempo plugin instead.

2011-09-05 12:02:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: Don't leak the videochain ts-offset element
	  Also don't leak the audiochain ts-offset element if one is
	  found but the sink doesn't support volume settings.

2011-09-05 11:55:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: Use gst_object_unref() instead of g_object_unref() for better debugging

2011-03-17 19:13:58 -0700  David Schleef <ds@schleef.org>

	* gst/videoscale/Makefile.am:
	* gst/videoscale/gstvideoscale.c:
	* gst/videoscale/gstvideoscale.h:
	* gst/videoscale/vs_image.h:
	* gst/videoscale/vs_lanczos.c:
	  videoscale: Add modified Lanczos scaling method
	  Adds a Lanczos-derived scaling method, which is rather slow, but very
	  high quality.  Adds a few properties that can be used to tune various
	  scaling properties: sharpness, sharpen, envelope, dither.  Not currently
	  Orcified, but was designed with that in mind.

2011-05-16 14:46:52 -0700  David Schleef <ds@schleef.org>

	* gst/playback/Makefile.am:
	* gst/playback/gstplaybin.c:
	* gst/playback/gstplaysink.c:
	* gst/playback/gstplaysinkvideoconvert.c:
	* gst/playback/gstsubtitleoverlay.c:
	  playback: Add define for colorspace element
	  Single point of change if you want to switch from ffmpegcolorspace
	  to colorspace.

2011-08-25 15:14:58 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: fix dynamically changing average period
	  The average_period_set variable can be accessed in different threads, so
	  always lock it when reading. Furthermore when switching to averaging
	  mode we should make sure we don't have cached buffers that aren't used
	  in that mode. And any modeswitch will cause the latency to change, so we
	  should post a NewLatency message

2011-08-23 10:11:52 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/videorate/Makefile.am:
	* gst/videorate/gstvideorate.c:
	* gst/videorate/gstvideorate.h:
	  videorate: Port to basetransform

2011-08-22 15:52:57 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  Correct added versions

2011-08-31 14:45:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: Only unref ts_offset elements if they're not NULL

2011-08-31 13:32:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video.h:
	  video: improve docs a little

2011-08-31 12:39:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Keep the chain mutex locked while connecting to the notify::caps signal

2011-08-30 14:04:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video.h:
	  video: add some more macros

2011-08-30 18:21:31 +1000  Jan Schmidt <thaytan@noraisin.net>

	* tests/examples/seek/seek.c:
	  seek: Accept pipeline descriptions for audiosink/videosink
	  Make the element_factory_make_or_warn utility function try parsing
	  the input string as a bin if element_factory_make() fails. This makes
	  the --audiosink/--videosink commandline options accept a pipeline
	  string.

2011-08-30 18:21:31 +1000  Jan Schmidt <thaytan@noraisin.net>

	* gst/playback/gstplaysink.c:
	  playsink: Try include 'pitch', if no other sink is provided
	  As a default, try the pipeline 'pitch ! audioconvert ! autoaudiosink'
	  before trying plain autoaudiosink

2011-08-29 13:33:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: fix ts_offset refcounting

2011-08-29 13:28:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	* gst/playback/gstplaysinkaudioconvert.c:
	* gst/playback/gstplaysinkvideoconvert.c:
	  base: port to 0.11

2011-08-29 11:42:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	  audio: fix after merge

2011-08-29 11:38:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/pbutils/gstdiscoverer.h:
	  pbutils: port to new API

2011-08-29 11:37:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/ogg/gstoggmux.c
	  gst-libs/gst/audio/audio.c
	  gst-libs/gst/audio/audio.h
	  gst-libs/gst/audio/multichannel.h
	  gst-libs/gst/pbutils/Makefile.am
	  gst-libs/gst/pbutils/gstdiscoverer.c
	  gst/playback/gstplaysinkaudioconvert.c
	  gst/playback/gstplaysinkvideoconvert.c
	  win32/common/libgstaudio.def

2011-08-27 14:57:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  pbutils: don't depend on libgstvideo just to parse some caps
	  Let's extract those ints and fractions ourselves and not depend
	  on libgstvideo.

