Release 0.10.36RELEASE-0.10.36

7 files changed, 7052 insertions, 18 deletions

diff --git a/ChangeLog b/ChangeLog
index fe859b068..ac8629d89 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,5347 @@
-=== release 0.10.35 ===
+=== 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>
 
-2011-06-15  Tim-Philipp Müller <tim.muller@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:
-	  releasing 0.10.35, "Short Notice"
+	  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
+
+=== release 0.10.35 ===
+
+2011-06-15 19:29:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	  Release 0.10.35
+	  This is an ad-hoc release that is almost identical to 0.10.34:
+	  * work around GLib atomic ops API change
+	  * don't use G_CONST_RETURN in public headers
+	  * subparse: typefinding fixes for subtitles in non-UTF8 charsets
+
+2011-06-15 15:08:32 +0100  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
+
+	* gst-plugins-base.spec.in:
+	  Add gobject introspection files to spec
+
+2011-06-15 14:53:56 +0100  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
+
+	* gst-plugins-base.spec.in:
+	  remove old v4l plugin from spec file
+
+2011-06-15 14:49:41 +0100  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
+
+	* tests/examples/Makefile.am:
+	  Add missing dist subdir
 
 2011-06-15 14:21:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
 
@@ -70,6 +5408,1007 @@
 	  lang: fix possible array overrun
 	  We where checking for i<G_N_ELEMENTS, but where accessing i+1.
 
+2011-06-14 10:31:18 +0530  Debarshi Ray <rishi@gnu.org>
+
+	* gst-libs/gst/pbutils/codec-utils.c:
+	  codec-utils: restore 7350 as a valid sampling frequency for AAC
+	  This was lost during c77f88cac675a1dbb89e40da8e3c28320523bfca.
+
+2011-05-31 22:14:09 -0700  David Schleef <ds@schleef.org>
+
+	* gst/audioresample/resample.c:
+	* gst/audiotestsrc/gstaudiotestsrc.c:
+	* gst/videoscale/vs_4tap.c:
+	* gst/videotestsrc/generate_sine_table.c:
+	* gst/videotestsrc/videotestsrc.c:
+	* tests/icles/test-xoverlay.c:
+	  convert M_PI to G_PI, for msvc
+
+2011-06-07 21:30:18 -0700  David Schleef <ds@schleef.org>
+
+	* ext/ogg/gstoggmux.c:
+	* ext/ogg/gstoggmux.h:
+	  oggmux: refactor how EOS is determined
+	  This decreases the number of buffers held on each pad by one,
+	  eliminating next_buffer.  Simplifies the logic by relying solely
+	  on CollectPads to let us know when a pad is in EOS.  As a side
+	  benefit, the collect pads related code is structured more like
+	  other CollectPad users.
+	  The previous code would occasionally mark the wrong pad as EOS,
+	  causing the code to get in a state where all the streams were
+	  finished, but EOS hadn't been sent to the source pad.
+
+2011-06-09 18:30:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst-libs/gst/tag/gstid3tag.c:
+	* gst-libs/gst/tag/gstvorbistag.c:
+	* gst-libs/gst/tag/tag.h:
+	  libs: replace G_CONST_RETURN with 'const'
+	  G_CONST_RETURN will be deprecated soon.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=652211
+
+2011-06-09 00:02:07 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  tag: xmp: Fix LocationShown syntax
+	  According to the specification, the LocationShown requires its
+	  struct fields to be inside a Bag type.
+
+2011-06-08 14:21:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  tag: xmp: Add room for extra namespace definitions
+	  Adds an extra field to the namespace definitions of the schemas
+	  so they can add the namespace of any array/struct fields they
+	  might use internally.
+
+2011-06-08 12:21:43 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/pango/Makefile.am:
+	* gst/audioresample/Makefile.am:
+	* tests/check/Makefile.am:
+	* tests/examples/v4l/Makefile.am:
+	  GST_PLUGINS_BASE_LIBS is not defined in -base.
