=== release 0.10.36 ===

2012-02-20  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  releasing 0.10.36, "Better"

2012-02-20 23:19:49 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/ca.po:
	* po/id.po:
	  po: update translations

2012-02-17 15:08:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* win32/common/libgstaudio.def:
	  docs: add new audio base class API to docs and .def file

2012-01-30 15:55:26 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: only send new data immediately if there are no queued messages
	  Even if watch->messages->length is 0 there may still be some
	  data from a message that was only written partially at the
	  previous attempt stored in watch->write_data, so check for
	  that as well. We don't want to write data into the middle
	  of another message, which could happen when there wasn't
	  enough bandwidth.
	  https://bugzilla.gnome.org/show_bug.cgi?id=669039

2012-02-16 12:19:20 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	  audiodecoder: add some properties to tweak baseclass behaviour
	  ... so subclass can also rely upon never being bothered with some NULL buffer
	  it can't do any interesting with, or with any data before it received
	  any format configuration (and setup properly).

2012-02-16 12:18:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstaudioencoder.h:
	  audioencoder: add some properties to tweak baseclass behaviour
	  ... so subclass can also rely upon never being bothered with less data
	  than it desires or with some NULL buffer it can't do any interesting with.

2012-02-16 12:15:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: assert some more that subclass parsed frame has proper len

2012-02-14 19:23:27 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: try harder to obtain a duration if we don't get one right away
	  If we don't get a duration right away, set the pipeline to playing
	  and sleep a bit, then try again. This is ugly, but the least worst
	  we can do right now. The alternative would be to make parsers etc.
	  return some bogus duration estimate even after only having pushed
	  a single frame, for example.
	  Fixes discoverer showing 0 durations for some mp3 and aac files
	  (e.g. soweto-adts.aac).

2012-02-05 13:55:40 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/_stdint.h:
	* win32/common/config.h:
	  0.10.35.3 pre-release

2012-02-01 15:28:45 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstoggstream.h:
	  oggdemux: fix granpos interpolation violating max keyframe distance
	  In case many packets fit on a page, we may not see a granpos for
	  a while, and granpos interpolation can wrap the 'frames since last
	  keyframe' part of the granpos, generating a granpos which is smaller
	  than what it should be.
	  This is fixed by detecting keyframe packets (at least for Theora),
	  and updating the last keyframe granpos from this.
	  This may still be generating potentially wrong granpos for streams
	  which have a Theora like granpos (keyframes, a max keyframe distance
	  and a count of frames since last keyframe), and which allow implicit
	  granules on packets. For these streams, a custom keyframe detection
	  routine should be plugged into their GstOggStream mapper.
	  https://bugzilla.gnome.org/show_bug.cgi?id=669164

2012-02-01 16:46:13 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vorbis/gstvorbisparse.c:
	  vorbisparse: pedantically recognize undefined headers too

2012-02-01 16:32:24 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vorbis/gstvorbisparse.c:
	  vorbisparse: fix header detection
	  It was matching non header packets.
	  This fixes various leaks, where buffers would be pushed onto a headers
	  list, but never popped.
	  Might also fix corruption as those buffers were dropped from the output
	  silently...
	  https://bugzilla.gnome.org/show_bug.cgi?id=669167

2012-01-23 09:28:18 -0800  David Schleef <ds@schleef.org>

	* gst-libs/gst/interfaces/propertyprobe.c:
	  propertyprobe: fix documentation

2012-01-18 14:58:08 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: do not try to deactivate an inactive group
	  A group may have failed to activate due to an error (for instance,
	  having set the URI to a non existent location in about-to-finish).
	  https://bugzilla.gnome.org/show_bug.cgi?id=666395

2012-01-17 16:05:41 +0200  Anssi Hannula <anssi.hannula@iki.fi>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: fix state change stall on PAUSED->READY->PAUSED
	  After a PAUSED->READY change the sink pads are currently not set to
	  blocking state. When the element is set back to PAUSED, the change will
	  be done asynchronously, but as the _pad_blocked_cb() callback is now not
	  called, the state change never completes.
	  Fix that by setting the sink pads to blocking state on a PAUSED->READY
	  change, which ensures that the _pad_blocked_cb() is called when needed
	  on any future READY->PAUSED change. The sink pads are already put to
	  blocking state on NULL->READY change, so this behavior is consistent.
	  Fixes bug #668097.

2012-01-19 16:40:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: avoid unlikely NULL dereference

2012-01-19 16:35:54 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/videoscale/vs_fill_borders.c:
	  videoscale: prevent implicit upgrade to integer type and sign extension

2012-01-19 16:35:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tools/gst-discoverer.c:
	  gst-discoverer: remove extraneous variable

2012-01-19 16:32:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: verify linking to overlay element

2012-01-19 16:32:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: avoid finding sink in NULL bin in corner case

2012-01-19 16:29:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/tag/gstexiftag.c:
	  tag: exif: add missing break

2012-01-17 18:19:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: initialize variable
	  ... to help out challenged compiler.

2012-01-16 11:43:25 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/alsa/gstalsasink.c:
	  alsasink: fix high sample rates being rejected
	  An ALSA sink may select a different rate (as we use the _set_rate_near
	  API, which is not guaranteed to set the exact target rate).
	  The rest of the code seems to already handle this well, as output
	  from a 88200 Hz file seems to have the correct pitch when selecting
	  a 96 kHz rate.

2012-01-16 11:40:47 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/alsa/gstalsasink.c:
	  alsasink: fix rate match message mistaking error code for sample rate

2012-01-13 16:57:15 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>

	* Android.mk:
	  Android, Add explicit path for zlib
	  This change fixes building gst-libs/gst/tag/ code with
	  the Android buildsystem.

2012-01-13 14:50:49 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>

	* ext/vorbis/gstvorbisdec.c:
	  Fix wrong access to undefined struct member
	  For the USE_TREMOLO case, GstVorbisDec doesn't have
	  a vb member. Besides, Tremolo's vorbis_dsp_synthesis()
	  expects a vorbis_dsp_state to be passed as first
	  argument. Not a vorbis_block.

2012-01-13 14:47:13 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>

	* ext/vorbis/gstvorbisdec.c:
	  Fix TREMELO -> TREMOLO typo

2012-01-12 16:24:01 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraparse.c:
	  theoraparse: fix array leak

2012-01-12 14:26:05 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: fix structure leak
	  I hit the 'misc' one, but let's also make sure the topology
	  one get freed as well, though I do not know if this can happen
	  twice.

2012-01-11 20:47:00 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>

	* gst-libs/gst/video/Makefile.am:
	  Add missing DEFAULT_INCLUDES on androgenizer call
	  Fix building of the libgstvideo module on Android by adding the
	  missing and needed $(DEFAULT_INCLUDES) to CFLAGS for the
	  androgenizer call on gst-libs/gst/video/Makefile.am
	  Before this change, building was failing due to gst-plugins-base/
	  and gst-plugins-base/gst-libs/gst/video being left out of the
	  include path.

2012-01-11 16:17:42 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: fix push mode chain leak
	  When I first implemented push mode seeking, I removed the chain
	  freeing there as it could be used later. The current code does not
	  seem to do that though, so I'm restoring the previous freeing,
	  which plugs the leak while apparently not reintroducing use of
	  freed data with chained and normal files, both with gst-launch
	  playbin2 and Totem.

2012-01-11 12:52:17 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	  discoverer: fix leaks caused by some base class dtors not being called

2012-01-11 12:16:28 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: fix caps and discoverer object ref leaks

2012-01-11 11:55:59 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: add a few consts where appropriate

2012-01-11 11:55:36 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: fix pad leak

2012-01-10 18:27:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: use GST_TYPE_TAG_LIST for tag lists
	  They may not be structures in 0.11/1.0.

2012-01-10 18:07:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: fix potential tag list leaks
	  Not that I have ever seen these in practice, but if they
	  can't happen we may just as well just assign the new tag
	  list. Merge properly to be on the safe side, and also
	  avoid a useless tag list copy in the normal case where
	  there is no tag list yet.

2012-01-10 17:48:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: fix potential caps leak
	  in last else chunk.

2012-01-10 16:57:04 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: fix tag list leak

2012-01-10 16:51:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: fix pad leak

2012-01-10 16:14:29 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: fix hang on small truncated files
	  A first hang was happening when trying to locate a page backwards,
	  where we'd sync forever on the same page.
	  With that fixed, a second hang would happen after preparing an EOS
	  event, but with no chain created yet to send it to, the pipeline
	  would stay idle forever.
	  An element error is now emitted for this case.

2012-01-09 12:31:02 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/playback/gstplay-enum.h:
	  playback: document DEINTERLACE flag

2011-12-16 15:27:24 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: assume live stream if byte size cannot be determined
	  This prevents trying to seek and failing, then ending up unable
	  to stream because we can't get back at the headers.
	  A more robust way would be to find a good place to reinject the
	  headers when a seek fails, but I can't seem to get this to work.

2012-01-07 20:12:17 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: make hostname lookup more thread-safe
	  Don't write IP number string to return into a static
	  array which is shared amongst all threads (note: of
	  course a copy is returned).
	  https://bugzilla.gnome.org/show_bug.cgi?id=666711

2012-01-07 19:39:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: make is_subtitle_caps thread-safe

2011-11-01 17:57:59 +0100  Havard Graff <havard.graff@tandberg.com>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	* gst-libs/gst/tag/tags.c:
	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/encoding/gstsmartencoder.c:
	* gst/playback/gstplaysink.c:
	* tools/gst-discoverer.c:
	  Fix various unlikely, but still potential memoryleaks in error code paths
	  https://bugzilla.gnome.org/show_bug.cgi?id=667311

2011-10-22 16:41:23 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: implement get_caps vfunc
	  This allows downstream elements to query what caps are available.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667312

2012-01-05 12:23:08 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tools/gst-discoverer.c:
	  tools: avoid unportable vararg macro construct in gst-discoverer
	  https://bugzilla.gnome.org/show_bug.cgi?id=667306

2012-01-01 20:44:08 +0100  Idar Tollefsen <itollefs@cisco.com>

	* configure.ac:
	  build: Run platform check for platform specific configuration.

2011-10-12 11:28:10 +0200  Pascal Buhler <pabuhler@cisco.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	  rtcpbuffer: prevent overflow of 16bit header length.
	  RTCP header can be  (2^16 + 1) * 4 bytes long, so when validating a bogus
	  packet it was possible to get a 16bit overflow resulting in a length of 0.
	  This would put the gst_rtcp_buffer_validate_data function in a endless loop.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667313

2011-09-24 14:05:42 +0200  Havard Graff <havard.graff@tandberg.com>

	* gst/videotestsrc/videotestsrc.c:
	  videotestsrc: keep the calculation fixed-point
	  https://bugzilla.gnome.org/show_bug.cgi?id=667315

2011-08-04 11:30:05 +0200  Idar Tollefsen <itollefs@cisco.com>

	* ext/pango/gstclockoverlay.c:
	* ext/pango/gsttimeoverlay.c:
	  pango: changes includes from brackets to quotes for local files
	  https://bugzilla.gnome.org/show_bug.cgi?id=667316

2012-01-04 19:39:28 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 63d592e to cb5da59

2012-01-03 11:04:23 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: force fallback buffer_alloc when other pad not available
	  ... to avoid unnecessary spurious errors (upon e.g. shutdown).
	  If a real error is applicable in this unusual circumstance (missing other pad),
	  other (STREAM_LOCK protected) call paths can take care of that.

2012-01-03 11:02:17 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: avoid crashing when operating on released pad

2011-12-27 14:37:26 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/ogg/gstoggmux.c:
	  oggmux: fix leak when initializing pads
	  Pads are initialized twice: when requesting pads and when
	  initializing collectpads. Avoid double initialization by
	  checking if collectpads are still going to be initialized when
	  creating request pads.

2011-12-23 22:51:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: fix template caps creation on big endian systems

2011-12-23 22:24:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/gstexiftag.c:
	* tests/check/libs/tag.c:
	  tag: fix writing of Exif tag payloads <= 4 bytes
	  When the payload for an Exif tag is less than or equal to 4 bytes,
	  the data is simply put into the offset field. Fix writing these
	  kinds of payloads on big endian systems (and possibly also on
	  little endian systems). The caller will have already formatted
	  the bytes in memory according to the writer's endianness, so just
	  write out the bytes as they are in this case. Fixes tags unit test
	  on big endian systems.