2011-08-27 13:31:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* win32/common/libgstaudio.def:
	  audio: add GstBaseAudioDecoder and GstBaseAudioEncoder to build
	  However, libgstaudio now depends on libgstvideo (via pbutils).
	  https://bugzilla.gnome.org/show_bug.cgi?id=642690
	  API: gst_audio_info_clear()
	  API: gst_audio_info_convert()
	  API: gst_audio_info_copy()
	  API: gst_audio_info_free()
	  API: gst_audio_info_from_caps()
	  API: gst_audio_info_init()
	  API: gst_audio_info_to_caps()
	  API: gst_base_audio_decoder_finish_frame()
	  API: gst_base_audio_decoder_get_audio_info()
	  API: gst_base_audio_decoder_get_byte_time()
	  API: gst_base_audio_decoder_get_delay()
	  API: gst_base_audio_decoder_get_latency()
	  API: gst_base_audio_decoder_get_max_errors()
	  API: gst_base_audio_decoder_get_min_latency()
	  API: gst_base_audio_decoder_get_parse_state()
	  API: gst_base_audio_decoder_get_plc()
	  API: gst_base_audio_decoder_get_plc_aware()
	  API: gst_base_audio_decoder_get_tolerance()
	  API: gst_base_audio_decoder_get_type()
	  API: gst_base_audio_decoder_set_byte_time()
	  API: gst_base_audio_decoder_set_latency()
	  API: gst_base_audio_decoder_set_max_errors()
	  API: gst_base_audio_decoder_set_min_latency()
	  API: gst_base_audio_decoder_set_plc()
	  API: gst_base_audio_decoder_set_plc_aware()
	  API: gst_base_audio_decoder_set_tolerance()
	  API: gst_base_audio_encoder_finish_frame()
	  API: gst_base_audio_encoder_get_audio_info()
	  API: gst_base_audio_encoder_get_frame_max()
	  API: gst_base_audio_encoder_get_frame_samples()
	  API: gst_base_audio_encoder_get_hard_resync()
	  API: gst_base_audio_encoder_get_latency()
	  API: gst_base_audio_encoder_get_lookahead()
	  API: gst_base_audio_encoder_get_mark_granule()
	  API: gst_base_audio_encoder_get_perfect_timestamp()
	  API: gst_base_audio_encoder_get_tolerance()
	  API: gst_base_audio_encoder_get_type()
	  API: gst_base_audio_encoder_proxy_getcaps()
	  API: gst_base_audio_encoder_set_frame_max()
	  API: gst_base_audio_encoder_set_frame_samples()
	  API: gst_base_audio_encoder_set_hard_resync()
	  API: gst_base_audio_encoder_set_latency()
	  API: gst_base_audio_encoder_set_lookahead()
	  API: gst_base_audio_encoder_set_mark_granule()
	  API: gst_base_audio_encoder_set_perfect_timestamp()
	  API: gst_base_audio_encoder_set_tolerance()

2011-08-27 13:15:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  docs: add since markers to baseaudio{decoder,encoder} documentation

2011-08-27 12:47:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudiodecoder, baseaudioencoder: fix some compiler warnings
	  Leaving the GST_USE_UNSTABLE_API guards in until some of the
	  ported decoders have been updated and it's clear that I didn't
	  mess up anywhere porting things to the new audio API.