+
+2011-06-08 11:33:07 +0200  Christophe Fergeau <cfergeau@redhat.com>
+
+	* tests/examples/audio/Makefile.am:
+	  examples: don't link testchannels example with system libgstaudio
+	  The testchannels audio test program is using -lgstaudio-0.10 to link
+	  with libgstaudio which won't use the gstaudio library that was just
+	  built but the one from the system. This is an issue since it means
+	  we won't be testing the code from the current source tree, and it
+	  also breaks the build when building on a system which don't have
+	  a libgstaudio yet.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=652100
+
+2011-06-08 11:11:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* docs/design/design-decodebin.txt:
+	  docs: add some text about parser/decoder autoplugging issues
+
+2011-06-06 14:41:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst-libs/gst/tag/gsttagdemux.c:
+	  tagdemux: no input data implies no type can be found
+	  ... and posting a proper error message to this effect is appropriately
+	  informative and prevents auto-plugging otherwise stalling.
+
+2011-06-06 12:48:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* ext/ogg/gstoggmux.c:
+	* ext/ogg/gstoggmux.h:
+	  oggmux: determine granulepos metadata using stream mapper whenever possible
+	  ... which unfortunately is not the case for all types, but at least so for
+	  most common ones.
+
+2011-06-06 12:46:05 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: convert incoming buffer timestamp to running time
+	  ... so all subsequent manipulation can take place in the proper timeline
+	  without further ado.
+
+2011-06-01 20:48:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: remove superfluous code
+	  ... since there is nothing in oggstream that cares (or even should)
+	  about granulepos for what is being asked from it.
+
+2011-06-04 13:36:55 -0700  David Schleef <ds@schleef.org>
+
+	* gst/adder/gstadder.c:
+	  adder: Work around changes in g_atomic API
+	  See #651514 for details.
+
+2011-05-31 20:38:56 -0700  David Schleef <ds@schleef.org>
+
+	* gst-libs/gst/pbutils/gstdiscoverer.c:
+	  discoverer: fix c99-ism
+
+2011-06-03 16:29:00 +0200  Luis de Bethencourt <luis.debethencourt@collabora.como>
+
+	* ext/theora/gsttheoraenc.c:
+	  theora: separate encode and push block in chain, into own function.
+
+2011-06-02 19:08:41 +0200  Luis de Bethencourt <luis.debethencourt@collabora.como>
+
+	* ext/theora/gsttheoraenc.c:
+	  theora: use fixed src cap pads
+
+2011-06-02 18:57:05 +0200  Luis de Bethencourt <luis.debethencourt@collabora.como>
+
+	* ext/theora/gsttheoraenc.c:
+	* ext/theora/gsttheoraenc.h:
+	  theora: set the width/height/par on the srcpad caps
+
+2011-06-02 17:29:53 +0200  Luis de Bethencourt <luis.debethencourt@collabora.como>
+
+	* ext/theora/gsttheoraenc.c:
+	  theora: get sink caps info from downstream element pad
+	  https://bugzilla.gnome.org/show_bug.cgi?id=651564
+
+2011-05-27 14:41:39 -0700  Patrick McCarty <patrick.mccarty@intel.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	  pbutils: add description for wbmp images.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=651294
+
+2011-06-02 00:55:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefinding: add typefinder for WAP WBMP bitmaps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=651294
+
+2011-06-02 11:53:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/playback/gstplaysinkaudioconvert.c:
+	* gst/playback/gstplaysinkvideoconvert.c:
+	  playsink: Fix deadlock in the audio/video converter bins when linking fails
+
+2011-06-01 17:31:35 +0200  Edward Hervey <edward.hervey@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	  check: ... and don't forget to add the new arm header
+	  Forgot it in my previous commit
+
+2011-06-01 17:24:30 +0200  Edward Hervey <edward.hervey@collabora.co.uk>
+
+	* tests/check/libs/libsabi.c:
+	* tests/check/libs/struct_arm.h:
+	  libsabi: Add structure sizes for arm
+
+2011-05-31 19:57:57 -0700  David Schleef <ds@schleef.org>
+
+	* gst-libs/gst/fft/gstfftf32.c:
+	* gst-libs/gst/fft/gstfftf64.c:
+	* gst-libs/gst/fft/gstffts16.c:
+	* gst-libs/gst/fft/gstffts32.c:
+	  fft: s/M_PI/G_PI/ for MSVC
+
+2011-05-31 11:05:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/volume/gstvolume.c:
+	* gst/volume/gstvolumeorc.orc:
+	* tests/check/elements/volume.c:
+	  volume: Fix handling of volume>=4.0 for 8 and 16 bit integer formats
+	  Also add a unit test for this. Previously volumes bigger than 4.0
+	  would have resulted in overflows in the fixed point processing.