2011-12-22 16:54:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: add a few more debug statements

2011-12-22 16:53:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	  audiodecoder: tweak documentation

2011-12-22 07:53:39 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst-libs/gst/tag/gstxmptag.c:
	* tests/check/libs/tag.c:
	  tag: xmp: Keep compatibility with our old generated xmp
	  We used to add a trailing \n to the end of generated xmp packets.
	  Windows viewer was unhappy with it and we fixed it in
	  96d2120c2bb0b29e1849098198f5fbef81939cdd
	  The problem is that this caused xmp generated before this fix
	  to not be recognized and parsed anymore. This patch makes it
	  recognize xmp with the trailing \n and without, fixing the
	  regression. Also adds tests for it.

2011-12-14 16:34:39 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/video/video-blend.c:
	  gstvideo: fix a RGB ordering mixup in colorspace conversion code

2011-12-20 12:42:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	  audiodecoder: set a non-zero default maximum tolerated errors
	  Whereas the previous default 0 was backwards compatible in that it lead
	  to erroring out immediately upon any error, elements that are really
	  ported and using the base class error macro can be assumed to intend to
	  improve behaviour rather than maintaining the old one.  So, make it easy
	  on those and any future one and tolerate some errors by default, as intended.
	  Fixes #666579.

2011-12-15 11:01:01 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst-libs/gst/tag/gstexiftag.c:
	  tag: exif: do not include \0 in size passed to g_convert
	  When using g_convert, we should only pass the length
	  of the string content (without the \0) as g_convert will
	  only parse the real contents when changing formats. Including
	  the \0 causes it to add another \0, increasing the string
	  size when not needed.
	  For example, when writting a North geo location ref entry, that should
	  be a string with a single N letter, it would write:
	  "N\0\0", causing the string to have size 3, instead of 2 as expected.
	  In our case, we can pass -1 and let g_convert calculate the strlen as
	  we don't use the length anywhere else.
	  This fixes jifmux's tests on gst-plugins-bad.

2011-10-03 14:51:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: tweak chain topology description
	  ... to also properly indicate chain's endpad if no elements are in the
	  chain (due to the endpad being a raw demuxer pad, or one setup without
	  decoders since uridecodebin or higher up decided not to need those).

2011-12-13 12:55:45 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: fix late buffer leak

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

2011-12-10 02:08:49 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* win32/common/_stdint.h:
	* win32/common/audio-enumtypes.c:
	* win32/common/config.h:
	* win32/common/gstrtsp-enumtypes.c:
	  0.10.35.2 pre-release

2011-12-10 01:36:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/LINGUAS:
	* po/cs.po:
	* po/eo.po:
	* po/es.po:
	* po/gl.po:
	* po/lv.po:
	* po/sr.po:
	  po: update translations

2011-12-09 15:39:12 +0000  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>

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

2011-12-09 01:31:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c:
	  typefindfunctions: only typefind text with a BOM as text/utf16 or text/utf32
	  We added the utf typefinder because the mp3 typefinder was a tad
	  overzealous when it came to typefinding things as mp3, and replaced
	  it with even more overzealous utf16/32 typefinders.
	  Fixes unit test.

2011-12-07 18:45:28 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/video/video-overlay-composition.c:
	* gst-libs/gst/video/video-overlay-composition.h:
	  video: make composition_blend() return a boolean
	  Not that anyone will ever check that, and it's not clear what
	  they're supposed to do if it fails, but at least it's there.

2011-12-07 18:31:58 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/video/video-overlay-composition.c:
	* gst-libs/gst/video/video-overlay-composition.h:
	  docs: add new API to docs

2011-12-07 17:57:08 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/video/video-overlay-composition.c:
	* gst-libs/gst/video/video-overlay-composition.h:
	* tests/check/libs/video.c:
	* win32/common/libgstvideo.def:
	  video: add seqnum getters for overlay compositions and rectangles
	  API: gst_video_overlay_composition_get_seqnum()
	  API: gst_video_overlay_rectangle_get_seqnum()

2011-11-23 15:45:57 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* gst-libs/gst/video/video.c:
	  video: support any type of video in _parse_caps
	  Slight change in semantics for convenience. Shouldn't cause any
	  problems since this function is usually only used on pre-filtered
	  caps and not random caps, and it's hard to imagine a situation
	  where someone would want to rely on the previous behaviour.

2011-12-06 21:57:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: don't leak previous buffer when shutting down
	  Implement stop vfunc after port to basetransform, so we
	  can clean up properly. Fixes make elements/videorate.valgrind

2011-12-06 20:30:55 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/libs/video.c:
	  tests: fix calculation of last pixel offset in video unit test
	  And check the right buffer (pix2) in one case.

2011-12-06 15:01:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/fft/Makefile.am:
	  examples: fix build of fft example
	  Should link against our own libgstfft-0.10.

2011-12-06 14:55:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/video/video.c:
	  video: fix leak in gst_video_format_new_template_caps()
	  g_value_reset() is not the same as g_value_unset()

2011-11-23 15:43:46 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: add suport for hardware accelerated videos
	  Don't plug converters for non-raw video.

2011-12-05 15:48:07 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/video/video-overlay-composition.c:
	  video: don't use deprecated GStaticMutex with newer glib versions

2011-12-05 15:34:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/Makefile.am:
	  examples: dist fft sub-directory

2011-11-28 10:05:50 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: unpremultiply text image
	  The GstVideoOverlayComposition only supports unpremultiplied ARGB
	  (for now anyway, support for pre-multiplied alpha is planned.)

2011-11-23 12:49:02 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	  textoverlay: Attach OverlayComposition to buffers when needed
	  Add video/x-surface support in the caps
	  We should then attach it whenever the sink supports it, but this
	  is working for the time being

2011-11-18 13:22:52 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	  textoverlay: Make the text_image data a buffer
	  This way we won't free data that would be attached to some buffer.

2011-11-18 11:04:47 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: Sync the caps with the new supported formats
	  Thanks to the use of the new video composition library, we gain support to
	  more colospaces and formats, let's state it.

2011-11-16 17:54:43 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	  textoverlay: Make use of the new video blending utility

2011-11-25 16:46:09 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/libs/video.c:
	  tests: add basic unit test for video overlay composition and rectangles

2011-11-12 14:59:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/video/Makefile.am:
	* gst-libs/gst/video/video-overlay-composition.c:
	* gst-libs/gst/video/video-overlay-composition.h:
	* win32/common/libgstvideo.def:
	  video: add video overlay composition API for subtitles
	  Basic API to attach overlay rectangles to buffers,
	  or blend them directly onto raw video buffers.
	  To be used primarily for things like subtitles or
	  logo overlays, not meant to replace videomixer.
	  Allows us to associate subtitle overlays with
	  non-raw video surface buffers, so that subtitles
	  are not lost and can instead be rendered later
	  when those surfaces are displayed or converted,
	  whilst re-using all the existing overlay plugins
	  and not having to teach them about our special
	  video surfaces. Could also have been made part
	  of the surface buffer abstraction of course, but
	  a secondary goal was to consolidate the blending
	  code for raw video into libgstvideo, and this
	  kind of API allows us to do both in a way that's
	  minimally invasive to existing elements, and at
	  the same time is fairly intuitive.
	  More features and extensions like the ability to
	  pass the source data or text/markup directly will
	  be added later.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665080
	  API: gst_video_buffer_get_overlay_composition()
	  API: gst_video_buffer_set_overlay_composition()
	  API: gst_video_overlay_composition_new()
	  API: gst_video_overlay_composition_add_rectangle()
	  API: gst_video_overlay_composition_n_rectangles()
	  API: gst_video_overlay_composition_get_rectangle()
	  API: gst_video_overlay_composition_make_writable()
	  API: gst_video_overlay_composition_copy()
	  API: gst_video_overlay_composition_ref()
	  API: gst_video_overlay_composition_unref()
	  API: gst_video_overlay_composition_blend()
	  API: gst_video_overlay_rectangle_new_argb()
	  API: gst_video_overlay_rectangle_get_pixels_argb()
	  API: gst_video_overlay_rectangle_get_pixels_unscaled_argb()
	  API: gst_video_overlay_rectangle_get_render_rectangle()
	  API: gst_video_overlay_rectangle_set_render_rectangle()
	  API: gst_video_overlay_rectangle_copy()
	  API: gst_video_overlay_rectangle_ref()
	  API: gst_video_overlay_rectangle_unref()

2011-11-23 00:31:18 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/video/Makefile.am:
	* gst-libs/gst/video/video-blend.h:
	  video: hide private video-blend.[ch] from gobject-introspection
	  And remove unused fields from helper structure.

2011-11-15 18:00:00 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/video/videoblendorc-dist.c:
	* gst-libs/gst/video/videoblendorc-dist.h:
	  video: add fallbacks for compilation without orc

2011-10-17 17:25:11 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst-libs/gst/video/.gitignore:
	* gst-libs/gst/video/Makefile.am:
	* gst-libs/gst/video/video-blend.c:
	* gst-libs/gst/video/video-blend.h:
	* gst-libs/gst/video/videoblendorc.orc:
	  video: add some internal helper functions for image blending
	  This could be improved if we decide we don't need it to
	  be this generic/flexible.

2011-12-05 09:38:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/interfaces/xoverlay.c:
	  xoverlay: Fix mistakes in the sample code
	  Fixes bug #665430.

2011-12-04 20:50:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsamixer.c:
	* ext/ogg/gstoggdemux.c:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gstplaysink.c:
	* gst/playback/gststreamsynchronizer.c:
	* gst/tcp/gstmultifdsink.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.

2011-12-04 20:38:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsamixer.c:
	* ext/alsa/gstalsamixer.h:
	  alsamixer: use GRectMutext instead of GStaticRecMutex with newer glib versions

2011-12-04 20:21:26 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsamixer.c:
	* ext/alsa/gstalsamixer.h:
	  alsamixer: embed static mutexes into the mixer structure
	  instead of allocating them dynamically

2011-12-04 17:02:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/encoding/encoding.c:
	* tests/examples/overlay/gtk-xoverlay.c:
	* tests/examples/overlay/qt-xoverlay.cpp:
	* tests/examples/seek/jsseek.c:
	* tests/examples/seek/scrubby.c:
	* tests/examples/seek/seek.c:
	* tests/icles/stress-playbin.c:
	* tests/icles/test-colorkey.c:
	* tests/icles/test-xoverlay.c:
	* tools/gst-discoverer.c:
	  tools, tests: g_thread_init() is deprecated in glib master
	  It's not needed any longer.

2011-12-04 16:43:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsadeviceprobe.c:
	* ext/alsa/gstalsamixer.c:
	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasrc.c:
	* ext/ogg/gstoggdemux.c:
	* ext/pango/gsttextoverlay.c:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/app/gstappsink.c:
	* gst-libs/gst/app/gstappsrc.c:
	* gst-libs/gst/audio/gstaudiosink.c:
	* gst-libs/gst/audio/gstaudiosrc.c:
	* gst-libs/gst/audio/gstringbuffer.c:
	* gst-libs/gst/glib-compat-private.h:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* gst-libs/gst/video/convertframe.c:
	* gst/encoding/gststreamcombiner.c:
	* gst/encoding/gststreamsplitter.c:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybasebin.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gstplaysinkconvertbin.c:
	* gst/playback/gststreamsynchronizer.c:
	* gst/playback/gstsubtitleoverlay.c:
	* gst/playback/gsturidecodebin.c:
	* gst/tcp/gstmultifdsink.c:
	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.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.
	  Replace g_thread_create() with g_thread_try_new().

2011-12-04 15:23:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/xmpwriter.c:
	  xmpwriter: update for thread API deprecations in glib master

2011-12-04 13:43:06 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/fft/Makefile.am:
	  fft-example: re-add Makefile.am

2011-12-02 23:35:50 +0100  Stefan Sauer <ensonic@users.sf.net>

	* configure.ac:
	  configure: trim trailing whitespace

2011-12-02 23:34:47 +0100  Stefan Sauer <ensonic@users.sf.net>

	* configure.ac:
	* tests/examples/Makefile.am:
	* tests/examples/fft/.gitignore:
	* tests/examples/fft/fftrange.c:
	  tests: add a test for fft result value-ranges
	  Add a small example that uses ffts of various types and parameters and check the
	  result value ranges.