2011-08-27 12:41:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioutils.c:
	* gst-libs/gst/audio/gstbaseaudioutils.h:
	  baseaudioutils: remove, merged into or superseded by audio.c

2011-08-27 12:39:50 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  baseaudioencoder: port to new GstAudioInfo API

2011-08-27 12:37:16 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  baseaudiodecoder: port to GstAudioInfo API

2011-08-27 11:43:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	  audio: add gst_audio_info_{init,clear} and gst_audio_info_{copy,free}

2011-08-22 20:15:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/multichannel.c:
	* gst-libs/gst/audio/multichannel.h:
	  audio: add GstAudioFormat, GstAudioFormatInfo and GstAudioInfo
	  Same as in 0.11, but with caps parsing/serialising for 0.10 style
	  caps. Add setting default channel positions.

2011-08-17 18:48:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: remove leftover experimental code

2011-08-17 18:32:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioutils.c:
	* gst-libs/gst/audio/gstbaseaudioutils.h:
	  audioutils: modify _parse, add GType support functions

2011-08-16 21:11:42 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  baseaudiodecoder: move properties to private storage and add _get/_set

2011-08-16 21:11:52 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	  baseaudiodecoder: rename property

2011-08-16 20:39:07 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  baseaudiodecoder: replace context helper structure by various _get/_set

2011-08-16 18:59:13 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  baseaudioencoder: move properties to private storage and add _get/_set

2011-08-16 18:25:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: rename some properties

2011-08-16 18:23:14 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  baseaudioencoder: replace context helper structure by various _get/_set

2011-08-16 17:27:07 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	* gst-libs/gst/audio/gstbaseaudioutils.c:
	* gst-libs/gst/audio/gstbaseaudioutils.h:
	  baseaudio: rename GstAudioState to GstAudioFormatInfo

2011-06-17 11:54:08 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: TEMP; avoid some imperfect ts jitter ?
	  ... even when not in perfect mode ?

2011-04-28 12:01:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: debug format fixes

2011-04-28 12:01:30 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	  baseaudiodecoder: debug format fix

2011-03-31 14:03:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	  baseaudiodecoder: fixup documentation

2011-03-29 15:51:40 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	  baseaudiodecoder: fix FLUSH_STOP actions

2011-03-28 13:16:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	  baseaudiodecoder: preserve upstream seek event seqnum

2011-03-22 11:09:56 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: use buffer running time for granule calculation

2011-03-22 10:45:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	  baseaudiodecoder: minor fix in ts resync

2011-03-21 11:40:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  baseaudiodecoder: improve glitch resilience
	  Provide a replacement for GST_ELEMENT_ERROR to avoid aborting at the first
	  atom out of place, while on the other hand not failing indefinitely.

2011-03-17 12:09:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  baseaudiodecoder: add limited legacy seeking support

2011-03-16 14:41:40 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	  baseaudiodecoder: cater for audio-codec tag

2011-03-10 16:01:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  baseaudiodecoder: initial version

2011-03-16 18:41:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: misc fixes

2011-03-15 17:27:42 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	* gst-libs/gst/audio/gstbaseaudioutils.c:
	* gst-libs/gst/audio/gstbaseaudioutils.h:
	  baseaudio: add audioutils for caps and query handling helper utils

2011-03-14 12:39:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  baseaudioencoder: mark unstable API

2011-03-10 15:12:54 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: fix clearing context

2011-03-10 15:12:19 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  baseaudioencoder: simplify latency variable handling

2011-03-10 14:28:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  baseaudioencoder: minor fixes and code simplifications
	  Also modify and elaborate a bit on pre_push (though currently unused to no harm).

2011-03-09 12:44:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: additional documentation on granule semantics and configuration

2011-03-09 12:24:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: elaborate property names

2011-03-09 12:22:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  baseaudioencoder: rename state field xint to is_int

2011-03-09 12:18:56 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	  baseaudioencoder: gtk-doc syntax fixes

2011-03-09 12:17:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  baseaudioencoder: minor fix and cleanup

2011-03-01 14:08:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  baseaudiocodec: ... and also rename to baseaudiodecoder

2011-03-01 13:58:31 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  gst-libs/gst/audio: Remove baseaudiodecoder
	  Adds little beyond baseaudiocodec (seeking, bit of query), and what it adds
	  is mainly out-of-scope (e.g. decoder seeking, should be done by upstream
	  demuxer/parser) and/or based on non-prime example (mad).