+	  Fixes bug #649642.
+
+2011-05-29 13:32:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* tests/check/elements/adder.c:
+	* tests/check/elements/ffmpegcolorspace.c:
+	* tests/check/elements/vorbistag.c:
+	* tests/check/libs/rtp.c:
+	* tests/check/pipelines/theoraenc.c:
+	  tests: fix some more unused-but-set-variable warnings with gcc 4.6
+
+2011-05-28 16:14:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* win32/common/libgstvideo.def:
+	  win32: update .def file for new API
+
+2011-05-28 12:39:06 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* Makefile.am:
+	* tests/check/elements/.gitignore:
+	  Ignore new playbin2-compress test binary
+	  And add old testchannels binary to CRUFT_FILES.
+
+2011-05-27 23:31:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst-libs/gst/video/video.h:
+	  video: sprinkle some G_GNUC_CONST
+	  Mark functions that have no effect besides their return value and
+	  only inspect their input arguments with G_GNUC_CONST. (We just
+	  ignore the g_return_val_if_fail() guards for this)
+
+2011-05-27 23:25:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst-libs/gst/video/video.h:
+	  video: clean up header file
+	  Sprinkle some spaces and newlines here and there.
+
+2011-05-27 15:03:19 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* configure.ac:
+	* gst-libs/gst/audio/.gitignore:
+	* gst-libs/gst/audio/Makefile.am:
+	* gst-libs/gst/audio/testchannels.c:
+	* tests/examples/Makefile.am:
+	* tests/examples/audio/.gitignore:
+	* tests/examples/audio/Makefile.am:
+	* tests/examples/audio/testchannels.c:
+	  audio: move testchannels example to 'tests/examples' dir
+	  Also fix it up a little to not include 'c' file but link to the libs instead.
+
+2011-05-27 11:39:21 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/pbutils/codec-utils.c:
+	  code-utile: fix level descriptions for fgs
+	  fgs levels range from 8-13 and are mapped to 0-5.
+
+2011-05-25 14:38:21 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/pbutils/codec-utils.c:
+	  codec-utils: fix mpeg4 level verification
+	  The current condition would never be true. As levels<6 are asp and levels>7 and
+	  <14 are fgs, we should return NULL for cases 6,7,14,15.
+
+2011-05-26 12:33:08 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* sys/xvimage/xvimagesink.c:
+	  xvimagesink: Fallback to non-XShm mode if allocating the XShm image failed
+	  Fixes bug #630442.
+
+2011-05-26 12:30:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* sys/ximage/ximagesink.c:
+	  ximagesink: Fallback to non-XShm mode if allocating the XShm image failed
+	  Fixes bug #630442.
+
+2011-05-26 11:41:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin2: Let the input-selectors sync all streams to the running time
+	  This is especially needed when switching between a non-sparse and sparse
+	  video stream, see bug #537382. It also lowers the time needed for switching
+	  between streams a bit.
+
+2011-01-20 00:52:50 -0700  Lane Brooks <dirjud@gmail.com>
+
+	* ext/pango/gsttextoverlay.c:
+	* ext/pango/gsttextoverlay.h:
+	  textoverlay: added 'outline-color' parameter to control whether text gets a shadow
+
+2011-01-20 00:42:39 -0700  Lane Brooks <dirjud@gmail.com>
+
+	* ext/pango/gsttextoverlay.c:
+	* ext/pango/gsttextoverlay.h:
+	  textoverlay: added 'shadow' option to control whether text gets a shadow
+
+2011-05-26 10:48:05 +0200  Jindrich Makovicka <makovick@gmail.com>
+
+	* ext/pango/gsttextrender.c:
+	  textrender: Correctly negotiate with downstream instead of just using random caps
+	  Fixes bug #638897.