2011-09-13 21:10:43 +0200  Piotr Fusik <fox@scene.pl>

	* docs/design/design-audiosinks.txt:
	* docs/design/design-decodebin.txt:
	* docs/design/design-encoding.txt:
	* docs/design/design-orc-integration.txt:
	* docs/design/draft-keyframe-force.txt:
	* docs/design/draft-va.txt:
	* ext/alsa/gstalsamixer.c:
	* ext/libvisual/visual.c:
	* ext/ogg/README:
	* ext/ogg/gstoggdemux.c:
	* ext/theora/gsttheoradec.c:
	* ext/theora/gsttheoradec.h:
	* ext/theora/gsttheoraparse.c:
	* ext/vorbis/gstvorbisdec.c:
	* gst-libs/gst/app/gstappsink.c:
	* gst-libs/gst/app/gstappsrc.c:
	* gst-libs/gst/app/gstappsrc.h:
	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	* gst-libs/gst/audio/gstringbuffer.c:
	* gst-libs/gst/audio/multichannel.h:
	* gst-libs/gst/fft/gstfftf32.c:
	* gst-libs/gst/fft/gstfftf64.c:
	* gst-libs/gst/fft/gstffts16.c:
	* gst-libs/gst/fft/gstffts32.c:
	* gst-libs/gst/interfaces/navigation.c:
	* gst-libs/gst/interfaces/xoverlay.c:
	* gst-libs/gst/netbuffer/gstnetbuffer.c:
	* gst-libs/gst/pbutils/descriptions.c:
	* gst-libs/gst/pbutils/encoding-profile.c:
	* gst-libs/gst/pbutils/encoding-target.h:
	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	* gst-libs/gst/rtp/gstrtpbuffer.c:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* gst-libs/gst/rtsp/gstrtsprange.c:
	* gst-libs/gst/tag/gstexiftag.c:
	* gst-libs/gst/tag/gstvorbistag.c:
	* gst-libs/gst/tag/gstxmptag.c:
	* gst-libs/gst/tag/id3v2.3.0.txt:
	* gst-libs/gst/tag/id3v2.4.0-frames.txt:
	* gst-libs/gst/tag/id3v2.4.0-structure.txt:
	* gst/adder/gstadder.c:
	* gst/audioconvert/audioconvert.c:
	* gst/audiorate/gstaudiorate.c:
	* gst/audioresample/gstaudioresample.c:
	* gst/audioresample/resample.c:
	* gst/encoding/gststreamsplitter.c:
	* gst/ffmpegcolorspace/avcodec.h:
	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	* gst/ffmpegcolorspace/imgconvert.c:
	* gst/ffmpegcolorspace/imgconvert_template.h:
	* gst/ffmpegcolorspace/mem.c:
	* gst/playback/README:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybasebin.c:
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gstplaysink.c:
	* gst/playback/gsturidecodebin.c:
	* gst/tcp/gstmultifdsink.c:
	* gst/tcp/gsttcp.c:
	* gst/typefind/gsttypefindfunctions.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* m4/freetype2.m4:
	* sys/v4l/v4lmjpegsrc_calls.c:
	* sys/v4l/videodev_mjpeg.h:
	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.c:
	* sys/xvimage/xvimagesink.h:
	* tests/check/elements/adder.c:
	* tests/check/elements/audioresample.c:
	* tests/check/elements/gnomevfssink.c:
	* tests/check/elements/textoverlay.c:
	* tests/examples/encoding/encoding.c:
	  various: typo fixes
	  Fix typos in code and docs. Fixes. #658984

2011-12-01 11:59:17 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/adder/gstadder.c:
	  adder: be more graceful in the clipfunction
	  Doing dynamic pipelines is hard in 0.10. As we don't have the sticky events in
	  0.10 and sending such events in special elements like adder and tee was outvoted
	  on last attempt, be graceful to the misbehaviour instead.

2011-12-01 01:22:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/audioresample.c:
	  tests: fix caps leak in audioresample tests

2011-12-01 01:07:26 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/pipelines/basetime.c:
	  tests: fix memory leak in basetime test

2011-11-30 23:58:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: tone down debug message about file URIs with spaces
	  Complain a bit less loudly about URIs that have not been
	  escaped properly.

2011-11-30 23:15:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	* ext/alsa/gstalsasrc.h:
	  Revert "alsasrc: Improve timestamp accuracy"
	  This reverts commit 0b774e0b7cf7a8ef1780fb6100228ca6e8ca8bcf.

2011-11-30 23:15:22 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	  Revert "alsasrc: Fix some compilation errors"
	  This reverts commit 2b84f5bd74ddb50f7832917ea8b4dd38d005631b.

2011-11-30 23:15:12 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	  Revert "alsa: Remove unused but set variable"
	  This reverts commit e9aed7f31c7e9e415f733e147140ce3ef2f57a61.

2011-11-30 23:15:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	* ext/alsa/gstalsasrc.h:
	  Revert "alsasrc: fail gracefully when ALSA does not give timestamps"
	  This reverts commit c7282a5718c7f31f84fb31b2c38fab0f9a38e2b0.

2011-11-30 23:14:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	  Revert "alsasrc: handle the case where the drivers don't supply timestamps"
	  This reverts commit 8154b69112cdc4830cd6002ec6c1f2917d30437b.

2011-11-28 10:55:39 +0100  Stefan Sauer <ensonic@google.com>

	* ext/alsa/gstalsasrc.c:
	  Revert "alsasrc: style fix"
	  This reverts commit f70ca6d4cbfd2b672dcc7215814bf6b39ce2c3f8.

2011-11-30 14:25:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Don't send undefined NEWSEGMENT events to the internal elements
	  This happens when the internal elements are added before any NEWSEGMENT
	  event arrived and in that case we shouldn't send a NEWSEGMENT event
	  to the internal elements at all. They will get the NEWSEGMENT event
	  from upstream later.

2011-11-29 14:15:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: Fix decoder-sink compatibility check for raw audio/video formats
	  If the sink supports raw audio/video, we first check
	  if the decoder could output any raw audio/video format
	  and assume it is compatible with the sink then. We don't
	  do a complete compatibility check here if converters
	  are plugged between the decoder and the sink because
	  the converters will convert between raw formats and
	  even if the decoder format is not supported by the decoder
	  a converter will convert it.
	  We assume here that the converters can convert between
	  any raw format.
	  Fixes bug #665120.

2011-11-29 09:11:21 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: fix compiler warning

2011-11-29 08:49:53 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	* win32/common/libgstvideo.def:
	  libgstvideo: minor fixes to key unit events
	  Make out args to gst_video_event_parse_{downstream|upstream}_force_key_unit
	  optional, update libgstvideo.def and fix docs a bit.
	  API: gst_video_event_new_upstream_force_key_unit
	  API: gst_video_event_new_downstream_force_key_unit
	  API: gst_video_event_is_force_key_unit
	  API: gst_video_event_parse_upstream_force_key_unit
	  API: gst_video_event_parse_downstream_force_key_unit
	  https://bugzilla.gnome.org/show_bug.cgi?id=607742

2011-06-05 01:49:38 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	  libgstvideo: Add force key unit events

2011-11-28 20:11:09 +0100  Philippe Normand <philn@igalia.com>

	* gst-libs/gst/fft/gstfft.h:
	* gst-libs/gst/fft/gstfftf32.h:
	* gst-libs/gst/fft/gstfftf64.h:
	* gst-libs/gst/fft/gstffts16.h:
	* gst-libs/gst/fft/gstffts32.h:
	  fft: Bracket public headers
	  This is especially needed if the gstfftw library is used from C++
	  code.
	  Fixes #665074

2011-11-28 20:10:18 +0100  Philippe Normand <phil@base-art.net>

	* gst/typefind/gsttypefindfunctions.c:
	  typefindfunctions: Fix compiler warning

2011-11-28 19:03:50 +0100  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: fix build error
	  fix build errors:
	  gsttypefindfunctions.c:248:25: error: 'low' may be used uninitialized in this function [-Werror=uninitialized]
	  gsttypefindfunctions.c:239:24: error: 'high' may be used uninitialized in this function [-Werror=uninitialized]
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>

2011-11-28 19:06:57 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Fix stupid mistake in last commit

2011-11-28 19:03:54 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Only return the converter caps if we actually have raw caps
	  Fixes bug #664818 (hopefully).

2011-11-28 17:59:32 +0100  Kipp Cannon <kcannon@cita.utoronto.ca>

	* gst/audioresample/gstaudioresample.c:
	  audioresample: Don't emit DISCONT buffers if no discontinuity happened
	  audioresample is derived from GstBaseTransform, and one of
	  GstBaseTransform's traits is that if the derived element does not
	  produce an output buffer from some input buffer then the first output
	  buffer after that gets flaged as a discontinuity, whether or not the
	  buffer actually is discontinuous from the output buffer that preceded
	  it. When downsampling, the audioresample element requires more than
	  one input sample for each output sample, and if the ratio of input to
	  output sample rates is high enough and the input buffers short enough
	  it can come to pass that the resampler does not receive enough samples
	  on its input to produce any output.  Currently the resampler returns
	  GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case,
	  causing the next buffer to be flagged as a discontinuity. If subsequent
	  elements in the pipeline reset themselves on disconts, this can cause
	  clicks and other undesireable behaviour.
	  Fixes bug #665004.

2011-09-30 20:00:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/typefind/Makefile.am:
	* gst/typefind/gsttypefindfunctions.c:
	  typefind: typefind UTF-16 and UTF-32
	  This avoids the MP3 typefinder from getting the highest score
	  every time it thinks there's something it might possibly be
	  able to parse.
	  https://bugzilla.gnome.org/show_bug.cgi?id=607619

2011-11-28 13:27:29 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoradec.c:
	* ext/theora/gsttheoradec.h:
	  Revert "theoradec: move the QoS logic to libgstvideo"
	  This reverts commit 149a4ce390a78e21309b210f7daba9db5d42afe6.
	  *grumble* I managed to merge something I did not mean to.

2011-11-28 13:26:53 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	* win32/common/libgstvideo.def:
	  Revert "libgstvideo: add a new API to handle QoS events and dropping logic"
	  This reverts commit eb03323fb683e06ed8e7f557037f13252f150c25.
	  *grumble* I managed to merge something I did not mean to.

2011-11-28 12:51:22 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasrc.c:
	* ext/gio/gstgiobasesink.c:
	* ext/gio/gstgiobasesrc.c:
	* ext/gnomevfs/gstgnomevfssink.c:
	* ext/gnomevfs/gstgnomevfssrc.c:
	* ext/libvisual/visual.c:
	* ext/ogg/gstoggaviparse.c:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggparse.c:
	* ext/ogg/gstogmparse.c:
	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextrender.c:
	* ext/theora/gsttheoradec.c:
	* ext/theora/gsttheoraenc.c:
	* ext/theora/gsttheoraparse.c:
	* ext/vorbis/gstvorbisdec.c:
	* ext/vorbis/gstvorbisenc.c:
	* ext/vorbis/gstvorbisparse.c:
	* gst-libs/gst/app/gstappsink.c:
	* gst-libs/gst/app/gstappsrc.c:
	* gst-libs/gst/cdda/gstcddabasesrc.c:
	* gst-libs/gst/tag/gsttagdemux.c:
	* gst/adder/gstadder.c:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audiorate/gstaudiorate.c:
	* gst/audioresample/gstaudioresample.c:
	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/encoding/gstencodebin.c:
	* gst/encoding/gstsmartencoder.c:
	* gst/encoding/gststreamcombiner.c:
	* gst/encoding/gststreamsplitter.c:
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	* gst/gdp/gstgdpdepay.c:
	* gst/gdp/gstgdppay.c:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaysink.c:
	* gst/playback/gststreamselector.c:
	* gst/playback/gststreamsynchronizer.c:
	* gst/playback/gstsubtitleoverlay.c:
	* gst/playback/gsturidecodebin.c:
	* gst/subparse/gstssaparse.c:
	* gst/subparse/gstsubparse.c:
	* gst/tcp/gstmultifdsink.c:
	* gst/tcp/gsttcpclientsink.c:
	* gst/tcp/gsttcpclientsrc.c:
	* gst/tcp/gsttcpserversrc.c:
	* gst/videorate/gstvideorate.c:
	* gst/videoscale/gstvideoscale.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* sys/v4l/gstv4lmjpegsink.c:
	* sys/v4l/gstv4lmjpegsrc.c:
	* sys/v4l/gstv4lsrc.c:
	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.c:
	* tests/check/elements/audiorate.c:
	* tests/check/elements/decodebin.c:
	* tests/check/elements/decodebin2.c:
	* tests/check/elements/playbin.c:
	* tests/check/elements/playbin2-compressed.c:
	* tests/check/elements/playbin2.c:
	* tests/check/elements/videoscale.c:
	  various: fix pad template leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=662664

2011-09-07 16:04:14 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoradec.c:
	* ext/theora/gsttheoradec.h:
	  theoradec: move the QoS logic to libgstvideo
	  https://bugzilla.gnome.org/show_bug.cgi?id=658241

2011-09-05 13:56:05 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	* win32/common/libgstvideo.def:
	  libgstvideo: add a new API to handle QoS events and dropping logic
	  https://bugzilla.gnome.org/show_bug.cgi?id=658241

2011-11-28 11:30:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstaudioencoder.h:
	  audioencoder: elaborate some documentation

2011-11-28 11:28:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	  audiodecoder: add some documentation

2011-11-21 14:26:54 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: really discard NULL decoded frame altogether
	  ... including any timestamp, rather than having that one influence base_ts.