2009-09-17 13:26:28 +0200  Iago Toral <itoral@igalia.com>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	  baseaudiodecoder: Return TRUE if we run into special conversion cases.

2009-09-01 14:17:53 +0200  Iago Toral <itoral@igalia.com>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  audio: initial version of GstBaseAudioCodec
	  Moved most of the code to GstBaseAudioCodec, GstBaseAudioDecode is
	  now really small, maybe we do not really need it (or its encoder
	  counterpart). Added more API for subclasses and documentation.

2009-08-14 09:45:52 +0200  Iago Toral <itoral@igalia.com>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  Added src_queries to decoder class. Added handle_discont to decoder class. Reworked reset. Various other minor fixes.

2009-08-06 15:28:00 +0200  Iago Toral <itoral@igalia.com>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  Added a draft implementation of gstbaseaudiodecoder

2011-03-01 11:56:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiodecoder.c:
	* gst-libs/gst/audio/gstbaseaudiodecoder.h:
	  Added audio directory for audio codec base classes

2011-02-18 16:38:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  audioencoders: add streamheader helper utility

2011-01-27 16:52:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudioencoder.h:
	  audioencoders: baseaudioencoder and ported encoders

2011-08-26 14:20:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videoconvert/gstvideoconvert.c:
	* gst/videoscale/gstvideoscale.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.c:
	  base: fix for allocation methods rename

2011-08-26 10:03:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* win32/common/libgstpbutils.def:
	  win32: Add new discoverer API

2011-08-26 10:03:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: Add new discoverer API

2011-08-24 16:29:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/pbutils/gstdiscoverer.h:
	* gst-libs/gst/pbutils/pbutils-private.h:
	* tools/gst-discoverer.c:
	  discoverer: retrieve audio track language from tags too
	  https://bugzilla.gnome.org/show_bug.cgi?id=657257

2011-08-24 15:09:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: consider subtitles as raw
	  Otherwise, discoverer will generated an "inner" codec
	  where there can be a tranformation (eg, kate -> DVD SPU,
	  and various ->text/x-pango-markup).
	  https://bugzilla.gnome.org/show_bug.cgi?id=639055

2011-08-24 15:05:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: add application/x-kate to subtitles caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=639055

2011-08-24 14:59:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: get language from other tags if we did not get it already
	  https://bugzilla.gnome.org/show_bug.cgi?id=639055

2011-08-24 15:04:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/pbutils/gstdiscoverer.h:
	* gst-libs/gst/pbutils/pbutils-private.h:
	* tools/gst-discoverer.c:
	  discoverer: add subtitles API
	  https://bugzilla.gnome.org/show_bug.cgi?id=639055

2011-08-21 14:51:45 -0700  David Schleef <ds@schleef.org>

	* gst/playback/gstplaysink.c:
	  playback: reference count ts_offset
	  Apparently this object is being used after it's freed.  This is one
	  way to fix it, although perhaps not the best way.  Fixes: #656715.

2011-08-25 17:41:53 +0200  Edward Hervey <bilboed@bilboed.com>

	* win32/common/libgstaudio.def:
	* win32/common/libgstinterfaces.def:
	* win32/common/libgsttag.def:
	* win32/common/libgstvideo.def:
	  win32: Update .def files

2011-08-25 17:41:30 +0200  Edward Hervey <bilboed@bilboed.com>

	* win32/common/_stdint.h:
	* win32/common/audio-enumtypes.c:
	* win32/common/audio-enumtypes.h:
	* win32/common/config.h:
	* win32/common/interfaces-enumtypes.c:
	* win32/common/video-enumtypes.c:
	* win32/common/video-enumtypes.h:
	  win32: Update pre-generated files

2011-08-25 17:41:11 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst-libs/gst/video/video.h:
	  video: Fix typo in interlaced flag (TTF => TFF)

2011-08-25 16:41:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video.h:
	  video: clean up the custom flags
	  Clean up the flags, make an enum of them. We can now do this because there are
	  no subclasses of buffer anymore.