+
+2011-05-26 10:43:51 +0200  Jindrich Makovicka <makovick@gmail.com>
+
+	* ext/pango/gsttextrender.c:
+	  textrender: Add bound checks to not write outside the image area
+
+2011-05-26 10:42:46 +0200  Jindrich Makovicka <makovick@gmail.com>
+
+	* ext/pango/gsttextrender.c:
+	  textrender: Prevent double unref of caps if the caps can't be set on the srcpad
+
+2011-05-26 10:31:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/gnomevfs/gstgnomevfssrc.c:
+	  gnomevfssrc: Keep track of interruptions during read with a flag
+
+2010-09-03 09:11:30 -0400  American Dynamics <GStreamer-Bugs@tycosp.com>
+
+	* ext/gnomevfs/gstgnomevfssrc.c:
+	* ext/gnomevfs/gstgnomevfssrc.h:
+	  gnomevfssrc: Add support for cancelling the read operations
+	  This allows the state change from PAUSED to READY to be faster.
+	  Fixes bug #628337.
+
+2011-05-25 14:14:46 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+	* sys/ximage/ximagesink.c:
+	  ximagesink: Remove g_assert from interface query
+
+2011-05-25 14:08:43 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+	* sys/xvimage/xvimagesink.c:
+	  xvimagesink: Remove the g_assert from interface query
+
+2011-05-26 00:17:40 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst/audiotestsrc/gstaudiotestsrc.c:
+	* gst/audiotestsrc/gstaudiotestsrc.h:
+	  audiotestsrc: add blue and violet noise by using spectral inversion
+	  Add blue and violet noise by spectral inversion of pink and red noise.
+	  Fixes #649969
+
+2011-05-25 23:40:26 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst/audiotestsrc/gstaudiotestsrc.c:
+	* gst/audiotestsrc/gstaudiotestsrc.h:
+	  audiotestsrc: add red (brownian) noise generator
+	  Add another noise generator which produces a quite dark noise color.
+	  Fixes parts of #649969.
+
+2010-09-27 13:32:31 +0400  Vladimir Eremeev <eremeev@atlantis.ru>
+
+	* tests/examples/seek/seek.c:
+	  seek: set selected/default audio/video sinks on playbin and playbin2
+	  https://bugzilla.gnome.org/show_bug.cgi?id=630322
+
+2011-05-25 19:03:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* tests/examples/seek/seek.c:
+	  seek: add --audiosink and --videosink command line options
+
+2011-05-25 18:50:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* tests/examples/seek/seek.c:
+	  seek: use the right GDK defines to differentiate between the backends
+
+2011-05-25 18:45:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* tests/examples/seek/seek.c:
+	  seek: use gst_filename_to_uri() to convert a filename to a uri
+
+2010-09-27 12:46:54 +0400  Vladimir Eremeev <eremeev@atlantis.ru>
+
+	* tests/examples/seek/seek.c:
+	  seek: make seek example work in win32
+	  https://bugzilla.gnome.org/show_bug.cgi?id=630322
+
+2011-05-25 16:08:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	  configure: update GLib requirement to >= 2.24
+	  Same as core (make implicit requirement explicit).
+	  http://gstreamer.freedesktop.org/wiki/ReleasePlanning/GLibRequirement
+
+2011-05-25 15:24:33 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/theora/gsttheoraenc.c:
+	  theoraenc: remove bogus <0 check for unsigned var
+	  bytes_written is a gsize which is unsigned and thus never < 0.
+
+2011-05-25 15:23:13 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/theora/gsttheoraenc.c:
+	  theoraenc: fix variable type for bytes_consumed
+	  th_encode_ctl() returns an int. Using a gsize result in bogus <0 checks.
+
+2011-05-25 15:04:20 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/riff/riff-read.c:
+	  riff: remove the g_return_if_fail as we test it below
+	  We don't want to return without setting taglist=NULL if asserts are on and with
+	  setting taglist=NULL otherwise.
+
+2011-05-25 14:28:18 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst/volume/gstvolume.c:
+	  volume: use a flag for 'mute' using the controller
+	  Previously we checked mute_csource to determine wheter we need to premultiply
+	  volumes and mute values. That fails as we unrefs mute_csource and set it to
+	  NULL after. Use an extra flag instead.