2011-11-28 10:55:39 +0100  Stefan Sauer <ensonic@google.com>

	* ext/alsa/gstalsasrc.c:
	  alsasrc: style fix
	  Use timestamp==0 instead of mixing it with !timestamp style checks.

2011-11-28 09:12:37 +0100  Stefan Sauer <ensonic@users.sf.net>

	* ext/alsa/gstalsasrc.c:
	  alsasrc: handle the case where the drivers don't supply timestamps
	  If highres-timestamp is 0, try lowres and if that fails fallback to system clock
	  timestamps.

2011-11-01 15:21:54 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	  oggmux: set collectpads2 not to wait on sparse streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=663174

2011-11-25 15:35:39 +0100  Josep Torra <n770galaxy@gmail.com>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: make identiy silent

2011-11-25 13:11:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/vorbis/Makefile.am:
	* 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:
	  audio: remove unstable API guards from the audio decoder and encoder base classes

2011-11-25 12:58:22 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  docs: mention explicitly that playbin2 signals are emitted from a streaming thread

2011-11-25 11:11:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Set the multiqueue limits to the playing limits after overrun too
	  We don't expect any new pads anymore and prerolling is finished now.

2011-11-25 11:08:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Cache the upstream seekability for demuxer decode chains and use it for the non-preroll multiqueue limits
	  After preroll the multiqueue limits are still set to the preroll
	  limits if use-buffering is set to TRUE. In that case we only want
	  time limits on the multiqueue if upstream is seekable.

2011-11-08 13:55:58 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: fix prerolling for low bitrate streams from hlsdemux
	  Such streams were detected as seekable, as the query on the typefind
	  element was testing the m3u8 file listing the actual streams, and
	  not going through the demuxer(s).
	  We now check for seekability for each multiqueue following a demuxer,
	  so the query will flow through the elements which might prevent seeking.
	  https://bugzilla.gnome.org/show_bug.cgi?id=647769

2011-10-24 11:46:05 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: minor cleanup

2011-09-27 16:45:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/riff/riff-ids.h:
	  libgstriff: add a couple tags that need skipping
	  Found in a sample in the wild, appears to be ID3 tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=660249

2011-11-24 14:41:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: Rename ARG_ enums to PROP_
	  This is more consistent with other code and these are
	  properties anyway, not arguments

2011-11-24 14:29:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	* gst/videorate/gstvideorate.h:
	  videorate: Add property to force an output framerate
	  API: GstVideoRate:force-fps
	  Changing the framerate during playback is not possible
	  with a capsfilter downstream if upstream is not using
	  gst_pad_alloc_buffer(). In that case there's no way in
	  0.10 to signal to videorate that the preferred framerate
	  has changed.
	  This new property will force the output framerate to
	  a specific value and can be changed during playback.

2011-11-24 12:38:54 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Reconfigure if we switch from raw to incompatible raw caps
	  We might need to add converters and worked in passthrough mode before.

2011-11-24 12:37:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Override acceptcaps function for the two ghostpads
	  The ghostpad acceptcaps functions are not valid in this case because
	  we don't only accept the caps accepted by the target but could also
	  insert converters. Fixes bug #663892.

2011-11-24 11:34:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkaudioconvert.c:
	  playsinkaudioconvert: use-volume and use-converters are no construct-only properties anymore
	  Fixes bug #663893.

2011-10-22 20:29:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: skip the second bisection when possible
	  If we already saw the keyframes that we need to find,
	  we do not need to bisect to find them.
	  This will always be the case for streams with audio only,
	  where each frame acts as a keyframe, but will occasionally
	  also happen for streams with video.
	  https://bugzilla.gnome.org/show_bug.cgi?id=662475

2011-10-22 20:20:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggdemux.h:
	  oggdemux: improve push time seeking
	  Various tweaks to improve convergence, in particular for
	  the worst case, which is now cut in about half.
	  https://bugzilla.gnome.org/show_bug.cgi?id=662475

2011-10-21 19:38:19 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggdemux.h:
	  oggdemux: gather some more stats about bisection
	  https://bugzilla.gnome.org/show_bug.cgi?id=662475

2011-11-23 16:09:13 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vorbis/gstvorbisenc.c:
	  vorbisenc: do not accept 256 channels, 255 is the max vorbis supports

2011-11-22 13:29:10 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: extract opus comments if available

2011-11-22 13:15:33 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: recognize opus headers from data, not packet count
	  Opus streams outside of Ogg may not have headers, and oggstream
	  may be used by oggmux to mux an Opus stream which does not come
	  from Ogg - thus without headers.
	  Determining headerness by packet count would strip the first two
	  packets from such an Opus stream, leading to a very small amount
	  of audio being clipped at the beginning of the stream.

2011-11-22 13:01:35 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: add some more debug info when determining start time

2011-11-22 12:55:56 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: fix opus duration calculation

2011-11-22 12:00:58 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	  oggstream: early out on headers when determining packet duration

2011-11-21 17:03:21 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstoggstream.h:
	  oggstream: account for opus pre-skip in granpos/time mapping

2011-11-22 10:04:12 +0100  René Stadler <rene.stadler@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: avoid removing children from bin twice
	  GstBin base class removes children in dispose, so we need to do the same.

2011-11-19 16:06:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggstream.c:
	  ogg: add opus support

2011-11-16 19:00:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vorbis/gstvorbisenc.c:
	  vorbisenc: reset tag setter interface when appropriate

2011-11-16 19:00:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: invalidate format info when setup negotiation failed
	  ... which ensures nothing subsequently tries to slip past _chain
	  and into a possibly improperly setup subclass.

2011-11-15 13:29:31 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: accept dropped buffers before we know the format
	  This allows flacdec to not emit audio for headers, while allowing
	  the base audio decoder to keep its timestamps in sync.

2011-11-14 12:45:31 +0100  Robert Swain <robert.swain@gmail.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audio: Remove some unused variables

2011-08-30 18:27:09 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	  rtcpbuffer: Add feedback message types from RFC 5104
	  These are Codec Control messages (CCM)
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-10-19 16:30:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: improve reverse playback
	  ... by doing some more (reverse) timestamp interpolating and
	  refactoring downstream pushing.
	  Fixes #661983.

2011-11-13 13:18:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class
	  API: GST_AUDIO_INFO_IS_VALID

2011-11-12 15:51:52 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* tests/examples/seek/jsseek.c:
	* tests/examples/seek/seek.c:
	* tests/icles/test-colorkey.c:
	* tests/icles/test-xoverlay.c:
	  tests: require Gtk+ 3.0 for examples and Gtk-based test apps
	  The Gtk+ dependency is entirely optional, we're just not
	  supporting Gtk+ 2.x any longer.

2011-11-07 17:36:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/Makefile.am:
	  audio: fix order in LIBADD
	  Local libs must come first.

2011-11-11 13:32:23 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: fix visualisations again
	  Make caps writable before merging other caps into them.

2011-11-10 15:55:31 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: make unsigned properties unsigned, not signed

2011-11-09 00:36:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	* configure.ac:
	  configure: suppress warnings about unused variables if debugging system is disabled in core
	  https://bugzilla.gnome.org/show_bug.cgi?id=662952

2011-10-27 14:48:52 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: continue processing text when silent
	  This prevents playback wegding when text buffers are
	  left to pile up.
	  https://bugzilla.gnome.org/show_bug.cgi?id=662829

2011-11-08 00:16:56 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* win32/common/libgstaudio.def:
	  win32: update .def file for new audiosink API
	  API: gst_base_audio_sink_get_alignment_threshold()
	  API: gst_base_audio_sink_set_alignment_threshold()
	  API: gst_base_audio_sink_get_discont_wait()
	  API: gst_base_audio_sink_set_discont_wait()

2011-11-07 23:41:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/examples/seek/seek.c:
	  examples: sprinkle GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS in seek test utility
	  https://bugzilla.gnome.org/show_bug.cgi?id=630497

2011-11-07 23:05:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pango/gsttextoverlay.c:
	* gst-libs/gst/audio/gstaudioiec61937.c:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/video/video.c:
	  docs: fix up some Since: markers

2011-11-04 10:34:27 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: fix speed level failure test
	  It was testing the opposite of what it thought it was.
	  https://bugzilla.gnome.org/show_bug.cgi?id=663390

2011-11-04 10:57:40 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: make logically static const data just so
	  https://bugzilla.gnome.org/show_bug.cgi?id=663391

2011-11-04 10:58:15 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: use th_packet_iskeyframe instead of peeking at bits
	  https://bugzilla.gnome.org/show_bug.cgi?id=663391

2011-11-04 10:59:00 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: trivial comment typos fixes
	  https://bugzilla.gnome.org/show_bug.cgi?id=663391

2011-11-04 10:59:12 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: warn when trying to set an ignored obsolete property
	  https://bugzilla.gnome.org/show_bug.cgi?id=663391

2011-11-04 11:10:46 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: refuse to get to READY if the encoder was disabled
	  https://bugzilla.gnome.org/show_bug.cgi?id=663391

2011-10-18 17:58:49 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: survive skeleton finding length behind our backs in push mode
	  In push mode, we determine duration by doing a seek to the end of the
	  stream. However, a skeleton stream with an index will cause the duration
	  to be known already, and we end up never setting the push_time_duration
	  variable which we use to know duration has been determined.
	  https://bugzilla.gnome.org/show_bug.cgi?id=662049

2011-10-05 15:29:54 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/gst-plugins-base.supp:
	  valgrind: add ALSA leaks fixed by snd_config_update_free_global
	  If they go when calling snd_config_update_free_global, they're
	  not really bug leaks, but more like intentional ones we don't
	  want to get told about.
	  https://bugzilla.gnome.org/show_bug.cgi?id=615342

2011-05-02 13:05:28 +0300  Felipe Contreras <felipe.contreras@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	  baseaudiosink: make discont-wait configurable
	  Now we can configure how much time to wait before deciding that a
	  discont has happened.
	  Also, adds getter and setter to allow derived implementations to set
	  this value upon construction.
	  Suggestions and several improvements by Havard Graff.
	  Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>

2011-11-07 11:31:47 +0100  Felipe Contreras <felipe.contreras@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: delay the resyncing of timestamp vs ringbuffertime
	  A common problem for audio-playback is that the timestamps might not
	  be completely linear. This is specially common when doing streaming over
	  a network, where you can have jittery and/or bursty packettransmission,
	  which again will often be reflected on the buffertimestamps.
	  Now, the current implementation have a threshold that says how far the
	  buffertimestamp is allowed o drift from the ideal aligned time in the
	  ringbuffer. This was an instant reaction, and ment that if one buffer
	  arrived with a timestamp that would breach the drift-tolerance, a resync
	  would take place, and the result would be an audible gap for the
	  listener.
	  The annoying thing would be that in the case of a "timestamp-outlier",
	  you would first resync one way, say +100ms, and then, if the next
	  timestamp was "back on track", you would end up resyncing the other way
	  (-100ms) So in fact, when you had only one buffer with slightly off
	  timestamping, you would end up with *two* audible gaps. This is the
	  problem this patch addresses.
	  The way to "fix" this problem with the previous implementation, would
	  have been to increase the "drift-tolerance" to a value that was greater
	  than the largest timestamp-outlier one would normally expect.  The big
	  problem with this approach, however, is that it will allow normal
	  operations with a huge offset timestamp vs running-time, which is
	  detrimental to lip-sync. If the drift-tolerance is set to 200ms, it
	  basically means that lip-sync can easily end up being off by that much.
	  This patch will basically start a timer when the first breach of
	  drift-tolerance is detected. If any following timestamp for the next n
	  nanoseconds gets "back on track" within the threshold, it has basically
	  eliminated the effect of an outlier, and the timer is stopped.  If,
	  however, all timestamps within this time-limit are breaching the
	  threshold, we are probably facing a more permanent offset in the
	  timestamps, and a resync is allowed to happen.
	  So basically this patch offers something as rare as both higher
	  accuracy, it terms of allowing smaller drift-tolerances, as well as much
	  smoother, less glitchy playback!
	  Commit message and improvments by Havard Graff.
	  Fixes bug #640859.