2011-08-25 16:30:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/convertframe.c:
	  convert: use new caps

2011-08-25 14:55:14 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: fix caps leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=657333

2011-07-08 23:06:46 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst-libs/gst/rtp/gstbasertppayload.c:
	  basertppayload: Make perfect timestamps reproducible across element restart
	  Without the perfect timestamp machinery, the RTP timestamp can be
	  computed directly from the running time of a buffer, but the perfect
	  timestamp patch broke that assumption. This patch restores it by
	  having the first perfect timestamp be the running time of that buffer
	  and counting from there.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=654434

2011-08-25 13:21:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	  audiotestsrc: use base class fill method

2011-08-24 17:39:11 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	  oggmux: fix leaks in skeleton writing
	  https://bugzilla.gnome.org/show_bug.cgi?id=563251

2011-08-18 16:36:23 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggmux.h:
	  oggmux: generate message headers from received tags
	  Some message headers can be deduced from tags (eg, "Language").
	  https://bugzilla.gnome.org/show_bug.cgi?id=563251

2011-08-18 10:05:17 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggparse.c:
	  ogg: use memory slices where appropriate
	  While there, avoid zeroing newly allocated memory where unnecessary
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-24 18:39:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/multichannel.h:
	* gst-libs/gst/riff/riff-media.c:
	  multichannel: add some more channels

2011-08-24 16:40:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/video/video.h:
	  audio/video: add format of the pack functions
	  Replace the unpack_size with an unpack_format, which is more descriptive of the
	  kind of data the unpack function will create.

2011-08-24 14:13:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	  audio: rename UNPOSITIONED to DEFAULT_POSITIONS
	  Rename the UNPOSITIONED flag to the DEFAULT_POSITIONS flag because that is
	  really what the resulting GstAudioInfo will contain as the chanel mappings.

2011-08-24 14:05:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkaudioconvert.c:
	* gst/playback/gstplaysinkvideoconvert.c:
	  playsink{audio,video}convert: Send NEWSEGMENT events to sinkpads instead of pushing them

2011-08-24 13:52:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/gstmetavideo.c:
	* gst-libs/gst/video/gstmetavideo.h:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	  video: Add an id to the video frame
	  Rename @view_id to @id.
	  Add an id to the video metadata. Add a method to get the metadata from a buffer
	  with the given id.
	  Make a method to map a frame with a certain id. This only maps the frame with
	  the given id on the video metadata. The generic frame id can be used when a
	  buffer carries multiple video frames such as in multiview mode but maybe also
	  when dealing with interlaced video that stores the fields in separate buffers.

2011-08-24 11:05:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	  audiotestsrc: fix build

2011-08-24 11:04:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/ogg/gstoggmux.c
	  ext/vorbis/gstvorbisenc.c