+
+2011-05-25 14:12:50 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/tag/gstexiftag.c:
+	  exiftag: reflow the code
+	  Move the warning on unsupported units to the swicth-case. Move fetching the
+	  pending tags down to where we use them.
+
+2011-05-25 13:59:57 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/tag/gstexiftag.c:
+	  exiftag: set value=1 if we found the token
+	  Otherwise we never write the tag. This would also be consistent with the code in
+	  deserialize_scene_type().
+
+2011-05-25 12:30:51 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* sys/xvimage/xvimagesink.c:
+	  xvimagesink: run gst-indent
+
+2011-05-25 12:29:21 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* sys/xvimage/xvimagesink.c:
+	  xvimagesink: remove unneded !=NULL checks
+	  We check for matching_attr!=NULL right before already.
+
+2011-05-24 00:13:04 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	  docs: massage the section file more
+	  Add more symbols (from unused.txt). Move the whole bunch of riff-fourcc defines
+	  to std section too (no one is hoing to document them, right).
+
+2011-05-24 00:12:26 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/video/video.c:
+	  docs: add missing parameter docs
+
+2011-05-23 23:53:38 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	  docs: move the riff structure to std-section
+	  If someone intents to document them and the fields we can move them back.
+
+2011-05-23 23:53:06 +0300  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/pbutils/gstpluginsbaseversion.c:
+	  docs: move pluginbaseversion to separate section as we have section docs
+
+2011-05-23 23:51:15 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/rtsp/gstrtspdefs.h:
+	  docs: add minimal docblobs for status code and headers
+	  Use a trick to avoid documenting all 100 enums.
+
+2011-05-23 23:41:56 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/interfaces/xoverlay.c:
+	* gst-libs/gst/interfaces/xoverlay.h:
+	  docs: update xoverlay docs for api addition and deprecation
+
+2011-05-23 23:12:50 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+	* gst-libs/gst/rtp/gstbasertpdepayload.c:
+	* gst-libs/gst/rtp/gstbasertpdepayload.h:
+	* gst-libs/gst/rtp/gstbasertppayload.c:
+	* gst-libs/gst/rtp/gstbasertppayload.h:
+	  docs: rtp library docs update
+
+2011-05-23 22:58:22 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/audio/gstringbuffer.h:
+	* gst-libs/gst/cdda/gstcddabasesrc.h:
+	* gst-libs/gst/interfaces/colorbalance.h:
+	* gst-libs/gst/interfaces/colorbalancechannel.h:
+	* gst-libs/gst/interfaces/mixer.h:
+	* gst-libs/gst/interfaces/mixeroptions.h:
+	* gst-libs/gst/interfaces/navigation.h:
+	* gst-libs/gst/interfaces/tuner.h:
+	* gst-libs/gst/video/gstvideofilter.h:
+	* gst-libs/gst/video/gstvideosink.h:
+	  docs: add missing documentation for various pieces
+
+2010-02-19 12:54:18 +0100  Thijs Vermeir <thijsvermeir@gmail.com>
+
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	  baseaudiosink: recalibrate clock on setcaps
+	  Because the spec for the ringbuffer can change when changing
+	  the caps, we must recalibrate the clock.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=610443
+
+2011-05-23 16:02:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/subparse/gstsubparse.c:
+	  subparse: Try to typefind even if conversion to UTF8 failed
+	  Fixes bug #600043.
+
+2011-05-23 16:02:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/subparse/gstsubparse.c:
+	  subparse: Compile the typefind regex with optimization to speed up matching
+
+2011-05-23 15:51:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/subparse/gstsubparse.c:
+	  subparse: Interprete typefind strings passed to GRegex as raw bytes instead of valid UTF8
+
+2011-05-23 15:21:59 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/video/convertframe.c:
+	  convertframe: fix docs
+	  Fixup paramter mismatch between func and prototype. Add missing parameter docs.