2011-11-07 11:18:34 +0100  Felipe Contreras <felipe.contreras@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: rename some variables

2011-05-21 16:16:42 +0300  Felipe Contreras <felipe.contreras@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: use gst_util_uint64_scale_int when appropriate
	  It's probably safer this way.

2011-05-21 15:49:20 +0300  Felipe Contreras <felipe.contreras@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	  baseaudiosink: split drift-tolerance into alignment-threshold
	  So that drift-tolerance is used for clock slaving resync, and
	  alignment-threshold is for timestamp drift.

2011-05-21 16:02:36 +0300  Felipe Contreras <felipe.contreras@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: trivial comment fixes
	  Some found by Havard Graff.
	  Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>

2011-11-04 10:37:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: Use gst_caps_merge() instead of gst_caps_union()
	  This keeps the caps order and is more efficient.

2011-11-04 10:36:51 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Use gst_caps_merge() instead of gst_caps_union()
	  This keeps the caps order and is more efficient.

2011-11-03 21:35:38 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>

	* gst-libs/gst/tag/Makefile.am:
	  Add missing default include paths to androgenizer call
	  Fixes building tag/ with Android's NDK

2011-11-03 14:10:31 +0200  Mart Raudsepp <mart.raudsepp@collabora.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Post all source pads in stream-topology messages as "element-srcpad" values
	  This allows us to easily get ahold of all pads on a stream-topology message, including
	  pre-decoder ones, while "pad" only gives us access to the raw pads (as used by discoverer).

2011-10-20 13:04:52 +0300  Mart Raudsepp <mart.raudsepp@collabora.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Use existing "caps" quark for one of the structure sets

2011-11-03 10:07:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Don't add identity multiple times

2011-10-19 14:13:39 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsink: send flush start/stop event when we switch elements
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-19 14:13:30 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkaudioconvert.c:
	* gst/playback/gstplaysinkconvertbin.c:
	* gst/playback/gstplaysinkconvertbin.h:
	  playsink: re-add identity where appropriate
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-19 14:12:01 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkaudioconvert.c:
	  playsink: lock the new {set,get}_property functions
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-17 23:14:54 +0000  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Be more consistent with ghostpad targets
	  Set up targets on READY->PAUSED state change to passthrough by
	  default. This prevents the targets from being unset on the
	  first run, while the 'raw' variable would mean that some
	  target is set.

2011-10-17 22:41:49 +0000  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: No need to remove the identity
	  The identity element should be handled by the GstBin's cleanup,
	  removing it on the remove_elements function might remove it
	  too soon, as this function can be called directly from playsink

2011-10-17 22:41:11 +0000  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Adding some debug messages
	  Adds a couple debug messages and some g_assert to make debugging
	  easier

2011-10-17 22:02:03 +0000  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/playback/gstplaysinkvideoconvert.c:
	  playsink-videoconvert: Fix warning on build
	  Remove unused variable

2011-10-17 21:05:30 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	* gst/playback/gstplaysinkaudioconvert.c:
	* gst/playback/gstplaysinkaudioconvert.h:
	* gst/playback/gstplaysinkconvertbin.c:
	* gst/playback/gstplaysinkconvertbin.h:
	* gst/playback/gstplaysinkvideoconvert.c:
	* gst/playback/gstplaysinkvideoconvert.h:
	  playsink: handle after-the-fact changes in converters/volume booleans
	  The playsink was nastily poking a boolean in the structure.
	  Make those booleans properties, so we are told when they change,
	  and rebuild the conversion bin when they do.
	  Some cleanup to go with it too.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-17 18:43:06 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsink: handle NULL cached caps in getcaps
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-17 18:06:00 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsink: consider both passthrough and converter caps in getcaps
	  Since we can switch between both modes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-17 17:54:27 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	* gst/playback/gstplaysinkconvertbin.h:
	  playsink: cache inner converter bin caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-17 17:26:48 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsink: keep both raw and non raw pipelines at all times
	  and switch between them as needed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-17 17:29:50 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsink: only compare against the media type we expect
	  ie, audio/x-raw- for audio, video/x-raw- for video.
	  Add a trailing - to be more specific. I doubt there's anything
	  like audio/x-rawhide or something, but you never know.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-17 16:55:30 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/Makefile.am:
	* gst/playback/gstplaysinkaudioconvert.c:
	* gst/playback/gstplaysinkaudioconvert.h:
	* gst/playback/gstplaysinkconvertbin.c:
	* gst/playback/gstplaysinkconvertbin.h:
	* gst/playback/gstplaysinkvideoconvert.c:
	* gst/playback/gstplaysinkvideoconvert.h:
	  playsink: refactor the converter bins since they are almost identical
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-17 13:00:05 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkaudioconvert.c:
	* gst/playback/gstplaysinkaudioconvert.h:
	* gst/playback/gstplaysinkvideoconvert.c:
	* gst/playback/gstplaysinkvideoconvert.h:
	  playsink: fix passthrough mode (hopefully)
	  The code was doing counterintuitive rewiring of pads when the
	  bin did not contain any elements. We now add an identity element
	  in that case, which makes it simpler, and should fix the AC3
	  passthrough mode when using pulseaudio (but I don't see the bug
	  here so can't test).
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-07 11:16:44 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkaudioconvert.c:
	* gst/playback/gstplaysinkvideoconvert.c:
	  playsink: handle NULL ghost pad target
	  For the src pad anyway.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-11-03 09:56:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaysinkaudioconvert.c:
	  Revert "playsinkaudioconvert: Fix warning when there is no target pad yet"
	  This reverts commit f35c51c14915729f0fdf2b348f351ea7e81027cc.
	  Better patch coming soon.

2011-10-28 10:07:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	  oggmux: Remove obsolete #include

2011-11-02 23:33:18 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/design/draft-subtitle-overlays.txt:
	  docs: add draft for subtitle overlays to design docs
	  Main purpose is to provide a generic way to make subtitles work on
	  top of non-raw video (vaapi, vdpau, etc.).

2011-11-02 15:31:11 -0400  Colin Walters <walters@verbum.org>

	* common:
	* configure.ac:
	  configure: Allow setting GLIB_EXTRA_CFLAGS
	  Similar to gstreamer commit bb2020b1e794210cf7d44c6626122f611016a620

2011-10-30 20:00:47 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: don't use soon-to-be-deprecated gst_filter_run()

2011-10-28 13:58:47 +0200  Mersad Jelacic <mersad@axis.com>

	* gst-libs/gst/audio/gstaudiosink.c:
	  audiosink: avoid deadlocking audioringbuffer thread
	  ... when it goes into wait for ringbuffer starting just after such
	  having been signalled.
	  Fixes #661738.

2011-04-26 22:20:29 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: extract SOF marker in jpeg typefinder
	  The SOF types are defined by http://www.w3.org/Graphics/JPEG/itu-t81.pdf
	  This is needed to make sure that we plug a jpeg decoder that
	  can handle the type of JPEG we have (e.g. lossless JPEG)
	  https://bugzilla.gnome.org/show_bug.cgi?id=556648

2009-08-10 01:48:29 +0000  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggmux.h:
	  oggmux: port to gstcollectpads2

2011-10-27 23:39:31 +1100  Jan Schmidt <thaytan@noraisin.net>

	* tests/examples/Makefile.am:
	  build: Fix build for moved volume subdir

2011-10-27 09:51:46 +0200  Stefan Sauer <ensonic@users.sf.net>

	* Makefile.am:
	* configure.ac:
	* tests/examples/Makefile.am:
	* tests/examples/audio/.gitignore:
	* tests/examples/audio/Makefile.am:
	* tests/examples/audio/volume.c:
	* tests/examples/volume/.gitignore:
	* tests/examples/volume/Makefile.am:
	* tests/examples/volume/volume.c:
	  volume: move volume example to audio

2011-10-27 09:42:36 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/audio/Makefile.am:
	  audio examples. fix the makefile

2011-10-27 09:33:55 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/volume/volume.c:
	  volume: make global vars static

2011-10-27 09:33:01 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/audio/.gitignore:
	* tests/examples/audio/Makefile.am:
	* tests/examples/audio/audiomix.c:
	  audiomix: add a simple audiomix example

2011-10-25 20:04:06 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/playback/gstplaysinkaudioconvert.c:
	  playsinkaudioconvert: Fix warning when there is no target pad yet

2011-10-13 11:34:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Link elements before testing if they can reach the READY state
	  This is made possible by filtering errors. This is required to let
	  harware accelerated element query the video context. The video context
	  is used to determine if the HW is capable, and thus if the element is
	  supported or not.
	  Fixes bug #662330.

2011-10-21 21:57:17 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/playback/gstplaybasebin.c:
	  playbasebin: remove avoidable call to gst_object_set_name

2011-10-21 21:41:03 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: remove avoidable call to gst_object_set_name

2011-10-21 21:39:01 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/channelmixtest.c:
	  audioconvert: bury dead test program

2011-10-20 10:13:46 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>

	* Android.mk:
	  Disable ext/vorbis for the android ndk build
	  It currently makes the build fail. Idea is to enable
	  it back again once its building problems get sorted
	  out.

2011-10-19 19:44:06 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: fix leaks of pad templates and internal proxy pads

2011-10-19 19:37:07 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: fix leak of element reference through pad block
	  If the pad block never happens because there is no data flow at all, the
	  callback is never fired and the reference is never released. This causes a
	  reference cycle between the pad and element, so valgrind is not very vocal
	  about it (memory is still reachable).

2011-10-18 21:42:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: having gather queue contents implies some draining is in order
	  ... which ensures e.g. processing and sending last fragment of reverse playback
	  downstream at EOS.

2011-10-19 15:28:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/vorbis/gstvorbisdec.c:
	  vorbisdec: do not try to read past the buffer array
	  https://bugzilla.gnome.org/show_bug.cgi?id=662108

2011-10-18 21:40:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vorbis/gstvorbisdec.c:
	  vorbisdec: only finish header packet frame if received in-stream
	  ... rather than scaring audiodecoder with a frame extracted from caps.
	  Fixes #662108 (partially).

2011-10-19 10:41:31 +0200  Stefan Sauer <ensonic@users.sf.net>

	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.c:
	  x(v)imagesink: make it more clean that "synchronous" props are not for avsync

2011-10-19 00:32:13 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: fix unused variable compiler warning if debugging in core is disabled
	  https://bugzilla.gnome.org/show_bug.cgi?id=660150

2011-10-18 13:00:29 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: fix event unref in (rare) error case

2011-10-07 17:41:32 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: fire drained signal where appropriate
	  This will allow playbin2 to send its about-to-finish signal.
	  Taken out (apparently by mistake) by the EOS rewrite in july.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661202

2011-10-16 11:32:41 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: do not retry seeking indefinitely
	  https://bugzilla.gnome.org/show_bug.cgi?id=661897

2011-10-10 13:11:59 +0200  Brian Cameron <brian.cameron@oracle.com>

	* gst/videotestsrc/Makefile.am:
	  videotestsrc: fix LDADD missing GST_LIBS

2011-10-09 21:19:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vorbis/gstvorbisenc.c:
	* ext/vorbis/gstvorbisenc.h:
	  vorbisenc: only push header buffers following initial events

2011-10-09 16:48:18 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audioencoder: fix compile warning

2011-10-08 20:17:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/pipelines/vorbisenc.c:
	  tests: vorbisenc: adjust discontinuity checking to audioencoder behaviour
	  ... which still detects gaps and marks DISCONT, depending on configuration,
	  but may come up with somewhat different timestamps when crossing the gap.

2011-10-08 20:16:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/pipelines/vorbisdec.c:
	  tests: vorbisdec: properly configure audiodecoder when requiring perfect ts

2011-10-08 20:14:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/vorbisdec.c:
	  tests: vorbisdec: remove empty header buffer check
	  ... as empty buffers are discarded, and header buffers are now
	  also optionally retrieved from caps anyway.

2011-10-08 20:13:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: only resync to upstream upon discont in perfect ts mode
	  ... as documented, where discont is marked here if tolerance has been
	  exceeded.