2011-08-23 11:12:10 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: do not warn when reaching EOS while scanning for the end chain
	  After all, we were asking for it.
	  This gets rid of the last warning-about-expected-condition.
	  w00t.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-23 11:08:25 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: add media type to chain information reports
	  One more little step in making logs a little less abstruse.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-23 11:05:11 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: correctly identify skeleton EOS packet
	  It is 0 byte, and was triggering the "bad packet" logic.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-23 10:58:20 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: do not warn about expected occurences
	  In this case, finding a skeleton packet.
	  Once upon a time, it used to be rare indeed, but no more.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-23 10:47:53 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: do not warn when finding a non BOS page
	  After all, we do hope to find actual data for these streams.
	  However, warn if we could not set up a chain when we find a
	  non BOS page, as that means we don't have a valid Ogg stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-23 10:40:12 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: rename local variable for clarity
	  While the casual reader might end up bewildered by just why this
	  change might increase clarity, it just happens than, in the libogg
	  and associated sources, op is the canonical name for an ogg_packet
	  whlie og is the canonical name for an ogg_page, and reading this
	  code confuses me.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-23 10:32:36 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: do not try to determine duration of header packets
	  Headers are inherently durationless.
	  Instead, set duration to 0 to avoid increasing tracked granpos,
	  and do not warn about it, since it is totally expected.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-23 10:29:49 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: include stream type in warnings
	  It makes it easier to work out what's going on.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-23 10:28:33 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: set skeleton stream media type to application/x-ogg-skeleton
	  This is to match the typefinder, and to make logs clearer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657151

2011-08-17 17:09:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggmux.h:
	  oggmux: add skeleton write support
	  Version written is 3.0
	  Base times are left empty for now.
	  Content-Type should be the MIME type of the stream. It is set to
	  the GStreamer media type for now, which is probably the same for
	  the streams oggmux supports.
	  https://bugzilla.gnome.org/show_bug.cgi?id=563251

2011-08-23 20:34:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/theora/gsttheoradec.c:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	  video: fix chroma-site enums

2011-08-23 19:23:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video.c:
	  video: avoid gst-indent breaking the code

2011-08-23 19:04:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video.h:
	  video: fix docs

2011-08-23 18:57:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/theora/gsttheoradec.c:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	* gst/playback/gstsubtitleoverlay.c:
	* gst/videoconvert/gstvideoconvert.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.c:
	  video: add colorimetry info
	  Make enums for the chroma siting for easier use in the videoinfo.
	  Make enums for the color range, color matrix, transfer function and the
	  color primaries. Add these values to the video info structure in a Colorimetry
	  structure. These values define the exact colors and are needed to perform
	  correct colorspace conversion. Use a couple of predefined colorimetry specs
	  because in practice only a few combinations are in use.
	  Add view_id to the video frames to identify the view this frame represents in
	  multiview video.
	  Remove old gst_video_parse_caps_framerate, use the videoinfo for this.
	  Port elements to new colorimetry info.
	  Remove deprecated colorspace property from videotestsrc.

2011-08-22 14:56:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: do not skip sparse streams when determining start times
	  This fixes demuxing of streams containing only sparse streams,
	  which would cause an infinite loop in _read_end_chain.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657062

2011-08-22 14:55:59 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: do not ignore sparse streams' start time
	  But do not wait for them either, if we don't have a packet for them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657062

2011-07-21 17:16:26 -0400  Monty Montgomery <cmontgom@redhat.com>

	* ext/vorbis/gstvorbisenc.c:
	  vorbisenc: Relax overly-tight jitter tolerances in gstvobisenc
	  vorbisenc currently reacts in a rater draconian fashion if input
	  timestamps are more than 1/2 sample off what it considers ideal. If data
	  is 'too late' it truncates buffers, if it is 'too soon' it completely
	  shuts down encode and restarts it.  This is causingvorbisenc to produce
	  corrupt output when encoding data produced by sources with bugs that
	  produce a smple or two of jitter (eg, flacdec)

2011-08-22 16:21:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/vorbis/gstvorbisdec.c:
	* gst/audiotestsrc/gstaudiotestsrc.c:
	  audio: use convert audio helper

2011-08-22 16:11:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstringbuffer.c:
	  audio: move function to convert

2011-08-22 15:57:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* docs/design/draft-media-types.txt:
	* gst-libs/gst/video/gstmetavideo.h:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	  video: parse number of views
	  Parse the number of views in multiview video buffers.