+
+2011-05-23 15:08:24 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/audio/gstaudioclock.h:
+	* gst-libs/gst/audio/gstaudiofilter.h:
+	* gst-libs/gst/audio/gstaudiosrc.h:
+	* gst-libs/gst/audio/multichannel.h:
+	  docs: fixup audio-library docs
+
+2011-05-23 15:02:27 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/app/gstappsink.c:
+	* gst-libs/gst/app/gstappsrc.c:
+	* gst/app/gstapp.c:
+	  docs: fixup appsrc/sink api docs
+
+2011-05-23 14:53:26 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/audio/gstaudioiec61937.c:
+	* gst-libs/gst/audio/gstaudioiec61937.h:
+	  docs: fix docs for new api
+	  Some parameters where wrong, first line missed the ':' and return docs where
+	  broken.
+
+2011-05-23 14:45:23 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* docs/libs/gst-plugins-base-libs-docs.sgml:
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	  docs: update xmp api docs
+	  Add missing section. Add new section to main-sgml. Add missing function.
+
+2011-05-23 14:07:38 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  xmptag: remove late check
+	  We deref the pointer two lines before already and besides this internal function
+	  should not be called with this parameter=NULL.
+
+2011-05-23 14:01:29 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  xmptag: have the default branch as the last one
+
+2011-05-23 14:00:04 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  xmptag: an uint value can't be <0
+
+2011-05-23 13:53:06 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/video/video.c:
+	  whitespace: trim trailing whitespace
+
+2011-05-23 13:50:59 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/video/video.c:
+	  video.c: use a break and a final warning instead of early returns
+	  Use breaks for case branches instead of return 0. We don't expect these to
+	  happen anyway. Thus have a warning before the final return to make it easier to
+	  see when things go out of sync.
+
+2011-05-23 13:49:01 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/video/video.c:
+	  video.c: use g_assert_not_reached() for logical error here.
+	  This will help to detect them closer to the source if they ever happen.
+
+2011-05-20 10:48:39 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst-libs/gst/tag/lang.c:
+	  lang: fix possible array overrun
+	  We where checking for i<G_N_ELEMENTS, but where accessing i+1.
+
+2011-05-19 23:41:08 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst/audioconvert/gstaudioconvert.c:
+	  audioconvert: cleanup helper code
+	  make_lossless_changes() returns the same structure that we're passing (probably
+	  to enable chaining). Instead of reusing s and making it point to s2 as well,
+	  keep using s2. Drop the assignment which in the 2nd case is a dead one anyway.
+
+2011-05-19 23:25:24 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* docs/plugins/gst-plugins-base-plugins.args:
+	* docs/plugins/gst-plugins-base-plugins.hierarchy:
+	* docs/plugins/gst-plugins-base-plugins.interfaces:
+	* docs/plugins/gst-plugins-base-plugins.prerequisites:
+	  docs: update plugin introspection data
+	  Now more files are merged and produced in a canonical fashion, which hopefully
+	  creates less or no delta in the future.
+
+2011-05-19 22:56:53 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From 9e5bbd5 to 69b981f
+
+2011-05-19 13:40:29 +0100  Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
+
+	* gst-plugins-base.spec.in:
+	  Add new header file
+
+2011-05-19 08:30:14 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst/encoding/gstencodebin.c:
+	  encodebin: Autoplug formatters
+	  Autoplug formatters for streams if a formatter with secondary or
+	  higher rank is found. Formatters are autoplugged when there is no
+	  muxer or when the muxer doesn't implement the tagsetter interface.
+	  Currently only the first formatter found is plugged, this might
+	  help in lots of cases, but it doesn't solve the
+	  'lamemp3 ! xingmux ! id3mux'
+	  case.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=649841
+
+2011-05-19 08:27:29 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst/encoding/gstencodebin.c:
+	  encodebin: fix typos
+
+2011-05-18 22:07:58 +0200  Aleix Conchillo Flaque <aleix@oblong.com>
+
+	* ext/vorbis/gstvorbisdec.c:
+	  vorbisdec: Handle headers in caps
+
+2011-05-18 16:09:47 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From fd35073 to 9e5bbd5
+
+2011-05-18 13:18:15 +0200  Robert Swain <robert.swain@collabora.co.uk>
+
+	* gst-libs/gst/video/video.c:
+	* gst-libs/gst/video/video.h:
+	* tests/check/libs/video.c:
+	  gstvideo: Add gst_video_get_size_from_caps function
+	  gst_video_get_size_from_caps () allows easy calculation of the raw video
+	  buffer size from some fixed video caps.