2011-10-08 20:11:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: fix timestamp tolerance handling

2011-10-08 20:09:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: handle empty input by discarding

2011-10-07 14:52:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vorbis/Makefile.am:
	* ext/vorbis/gstvorbisdec.c:
	* ext/vorbis/gstvorbisdec.h:
	  vorbisdec: port to audiodecoder

2011-10-07 14:33:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: make upstream queries MT-safe

2011-10-07 14:32:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: make upstream queries and events MT-safe

2011-10-05 15:43:35 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/vorbis/Makefile.am:
	* ext/vorbis/gstvorbisenc.c:
	* ext/vorbis/gstvorbisenc.h:
	  vorbisenc: port to audioencoder

2011-10-06 18:21:29 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/elements/audiotestsrc.c:
	  tests: actually test what we said we would
	  All tests were testing the default sine wave
	  https://bugzilla.gnome.org/show_bug.cgi?id=661106

2011-10-06 18:20:32 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	  audiotestsrc: add missing break
	  And make violet noise usable
	  https://bugzilla.gnome.org/show_bug.cgi?id=661105

2011-10-06 15:38:49 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaysinkaudioconvert.c:
	* gst/playback/gstplaysinkvideoconvert.c:
	  playsink: fix caps negotiation through the new convenience bins
	  The bins' getcaps was bypassing the inner elements, and thus
	  failing to account for the caps transformations they allow,
	  which caused YUV video pipelines to fail with ximagesink, which
	  does not support YUV, even though the convenience bin includes
	  a colorspace converter for just this purpose.
	  https://bugzilla.gnome.org/show_bug.cgi?id=660816

2011-10-06 11:53:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: fix mismatch between video/ and video/x-dvd-subpicture
	  The new code was checking for a prefix, and would find video/
	  first. Check in two passes, first checking for a perfect match,
	  and falling back to a prefix check if nothing was found.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657261

2011-10-04 21:17:37 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/encoding/gstencodebin.c:
	  encodebin: Re-enable parsers
	  Re-enable parsers in encodebin to allow more passthrough scenarios
	  to work. Specially the ones that require changing 'stream formats'.
	  i.e. h264 in mkv to mpegts.

2011-10-05 12:45:19 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: Add audio- and text-sink props

2011-10-04 23:09:42 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	  auditestsrc: indent fix

2011-10-04 16:22:55 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/playback/gstplaysink.c:
	  playsink: Add video-sink property
	  The video-sink property allows manual specification via g_object_set ()
	  of the video sink element to be used.

2011-10-03 15:20:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: Minor cleanup of decoder-sink compatibility checking code

2011-09-30 12:29:34 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/playback/gstplaybin2.c:
	  playbin2: Make sure that the decoders we plug are compatible with the fixed sink
	  The fact that a decoder is not compatible with the fixed sink
	  is currently happenning in the case where we have hardware accelerated
	  video decoders on the system (especially vaapi elements that are actually plugged),
	  and the user is providing a sink that doesn't support the surface.
	  A simple example that shows how it used to crash on a system where gstreamer-vaapi
	  is installed:
	  gst-launch playbin2 video-sink=xvimagesink uri=/codec/supported/by/vaapi
	  What we are now doing in this case, is avoid using the accelerated
	  decoder and plug a "normal" decoder instead (if avalaible).
	  This commit doesn't handle the case where we have hardware accelerated
	  demuxing.

2011-02-18 11:48:37 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/encoding-profile.c:
	* gst-libs/gst/pbutils/encoding-profile.h:
	* win32/common/libgstpbutils.def:
	  encoding-profile: add a function to create a profile from a discoverer info
	  Only A/V streams are added at the moment, there does not seem to be
	  a similar way to add other streams (eg, subtitles).
	  https://bugzilla.gnome.org/show_bug.cgi?id=642878

2011-09-27 00:26:29 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	* ext/alsa/gstalsasrc.h:
	  alsasrc: fail gracefully when ALSA does not give timestamps
	  https://bugzilla.gnome.org/show_bug.cgi?id=660170

2011-10-03 10:55:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Use a TIME limit for pre-rolling in live streams and not in non-live streams
	  Fixes bug #647769 for real.

2011-10-01 01:05:00 +0100  Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: add YV12 support
	  Basically the same as I420, just with chroma planes swapped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=660604

2011-09-30 09:44:12 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/encoding/gstencodebin.c:
	  encodebin: Fix typo on formatter adding condition
	  The condition is if the muxer doesn't have tag setter *and* isn't
	  a formatter itself. Any of those two conditions makes the muxer
	  good enough to not need a formatter.

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

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

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

2011-09-19 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 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 16:46:02 +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 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 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 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 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 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 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

2011-08-27 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 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 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

2011-08-24 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 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

2011-08-23 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

2011-08-22 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 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.

2011-08-18 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

2011-07-27 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

2011-08-15 01:22:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/gstid3tag.c:
	* tests/check/libs/tag.c:
	  tag: id3: avoid some more relocations in genre table

2011-08-12 12:07:32 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/audioresample.c:
	  audioresample: add FFT based checks
	  Send a few simple tones through audioresample and check
	  that the main frequency spot is the same for the input and
	  the resampled output.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656392

2011-08-15 23:41:24 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: add OSX specific hack to detect when a connection is refused
	  Unlike linux, OSX wakes up select with POLLOUT (instead of POLLERR) when
	  connect() is done async and the connection is refused. Therefore always check
	  for the socket error state using getsockopt (..., SO_ERROR, ...) after a
	  connection attempt.

2011-08-15 00:17:14 +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: add new license API to docs

2011-08-15 00:03:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: try pkg-config first when looking for zlib

2011-08-14 20:44:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/id3v2.3.0.txt:
	* gst-libs/gst/tag/id3v2.4.0-frames.txt:
	* gst-libs/gst/tag/id3v2.4.0-structure.txt:
	  tag: id3v2: add specs to git for reference

2011-08-14 13:32:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/id3v2.c:
	  tag: id3v2: avoid some relocations, make table static

2011-08-14 01:47:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/id3v2.c:
	* gst-libs/gst/tag/id3v2.h:
	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: add debug category for ID3 tag parsing

2011-07-18 18:09:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/tag/id3v2.c:
	* gst-libs/gst/tag/id3v2.h:
	* gst-libs/gst/tag/id3v2frames.c:
	* gst-libs/gst/tag/tag.h:
	* gst-libs/gst/tag/tags.c:
	* win32/common/libgsttag.def:
	  tag: id3v2: add id3v2 tag parsing helpers
	  https://bugzilla.gnome.org/show_bug.cgi?id=654388

2011-02-22 15:19:00 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst-libs/gst/tag/id3v2.c:
	  tag: id3v2: return ID3TAGS_BROKEN_TAG for unsupported versions
	  This prevents us for trying to work with a NULL taglist.

2011-01-02 19:23:51 +0000  Erich Schubert <erich@debian.org>

	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: 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

2010-09-24 15:19:15 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst-libs/gst/tag/id3v2.c:
	  tag: id3v2: Sanitize id3 frame names
	  This is similar to what is done in qtdemux. Avoids providing invalid
	  structure/tags names

2010-03-30 01:50:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: fix parsing of unsynced frames with data length indicator
	  Fixes bug #614158.

2010-03-20 00:54:14 +0100  Benjamin Otte <otte@redhat.com>

	* gst-libs/gst/tag/id3v2.c:
	  Add -Wwrite-strings to the configure flags
	  ... and fix all warnings

2009-12-13 13:19:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: prefer two letter ISO 639-1 code for extended comment

2009-10-09 15:59:25 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst-libs/gst/tag/id3v2.c:
	  tag: id3v2: 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-libs/gst/tag/id3v2.c:
	  tag: id3v2: cast pointer math results to glong

2009-10-09 13:38:17 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst-libs/gst/tag/id3v2.c:
	  tag: id3v2: don't cast, but use the right format specified instead
	  This correct some of the previous macos fixes.

2009-10-09 11:42:36 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst-libs/gst/tag/id3v2.c:
	  tag: id3v2: fix printf warnings on macosx

2009-10-07 14:03:20 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: fprintf, sprintf, sscanf need stdio.h

2009-09-22 15:03:20 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: Fix compile warnings with gcc 4.0.1.

2009-08-09 12:52:17 +0200  LoneStar <lone@auvtech.com>

	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: Try GST_*_TAG_ENCODING and locale encoding if tags are not UTF8
	  Fixes bug #499242.

2009-08-07 16:42:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: 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-libs/gst/tag/id3v2.c:
	  tag: id3v2: fix typo in debug message

2009-08-07 16:02:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/id3v2.c:
	* gst-libs/gst/tag/id3v2.h:
	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: 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.

2009-04-24 01:51:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/id3v2.c:
	  tag: id3v2: 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-libs/gst/tag/id3v2.c:
	  tag: id3v2: 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.

2008-06-04 10:42:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: 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-libs/gst/tag/id3v2frames.c: (parse_picture_frame):
	  Use new utility functions in libgsttag to process coverart (#512333).

2008-01-11 21:08:59 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: 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-libs/gst/tag/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.

2008-01-09 15:20:19 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  tag: id3v2: 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-libs/gst/tag/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.

2007-12-14 10:17:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: Parse WOAF frames and put the result into GST_TAG_CONTACT, which is where it would end up...
	  Original commit message from CVS:
	  * tag: id3v2: (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.

2007-11-14 21:39:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: 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-libs/gst/tag/id3v2.c:
	  * gst-libs/gst/tag/id3v2.h:
	  * gst-libs/gst/tag/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.

2007-10-11 17:55:29 +0000  Jason Kivlighn <jkivlighn@gmail.com>

	  tag: id3v2: 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-libs/gst/tag/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.

2007-10-06 16:13:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: 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-libs/gst/tag/gstid3demux.c:
	  * gst-libs/gst/tag/gstid3demux.h:
	  * gst-libs/gst/tag/id3v2.c:
	  * gst-libs/gst/tag/id3v2.h:
	  * gst-libs/gst/tag/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).

2007-03-12 13:28:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: 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-libs/gst/tag/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'.

2007-03-06 18:16:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise the four-digit number will be interp...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2.c: (id3demux_id3v2_frames_to_tag_list):
	  * gst-libs/gst/tag/id3v2.h:
	  * gst-libs/gst/tag/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.

2006-11-19 13:41:53 +0000  René Stadler <mail@renestadler.de>

	  tag: id3v2: Make sure that g_free always gets called on the same pointer that was returned by g_mallo...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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>

2006-11-01 13:59:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: We require a -base more recent than 0.10.9, so it's safe to use
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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.

2006-10-05 16:37:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: 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-libs/gst/tag/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.

2006-08-22 13:53:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: If strings in text fields are marked ISO8859-1, but contain valid UTF-8 already, then han...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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)

2006-08-16 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-libs/gst/tag/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).

2006-07-25 16:47:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as well, and add the version to...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2.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).

2006-07-23 11:33:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: 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-libs/gst/tag/gstid3demux.c: (plugin_init):
	  * gst-libs/gst/tag/id3v2.c:
	  (id3demux_add_id3v2_frame_blob_to_taglist):
	  * gst-libs/gst/tag/id3v2.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>

	  tag: id3v2: 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-libs/gst/tag/id3v2.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>

	  tag: id3v2: Don't use \n in debug lines
	  Original commit message from CVS:
	  * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	  (gst_avi_demux_process_next_entry):
	  Fix some leaks.
	  * gst-libs/gst/tag/id3v2.c: (id3demux_id3v2_frames_to_tag_list):
	  Don't use \n in debug lines.

2006-06-22 12:17:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: Set image type from APIC frame as "image-type" field of GST_TAG_IMAGE buffer caps (#344605).
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2frames.c: (parse_picture_frame):
	  Set image type from APIC frame as "image-type" field
	  of GST_TAG_IMAGE buffer caps (#344605).

2006-06-11 19:31:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: Extract images from ID3v2 tags (APIC frames). Fixes #339704.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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).

2006-05-28 10:05:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: A track/volume number or count of 0 does not make sense, just ignore it along with negati...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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).

2006-05-19 14:05:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Don't output any tag when we encounter a negative track number - the tag type is uint, so...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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)

2006-05-16 14:07:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Rework string parsing to always walk over BOM markers in UTF16 strings, using the endianness indicated by the innermost one ...
	  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-libs/gst/tag/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

2006-05-12 08:21:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: Some more debug info. No need to check whether the string returned by g_convert() is real...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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.

2006-05-10 13:51:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Fix parsing of numeric genre strings some more, by ensuring that we only try and parse st...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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

2006-04-28 11:37:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: Recognise and skip any byte order marker (BOM) in
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2frames.c: (has_utf16_bom),
	  (parse_split_strings):
	  Recognise and skip any byte order marker (BOM) in
	  UTF-16 strings.