2011-08-22 13:14:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/pango/gsttextoverlay.c

2011-08-22 13:06:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  gst-libs/gst/interfaces/videooverlay.c
	  gst-libs/gst/rtp/gstrtpbuffer.c
	  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/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

2011-08-22 12:22:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstogmparse.c:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/pbutils/descriptions.c:
	* gst-libs/gst/riff/riff-media.c:
	* gst-libs/gst/video/video.h:
	* tests/check/Makefile.am:
	* tests/check/elements/decodebin.c:
	  fourcc: remove fourcc
	  Remove fourcc in caps.
	  Fix pbutils descriptions.
	  Add more video macros
	  Fix some unit test

2011-08-22 12:21:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	  oggmux: fix compilation

2011-08-22 09:06:53 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: fix text buffer leak
	  Make sure to always unref the input text buffer.
	  Reported by bcxa.sz@gmail.com.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657049

2011-08-20 19:46:31 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst-libs/gst/video/gstvideosink.h:
	  docs: fix xref for the property

2011-08-20 19:16:42 +0200  Stefan Kost <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiofilter.c:
	* gst-libs/gst/interfaces/colorbalance.c:
	* gst-libs/gst/interfaces/mixer.c:
	* gst-libs/gst/interfaces/navigation.c:
	* gst-libs/gst/interfaces/streamvolume.h:
	* gst-libs/gst/interfaces/xoverlay.c:
	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	* gst-libs/gst/pbutils/install-plugins.h:
	* gst-libs/gst/rtp/gstrtpbuffer.c:
	* gst-libs/gst/rtsp/gstrtsptransport.c:
	* gst-libs/gst/rtsp/gstrtspurl.c:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	* gst-libs/gst/video/gstvideosink.h:
	  docs: handle warnings emitted by gtk-doc
	  This is useful and in most cases someone had put arbitrary markup into the docs,
	  misspelled xref'ed symbols, forgot to add stuff to the docs etc..

2011-08-20 17:53:11 +0200  Stefan Kost <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: partially revert my last commit
	  Somehow this was already there, but I missed that commit.

2011-08-20 14:11:11 +0200  Stefan Kost <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/tag/licenses.c:
	  docs: add new taglicense docs and clean them up
	  Avoid ugly docbook tags unless needed.

2011-08-20 12:37:10 +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/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.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: update for new translatable string

2011-08-20 12:36:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/Makefile.am:
	  tag: fix distcheck issue
	  Dist licenses dict.

2011-08-20 10:49:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/riff/riff-media.c:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/gstaudioquantize.c:
	* gst/audioconvert/gstchannelmix.c:
	  audio: rename INT -> INTEGER
	  Spell INTEGER fully instead of using the int abreviation.
	  Remove some old functions.

2011-08-19 17:41:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/alsa/gstalsa.c:
	* ext/ogg/gstoggstream.c:
	* ext/vorbis/gstvorbisenc.c:
	* gst/audioconvert/channelmixtest.c:
	* gst/encoding/gstencodebin.c:
	  more audio caps porting

2011-08-19 17:05:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	  adder: port to new caps

2011-08-19 17:05:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/seek/seek.c:
	  seek: fix playbin2 setup

2011-08-19 16:49:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/libvisual/Makefile.am:
	* ext/libvisual/visual.c:
	  visual: port some more to new audio caps

2011-08-19 16:01:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/riff/riff-media.c:
	  riff: port to new audio caps

2011-08-19 16:00:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	  audio: add function to build audio format

2011-08-19 14:07:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11

2011-08-19 11:55:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	  audio: add more macros

2011-08-19 10:06:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst/volume/gstvolume.c:
	  audiofilter: Pass a const pointer to the audio format info to ::setup()
	  It is not meant to be changed by the subclass.