+	  API: gst_video_get_size_from_caps()
+
+2011-05-18 12:24:02 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From 46dfcea to fd35073
+
+2011-05-18 09:34:52 +0200  Robert Swain <robert.swain@collabora.co.uk>
+
+	* ext/alsa/gstalsasrc.c:
+	  alsa: Remove unused but set variable
+	  Unused but set variables cause warnings in GCC 4.6.x and newer.
+
+2011-05-17 10:20:36 +0200  Edward Hervey <edward.hervey@collabora.co.uk>
+
+	* gst-libs/gst/rtsp/gstrtspconnection.c:
+	  rtsp: Fix typo which broke the build
+
+2011-05-16 15:35:50 +0200  Miguel Angel Cabrera Moya <madmac2501@gmail.com>
+
+	* gst-libs/gst/rtsp/gstrtspconnection.c:
+	  rtspconnection: not enter in not controllable state unless it is necessary
+	  When closing rtspsrc the state change blocks until the polling in the
+	  connection timeouts. This is because the second time we loop to read a
+	  full message controllable is set to FALSE in the poll group, even though no
+	  message is half read.
+	  This can be avoided by not setting controllable to FALSE the poll group
+	  unless we had begin to read a message.
+	  Fixes #610916
+
+2010-05-30 13:21:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/cdparanoia/gstcdparanoiasrc.c:
+	* ext/cdparanoia/gstcdparanoiasrc.h:
+	  cdparanoiasrc: fix build on OSX by #undef-ing VERSION before including system headers
+	  On OSX the cdparanoia headers include IOKit framework headers (in particular
+	  SCSICmds_INQUIRY_Definitions.h) which define a structure that has a member
+	  named VERSION, so we must #undef VERSION before including those for things
+	  to compile on OSX.
+	  Fixes #609918.
+
+2011-05-02 11:43:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/videorate/gstvideorate.c:
+	* gst/videorate/gstvideorate.h:
+	  videorate: optionally ensure maximum average output frame rate
+	  See #628764.
+
+2011-04-29 14:58:02 +0200  Alexey Fisher <bug-track@fisher-privat.net>
+
+	* gst/videorate/gstvideorate.c:
+	* gst/videorate/gstvideorate.h:
+	  videorate: optionally only drop frames to ensure maximum frame rate
+	  This adds option to arrange for maximal allowed variable frame rate.
+	  Fixes #628764.
+
+2011-04-26 13:37:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: use bitrate to configure streaming buffer-duration default case
+	  In particular, in audio only cases whose (estimated) metadata provides bitrate
+	  information, the buffer-size based on such bitrate (and buffer-duration)
+	  will be much more reasonable than queue2 default buffer-size.
+
+2011-04-26 11:27:40 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: remove some dead code
+	  ... which was dead as pads were never added to the list, and need not be added,
+	  since removing them is handled by a pad callback.
+
+2011-04-29 11:48:02 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* tests/examples/encoding/Makefile.am:
+	  encodebin: examples: Add missing base libs to makefile
+
+2011-04-28 10:58:15 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst/encoding/gstencodebin.c:
+	  encodebin: Check for missing converters
+	  Adds checks for missing video and audio converter elements
+
+2011-04-27 22:05:55 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst-libs/gst/tag/xmpwriter.c:
+	  tag: xmpwriter: Rename documentation headers
+	  Fix some wrong documentation headers from the first name
+	  given to this interface.
+
+2011-04-19 08:41:53 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* tests/check/libs/tag.c:
+	  tests: xmp: New tests for the Iptc4xmpExt tags
+
+2011-04-18 23:28:13 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  tag: xmp: Add Iptc4xmpExt schema support
+	  Adds Iptc4xmpExt schema with country, city and sublocation
+	  tags mapped
+
+2011-04-19 11:00:24 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  tag: xmp: Add support for reading struct tags
+	  Adds a context variable that controls if the parsing is on
+	  'top level' tags or inside a struct tag.