2006-04-17 10:01:51 +0000  Alex Lancaster <alexlan@fedoraproject.org>

	  tag: id3v2: Recognise TCO (Genre) tags in ID3v2.2
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2.c:
	  Recognise TCO (Genre) tags in ID3v2.2. Patch by Alex Lancaster
	  (Fixes #338713)

2006-03-30 23:37:16 +0000  Sébastien Moutte <sebastien@moutte.net>

	  tag: id3v2: 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

2006-03-22 13:00:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: 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-libs/gst/tag/gstid3demux.c: (gst_id3demux_chain):
	  Don't attempt typefinding on too-short buffers that have been
	  completely trimmed away.
	  * gst-libs/gst/tag/id3v2.c: (id3demux_read_id3v2_tag):
	  Improve the debug output

2006-03-16 16:06:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tag: id3v2: We only care about gain and peak data for the master volume.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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>

	  tag: id3v2: Read replay gain tags
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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).

2006-03-14 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-libs/gst/tag/gstid3demux.c: (plugin_init):
	  * gst-libs/gst/tag/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.

2006-02-18 20:48:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Handle 0 data size in otherwise valid frames.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2.c: (id3demux_id3v2_frames_to_tag_list):
	  * gst-libs/gst/tag/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

2006-02-16 10:58:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: 3 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-libs/gst/tag/id3v2.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)

2006-02-13 12:00:51 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Add more validation to ensure that a char encoding conversion produced a valid UTF-8 string.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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.

2006-02-04 13:30:12 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Adjust for data length indicators when parsing (Fixes #329810)
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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 13:06:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Never output a tag with a null contents string.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/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.

2006-01-30 23:13:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Someone should kick my butt. Remove ID3v1 tags from the end of the file.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3demux.c: (gst_id3demux_chain),
	  (gst_id3demux_read_id3v1), (gst_id3demux_sink_activate),
	  (gst_id3demux_send_tag_event):
	  * gst-libs/gst/tag/id3v2.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.

2006-01-25 18:23:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Never trust ANY information encoded in a media file, especially when it's giving you size...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame):
	  Never trust ANY information encoded in a media file, especially
	  when it's giving you sizes. (Fixes #328452)

2006-01-23 14:32:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Remove errant break statement, and fix compilation with older GCC.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2frames.c: (id3v2_tag_to_taglist):
	  Remove errant break statement, and fix compilation with
	  older GCC.

2006-01-23 09:22:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: Rewrite parsing of text tags to handle multiple NULL terminated strings. Parse numeric genre strings a...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2.c: (id3demux_read_id3v2_tag):
	  * gst-libs/gst/tag/id3v2.h:
	  * gst-libs/gst/tag/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.

2006-01-15 20:21:48 +0000  Sergey Scobich <sergey.scobich@gmail.com>

	  tag: id3v2: Fix compilation of id3demux when zlib is not present.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2frames.c: (id3demux_id3v2_parse_frame):
	  Fix compilation of id3demux when zlib is not present.
	  (Fixes #326602; patch by: Sergey Scobich)

2006-01-06 11:46:53 +0000  Edward Hervey <bilboed@bilboed.com>

	  tag: id3v2: Add gst_element_no_more_pads() for proper decodebin behaviour.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3demux.c: (gst_id3demux_add_srcpad):
	  Add gst_element_no_more_pads() for proper decodebin behaviour.
	  * gst-libs/gst/tag/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.

2005-12-28 18:55:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: If a broken tag has 0 bytes payload, at least still skip the 10 byte header
	  Original commit message from CVS:
	  * gst-libs/gst/tag/id3v2.c: (id3demux_read_id3v2_tag):
	  If a broken tag has 0 bytes payload, at least still skip
	  the 10 byte header

2005-12-18 15:14:44 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tag: id3v2: all new LGPL id3 demuxer, can use zlib for compressed frames
	  Original commit message from CVS:
	  * configure.ac:
	  Check for optional dependency on zlib for id3demux
	  * gst-libs/gst/tag/Makefile.am:
	  * gst-libs/gst/tag/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-libs/gst/tag/gstid3demux.h:
	  * gst-libs/gst/tag/id3v2.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-libs/gst/tag/id3v2.h:
	  * gst-libs/gst/tag/id3v2.4.0-frames.txt:
	  * gst-libs/gst/tag/id3v2.4.0-structure.txt:
	  * gst-libs/gst/tag/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

2011-08-11 18:50:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioresample/gstaudioresample.c:
	  audioresample: fix quality setting being ignored by the resampler state
	  https://bugzilla.gnome.org/show_bug.cgi?id=636562

2011-08-11 15:54:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* configure.ac:
	* gst/audioresample/resample.c:
	* gst/audioresample/resample_sse.h:
	* gst/audioresample/speex_resampler_double.c:
	* gst/audioresample/speex_resampler_float.c:
	  audioresample: use SSE/SSE2 when possible
	  Compile in the code on i386 and x86_64, and use ORC to determine
	  when the runtime platform can run the code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=636562

2011-08-11 19:23:42 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioresample/resample_sse.h:
	  audioresample: fix SSE2 building with double precision
	  The full double implementation was missing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=636562

2011-08-11 12:12:07 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst-libs/gst/tag/gstexiftag.c:
	  tag: exif: Check for utf8 before trying to convert
	  If the string is already on utf8, there is no need to
	  try to convert it, because it is useless and it might garble
	  the string.

2011-08-10 13:16:13 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/libs/tag.c:
	  tests: tag: exif: Add tests for 'non-trivial' chars
	  Adds two new cases to check that characters are properly
	  converted to ascii when writen to exif and parsed correctly
	  back to utf8 when read.

2011-08-09 16:02:28 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst-libs/gst/tag/gstexiftag.c:
	  tag: exif: Exif strings should be ascii
	  Use g_convert to turn all strings into extended ascii before writing
	  to the exif buffer and converting back from ascii to utf8 when
	  reading them.

2011-08-10 15:57:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* win32/common/libgsttag.def:
	  win32: update libgsttag.def for new API

2011-08-10 15:21:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/Makefile.am:
	  tag: don't build helper programs that generate/update data by default
	  No point building these by default. Also, these generated files
	  should go into the srcdir, not the builddir in this case, since
	  they're version controlled.

2011-08-10 15:20:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/mklicensestables.c:
	  tag: fix stray printf in mklicensestables
	  Don't dump debug output to stdout.

2011-08-10 15:06:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/licenses.c:
	  tag: fix compilation of new licenses code with GLib versions < 2.28
	  Add local g_variant_lookup_value() fallback for now when compiling
	  against older GLib versions.

2011-08-10 14:57:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/tag/licenses.c:
	* gst-libs/gst/tag/tag.h:
	  tag: add GType for GstTagLicenseFlags
	  API: gst_tag_license_flags_get_type()

2011-08-10 10:49:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/subparse/gstsubparse.c:
	  subparse: fix runtime warnings when doing position query
	  Add missing 'break'.

2011-07-15 13:19:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/libs/tag.c:
	* tests/files/Makefile.am:
	* tests/files/license-uris:
	  tag: add unit test for new license API
	  https://bugzilla.gnome.org/show_bug.cgi?id=646868

2011-07-15 13:14:16 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* .gitignore:
	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/tag/mklicensestables.c:
	  tag: add mklicensestables utility
	  Add (uninstalled) tool to create licenses-table.dat from liblicense's
	  RDF files. It's not very pretty and makes loats of assumptions about
	  the input, but should work. If things change, we can fix it then.
	  https://bugzilla.gnome.org/show_bug.cgi?id=646868

2011-07-15 13:07:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/tag/license-translations.dict:
	* gst-libs/gst/tag/licenses-tables.dat:
	* gst-libs/gst/tag/licenses.c:
	* gst-libs/gst/tag/tag.h:
	  tag: add convenience API to handle creative commons licenses
	  Based on liblicense's RDF files.
	  API: GstTagLicenseFlags
	  API: gst_tag_get_licenses()
	  API: gst_tag_get_license_flags()
	  API: gst_tag_get_license_nick()
	  API: gst_tag_get_license_title()
	  API: gst_tag_get_license_version()
	  API: gst_tag_get_license_description()
	  API: gst_tag_get_license_jurisdiction()
	  https://bugzilla.gnome.org/show_bug.cgi?id=646868

2011-08-08 10:00:40 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: bump probability if all frames we found are similar
	  Similar meaning same layer, same bitrate, and same number of channels
	  This fixes misdetection of (some MP3 files that have zero padding
	  between the ID3 tag and the MP3 stream) as H.264 video.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656018

2011-08-05 16:53:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/tag/gstvorbistag.c:
	  gstvorbistag: map ENCODER Vorbis comment to application-name
	  What GStreamer calls encoder ("encoder used to encode this stream") is
	  stored in the vendor string in Vorbis/Theora/Kate and possibly others.
	  The Vorbis comment packet used in those streams uses ENCODER as the name
	  of the encoding program, which GStreamer calls application-name.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656034

2011-08-05 11:32:09 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/volume/gstvolume.c:
	  volume: fix sample depth typo
	  https://bugzilla.gnome.org/show_bug.cgi?id=656022

2011-08-05 13:05:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/volume/gstvolumeorc-dist.c:
	  volume: Update disted ORC files

2011-08-03 14:14:55 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/encoding/gstencodebin.c:
	  encodebin: Set queues to silent=true
	  As encodebin doesn't connect to the queue signals, it can set
	  queues to silent mode to make queue not emit them.
	  Check https://bugzilla.gnome.org/show_bug.cgi?id=621299 for
	  more info on queue's silent property.

2011-08-03 13:40:19 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/encoding/gstencodebin.c:
	  encodebin: Fix typo on installing properties
	  queue buffers and bytes properties have ids swapped, fix it.

2011-08-03 10:18:29 +0200  Jonathan Liu <net147@gmail.com>

	* ext/ogg/gstoggstream.c:
	  oggstream: Fix crashes with 0-byte vorbis packets
	  Fixes bug #655574.

2011-07-28 14:43:53 +0200  Jens Georg <jensg@openismus.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  pbutils: Add SP levels 4a, 5 and 6
	  https://bugzilla.gnome.org/show_bug.cgi?id=655503

2011-07-26 16:10:17 +0200  Philip Jägenstedt <philipj@opera.com>

	* ext/theora/gsttheoradec.c:
	  theoradec: segfault on 0-byte ogg_packet in _chain_reverse

2011-07-29 10:23:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/Makefile.am:
	* win32/common/libgsttag.def:
	  Add new GstTagMux base class
	  Hook up new tag muxing base class to build system.
	  https://bugzilla.gnome.org/show_bug.cgi?id=555437
	  API: GstTagMux

2011-07-29 10:22:26 +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:
	* gst-libs/gst/tag/gsttagmux.c:
	* gst-libs/gst/tag/gsttagmux.h:
	  docs: add documentation for GstTagMux

2011-07-28 20:38:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/gsttagmux.c:
	  tagmux: require subclass to install sink pad template
	  Require the subclass to install both source and sink pad
	  templates. Also, print some warnings if the subclass doesn't
	  do that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=555437

2011-07-15 20:57:47 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/gsttagmux.h:
	  tagmux: const-ify GstTagList argument of render vfuncs

2011-07-15 20:39:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/gsttagmux.c:
	* gst-libs/gst/tag/gsttagmux.h:
	  tagmux: fix up private base class header so it can be made public
	  Move private bits into a private struct, add some padding.
	  https://bugzilla.gnome.org/show_bug.cgi?id=555437

2011-07-28 23:31:03 +0100  Michael Smith <msmith@songbirdnest.com>

	* gst-libs/gst/tag/gsttagmux.c:
	* gst-libs/gst/tag/gsttagmux.h:
	  tagmux: add support for end tags
	  Originally "id3tag: Add new id3 tagging plugin, supports v1, v2.3,
	  and v2.4." from gst-plugins-bad. This is an artificial bridge commit.

2010-06-06 18:00:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/tag/gsttagmux.c:
	  ext: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs

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>
	  * gst-libs/gst/tag/gsttagmux.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

2007-09-13 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:
	  * gst-libs/gst/tag/gsttagmux.c:
	  * tests/check/elements/apev2mux.c:
	  Update my mail address.

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:
	  * gst-libs/gst/tag/gsttagmux.c: (plugin_init):
	  * gst-libs/gst/tag/gsttagmux.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.