2011-08-18 16:20:57 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggparse.c:
	  ogg: do not use 32 bit modifiers to print serial numbers
	  If ints are 64 bits, 32 bits should get promoted in varargs anyway,
	  and we don't care about 16 bit ints.
	  This makes the code a lot more readable, and still gets us nice
	  hexadecimal 32 bit serialnos.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-18 19:36:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	  Merge branch 'master' into 0.11
	  Conflicts:
	  ext/ogg/gstoggmux.c
	  gst/playback/gstplaysink.c

2011-08-18 19:15:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasink.h:
	* ext/alsa/gstalsasrc.c:
	* ext/alsa/gstalsasrc.h:
	* ext/vorbis/gstvorbisdec.c:
	* ext/vorbis/gstvorbisdec.h:
	* ext/vorbis/gstvorbisdeclib.c:
	* ext/vorbis/gstvorbisdeclib.h:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiofilter.c:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/audio/gstaudioiec61937.c:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	* gst-libs/gst/audio/gstringbuffer.c:
	* gst-libs/gst/audio/gstringbuffer.h:
	* gst-libs/gst/audio/multichannel.c:
	* gst-libs/gst/audio/multichannel.h:
	* gst-libs/gst/video/video.h:
	* gst/adder/gstadder.c:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstaudioconvert.h:
	* gst/audioconvert/gstaudioquantize.c:
	* gst/audioconvert/gstchannelmix.c:
	* gst/audiorate/Makefile.am:
	* gst/audiorate/gstaudiorate.c:
	* gst/audiorate/gstaudiorate.h:
	* gst/audioresample/Makefile.am:
	* gst/audioresample/gstaudioresample.c:
	* gst/audioresample/gstaudioresample.h:
	* gst/audiotestsrc/Makefile.am:
	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/audiotestsrc/gstaudiotestsrc.h:
	* gst/playback/gstrawcaps.h:
	* gst/volume/gstvolume.c:
	  audio: rework audio caps.
	  Rework the audio caps similar to the video caps. Remove
	  width/depth/endianness/signed fields and replace with a simple string
	  format and media type audio/x-raw.
	  Create a GstAudioInfo and some helper methods to parse caps.
	  Remove duplicate code from the ringbuffer and replace with audio info.
	  Use AudioInfo in the base audio filter class.
	  Port elements to new API.

2011-07-27 11:05:31 +0000  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: Reconfigure when pads are added later
	  Instead of just assuming all pads are created at the same time,
	  remember which ones are actually new (via ->pending_blocked_pads).
	  This allows the following use-case to properly work:
	  * Upstream starts with audio-only
	  * Only that pad gets data, blocks and a real audio sink is created
	  * Upstream laters adds a video stream
	  * A new pad is requested, blocks and reconfiguration kicks in in
	  order to add a new real video sink

2011-08-18 09:37:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/README:
	  ogg: get the operator precedence right, even if only a doc
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-18 09:30:46 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: vorbis has a preroll of 2
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-17 19:40:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstoggstream.h:
	  oggstream: new convenience function to get a stream's media type
	  This will make logging a lot clearer, both in code and in output.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-17 18:48:54 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggmux.h:
	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstoggstream.h:
	  ogg: move the "always flush page" to oggstream
	  It avoids checking for specific media types in the muxer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-17 18:38:39 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	  oggmux: use oggstream to decide which BOS packets to place first
	  Ogg recommends video BOS packets to be first.
	  Use the "is_video" flag in oggstream to select those, rather than
	  check for known mime types.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-17 18:03:16 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggstream.h:
	  ogg: rationalize serialno type to guint32
	  It is a 32 bit unsigned number.
	  Sure, the libogg API uses a long, but that's an unfortunate oversight.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-17 17:39:18 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	  oggmux: factor the header packet creation code
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-17 17:18:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	  oggmux: headers should always have granpos 0
	  https://bugzilla.gnome.org/show_bug.cgi?id=656775

2011-08-18 09:48:16 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioresample/resample.c:
	  audioresample: fix build without orc
	  https://bugzilla.gnome.org/show_bug.cgi?id=656781

2011-08-17 17:24:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>