2006-05-18 12:46:08 +0000  James Doc Livingston <doclivingston@gmail.com>

	  gst-libs/gst/tag/gsttagmux.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>
	  * gst-libs/gst/tag/gsttagmux.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.

2006-05-01 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:
	  * gst-libs/gst/tag/gsttagmux.c:
	  * gst-libs/gst/tag/gsttagmux.h:
	  move taglib-based id3v2muxer to -good.  Fixes #336110.

2006-04-30 16:16:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst-libs/gst/tag/gsttagmux.c:
	  small cleanups
	  Original commit message from CVS:
	  small cleanups

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.

2006-04-28 15:33:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst-libs/gst/tag/gsttagmux.c:
	* gst-libs/gst/tag/gsttagmux.h:
	  pedantic cleanups
	  Original commit message from CVS:
	  pedantic cleanups

2006-04-01 16:50:49 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst-libs/gst/tag/gsttagmux.c:
	  add taglib checks and docs
	  Original commit message from CVS:
	  add taglib checks and docs

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.

2006-03-25 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.

2006-03-20 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.

2006-03-13 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)

2006-03-12 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.

2006-03-11 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 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

2011-07-28 11:21:26 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/encoding/gstencodebin.c:
	  encodebin: rename flags names
	  Rename flags names from native-audio/-video to
	  no-audio/video-conversion to be more explicit on what it does

2011-07-20 18:10:57 +0200  Stefan Sauer <ensonic@google.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: fix latency calculation for live elements
	  Max_latency was computed on already adjusted min_latency. Introduce a new
	  variable for clarity. Spotted by Blaise Gassend.
	  Fixes #644284

2011-07-28 11:44:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: fix max latency calculation
	  ... to allow infinite max, as also claimed by comment.

2011-06-01 10:21:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  baseaudiosink: drop samples that are too late
	  ... rather than having all of them rendered at 0 or subsequently aligned,
	  likely inevitably leading to repeated resyncing.

2011-07-26 13:51:31 +0200  Stefan Sauer <ensonic@google.com>

	* tests/check/pipelines/basetime.c:
	  basetime: fix failing test
	  Always use audiotestsrc as it seems to have been the intention according to the
	  comment header. The test does not work with live-audiosources.

2011-07-25 19:51:24 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/elements/playbin2-compressed.c:
	  tests: rename the test suite to match the binary
	  This unbreaks determining the name for make elements/playbin2-compressed.check
	  from the test output.

2011-07-25 19:39:55 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	  adder: rework pending event handling
	  Use atomic ops on pending flags. Rename the segment_pending to
	  new_segment_pending. Set new_segment_pending not when we received seek, but
	  when we received the first upstream new_segment.

2011-07-25 19:11:59 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/adder/gstadder.c:
	  adder: more debug logging for events

2011-07-26 12:33:56 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Allow all EOS to go through if we don't have a next group
	  Only drop them if the current group isn't drained .. AND there is a
	  next group to switch to.
	  Should Fix #655268

2011-07-25 18:37:15 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin2: Avoid resetting playsink when not needed
	  When we don't have specific {audio|video|text}-sink properties, don't
	  set them on playsink when reconfiguring.
	  If we do that, we end up setting the previous configured sink to
	  GST_STATE_NULL resulting in any potentially pending push being returned
	  with GST_FLOW_WRONG_STATE which will cause the upstream elements to
	  silently stop.
	  https://bugzilla.gnome.org/show_bug.cgi?id=655279

2011-07-25 12:04:02 +0200  Stefan Sauer <ensonic@google.com>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: improve the example
	  Mentioned that this is not ment to be used with subtitles and suggest alternatives.

2011-07-25 10:41:04 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Properly handle multi-stream chains
	  When we have a multi-stream (i.e. audio and video) input and the demuxer
	  adds/removes pads for a new stream (common in a mpeg-ts stream when the
	  program stream mapping is updated), the algorithm for EOS handling was
	  previously wrong (it would only drop the EOS of the *last* pad but would
	  let the EOS on the other pads go through).
	  The logic has only been changed a tiny bit for EOS handling resulting in:
	  * If there is no next group, let the EOS go through
	  * If there is a next group, but not all pads are drained in the active
	  group, drop the EOS event
	  * If there is a next group and all pads are drained, then the ghostpads
	  will be removed and the EOS event will be dropped automatically.

2011-07-23 14:21:27 +0200  Stefan Sauer <ensonic@google.com>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: add example for feeding from stdin

2011-07-23 13:46:31 +0200  Stefan Sauer <ensonic@google.com>

	* tests/check/pipelines/basetime.c:
	  test: print actual timestamp on failure

2011-07-20 13:46:31 +0200  Stefan Sauer <ensonic@google.com>

	* ext/pango/gsttextoverlay.c:
	  textoverlay: keep untimestamped textbuffer until next one
	  Instead of discarding untimestamped text-buffers immeditely after rendering,
	  keep them until we receive the next text buffer.
	  Fixes #654959

2011-07-15 16:46:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/decodebin2.c:
	  tests: add decodebin2 test for parser autoplugging
	  Make sure decodebin2 doesn't try to plug the same parser twice
	  in a row.

2011-07-06 19:40:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/decodebin.c:
	* tests/files/Makefile.am:
	* tests/files/test.mp3:
	  tests: add decodebin1 test for parser autoplugging
	  Make sure decodebin1 doesn't try to plug the same parser twice
	  in a row (so we can change all parsers to accept parsed input as
	  well without breaking applications still using the old decodebin1
	  element).

2011-07-07 15:02:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/playback/gstdecodebin.c:
	  decodebin: don't plug the same parser multiple times in a row
	  This allows us to make parsers accept both parsed and unparsed input
	  without decodebin plugging them in a loop until things blow up, ie.
	  without affecting applications that still use the old playbin or the
	  old decodebin.
	  (Making parsers accept parsed input is useful for later when we want
	  to use parsers to convert the stream-format into something the decoder
	  can handle. It's also much more convenient for application authors
	  who can plug parsers unconditionally in transcoding pipelines, for
	  example).

2011-07-14 13:56:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/pbutils/codec-utils.c:
	* win32/common/libgstpbutils.def:
	  docs: add Since marker to gtk-doc chunk for new codec utils API
	  And add new API to .def file.
	  API: gst_codec_utils_h264_get_level_idc()

2011-03-07 17:55:48 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/codec-utils.c:
	* gst-libs/gst/pbutils/codec-utils.h:
	  codec-utils: Add method to convert H.264 text level in a level_idc

2011-07-09 18:33:38 -0700  David Schleef <ds@schleef.org>

	* ext/ogg/gstoggmux.c:
	  oggmux: check for EOS on both current and best pad
	  Oops, need both.  Fixes #654270.

2011-07-09 18:24:26 -0700  David Schleef <ds@schleef.org>

	* ext/ogg/gstoggmux.c:
	  oggmux: check for EOS on current pad, not best
	  Fixes #654270.

2011-07-09 11:59:42 +0200  Piotr Fusik <fox@scene.pl>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: fixed detection of audio/x-sap
	  Fixes: #654295.
	  Signed-off-by: David Schleef <ds@schleef.org>

2011-06-30 20:33:36 +0200  Luis de Bethencourt <luis@debethencourt.com>

	* gst/encoding/gstencodebin.c:
	  encodebin: fix compiler warning
	  cspace and cspace2 may run uninitialized.

2011-06-29 13:12:49 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/encoding/gstencodebin.c:
	  encodebin: Add flags to disable conversion elements
	  Add a flags property and two flags to allow one to disable the
	  conversion elements within encodebin. Doing so insists that the
	  uncompressed input to encodebin for the appropriate stream type is
	  sufficient to meet the caps requirements of the encoders, muxers and
	  encodebin target.
	  This is mostly beneficial to bypass slow caps negotiations in the
	  conversion elements.

2011-06-29 09:59:05 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst-libs/gst/tag/gstxmptag.c:
	* tests/check/libs/tag.c:
	  tag: xmp: Remove extra chars from end of xmp packet
	  Windows picture viewer is unhappy with extra trailing chars at the
	  end of the xmppacket footer. So remove them as they aren't needed.

2011-06-29 11:30:51 +0200  Robert Swain <robert.swain@collabora.co.uk>

	* gst/encoding/gststreamsplitter.c:
	  streamsplitter: Fix getcaps src pad caps merge
	  Caps returned from gst_pad_peer_get_caps_reffed () may not be writable.
	  If they are not is should cause an assertion in gst_caps_merge (),
	  however, sometimes assertions are disabled in binary builds of -base and
	  it's safer to just be sure the caps are writable. Also, check that the
	  reffed caps pointer is not NULL.

2011-06-15 13:51:31 +0200  Philip Jägenstedt <philipj@opera.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: NULL check in degas_type_find
	  The length check isn't sufficient, an source might
	  report the correct length, but then still fail to
	  read the requested number of bytes for some reason.
	  https://bugzilla.gnome.org/show_bug.cgi?id=652642

2011-06-26 01:06:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/design/design-decodebin.txt:
	  docs: minor addition to decodebin2 design doc

2011-06-26 01:06:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/libs/navigation.c:
	  tests: the navigation interface isn't GstImplementsInterface-wrapped

2011-06-26 00:49:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/interfaces/streamvolume.h:
	  interfaces: GstStreamVolume isn't wrapped by GstImplementsInterface
	  This interface depends on properties and isn't per-instance.

2011-06-26 00:40:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspextension.h:
	  rtsp: GstRTSPExtension isn't wrapped by GstImplementsInterface
	  Fix copy'n'paste error in headers, GstRTSPExtension isn't
	  something that's per-instance.

2011-06-26 00:36:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst-libs/gst/tag/xmpwriter.h:
	  tag: GstXmpWriter doesn't use the GstImplementsInterface
	  No need for per-instance checking of interface implementation here,
	  presumably just a copy'n'paste issue.

2011-06-11 19:03:57 +1000  Jonathan Matthew <jonathan@d14n.org>

	* gst-libs/gst/pbutils/encoding-target.c:
	  encoding-target: set names on audio and video profiles
	  https://bugzilla.gnome.org/show_bug.cgi?id=652342

2011-06-23 11:28:04 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From 69b981f to 605cd9a

2011-06-18 13:32:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Bump git version after unplanned 0.10.35 release
	  Merge branch '0.10.35'
	  Conflicts:
	  configure.ac
	  docs/plugins/inspect/plugin-adder.xml
	  docs/plugins/inspect/plugin-alsa.xml
	  docs/plugins/inspect/plugin-app.xml
	  docs/plugins/inspect/plugin-audioconvert.xml
	  docs/plugins/inspect/plugin-audiorate.xml
	  docs/plugins/inspect/plugin-audioresample.xml
	  docs/plugins/inspect/plugin-audiotestsrc.xml
	  docs/plugins/inspect/plugin-cdparanoia.xml
	  docs/plugins/inspect/plugin-decodebin.xml
	  docs/plugins/inspect/plugin-encoding.xml
	  docs/plugins/inspect/plugin-ffmpegcolorspace.xml
	  docs/plugins/inspect/plugin-gdp.xml
	  docs/plugins/inspect/plugin-gio.xml
	  docs/plugins/inspect/plugin-gnomevfs.xml
	  docs/plugins/inspect/plugin-libvisual.xml
	  docs/plugins/inspect/plugin-ogg.xml
	  docs/plugins/inspect/plugin-pango.xml
	  docs/plugins/inspect/plugin-playback.xml
	  docs/plugins/inspect/plugin-subparse.xml
	  docs/plugins/inspect/plugin-tcp.xml
	  docs/plugins/inspect/plugin-theora.xml
	  docs/plugins/inspect/plugin-typefindfunctions.xml
	  docs/plugins/inspect/plugin-uridecodebin.xml
	  docs/plugins/inspect/plugin-videorate.xml
	  docs/plugins/inspect/plugin-videoscale.xml
	  docs/plugins/inspect/plugin-videotestsrc.xml
	  docs/plugins/inspect/plugin-volume.xml
	  docs/plugins/inspect/plugin-vorbis.xml
	  docs/plugins/inspect/plugin-ximagesink.xml
	  docs/plugins/inspect/plugin-xvimagesink.xml
	  gst-libs/gst/audio/Makefile.am
	  gst/subparse/gstsubparse.c
	  win32/common/_stdint.h
	  win32/common/config.h

2011-06-18 11:16:19 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Allow GError* argument to be NULL
	  This is how other methods taking GError* arguments behave.
	  Fixes #652838