+
+2011-04-18 16:54:54 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  tag: xmp: Add struct xmp tag type support
+	  Adds support for writing the xmp struct tag type, it is a compound tag
+	  that has inner tags.
+
+2011-04-18 23:16:59 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  tag: xmp: Fixing schema maps
+	  Do not forget to create a new schema for every supported schema
+	  instead of reusing the same object
+
+2011-04-18 10:20:00 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+	* gst-libs/gst/tag/gstxmptag.c:
+	  tag: xmp: Write the same tag to all schemas
+	  Instead of writing only the xmp tag for the first found entry
+	  that matches the gstreamer tag, look for all mappings to write
+	  the tag to different schemas.
+	  The rationale here is that some reader application might only
+	  be interested on a particular schema tags, so we should try
+	  to write as many tags for all schemas.
+
+2011-05-15 13:39:18 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* win32/common/libgstaudio.def:
+	  win32: Update libgstaudio.def for new symbols
+
+2011-05-14 17:27:30 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstringbuffer.c:
+	  baseaudiosink: Use g_str_equal() instead of strncmp()
+	  The strncmp is unnecessary anyway since one of the strings is a const
+	  string.
+
+2011-05-14 16:49:53 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	  baseaudiosink: Fix trivial indentation problems
+
+2011-03-07 20:49:16 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+	* docs/libs/gst-plugins-base-libs-docs.sgml:
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/audio/Makefile.am:
+	* gst-libs/gst/audio/gstaudioiec61937.c:
+	* gst-libs/gst/audio/gstaudioiec61937.h:
+	  audio: Add an IEC 61937 payloading library
+	  This can be used by sinks to take compressed formats, correctly payload
+	  these in IEC 61937 frames and feed these to sinks that support
+	  passthrough output over IEC 60958 (S/PDIF) or, in the case of MP3, over
+	  Bluetooth.
+	  Initial implementation includes AC3, E-AC3, MPEG-1, MPEG-2 (non-AAC),
+	  and DTS (type-I/II/II) payloading. More formats can be added as needed.
+	  API: gst_audio_iec61937_frame_size()
+	  API: gst_audio_iec61937_payload()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=642730
+
+2011-03-09 11:12:39 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	* gst-libs/gst/audio/gstbaseaudiosink.h:
+	  baseaudiosink: Allow subclasses to provide payloaders
+	  This allows subclasses to provide a "payload" function to prepare
+	  buffers for consumption. The immediate use for this is for sinks that
+	  can handle compressed formats - parsers are directly connected to the
+	  sink, and for formats such as AC3, DTS, and MPEG, IEC 61937 patyloading
+	  might be used.
+	  API: GstBaseAudioSinkClass:payload()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=642730
+
+2011-04-09 09:49:10 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstringbuffer.c:
+	  ringbuffer: Add support for E-AC3
+	  Adds support for pushing E-AC3 buffers and doing bytes-to-ms conversion
+	  correctly. The assumption (as with other formats) is that something like
+	  IEC 61937 payloading will be used. Correspondingly the ringbuffer spec
+	  is populated so that the data rate is 4x normal AC3.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=642730
+
+2011-03-14 15:51:40 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstringbuffer.c:
+	  ringbuffer: Add support for MPEG audio buffers
+
+2011-03-14 15:49:57 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstringbuffer.h:
+	  ringbuffer: Add AAC format types
+	  These are meant to be used for buffers containing AAC data. Nothing uses
+	  this yet, but for now it serves to distinguish from GST_BUFTYPE_MPEG
+	  which represents non-AAC MPEG audio.
+	  API: GST_BUFTYPE_MPEG2_AAC
+	  API: GST_BUFTYPE_MPEG4_AAC
+
+2011-03-09 22:57:00 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstringbuffer.c:
+	  ringbuffer: Add support for DTS buffers
+
+2011-05-14 11:42:50 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* configure.ac:
+	  configure: Require core 0.10.34.1 for the new ghostpad API
+
+2011-05-09 22:20:23 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin2: fix preroll for streams at low bitrates
+	  For streams at low bitrates we need to set a limit in time because the limit
+	  in bytes might not reached too late, sometimes more than 30 seconds.
+	  This limit can only be set if upstream is seekable (see #584104)
+	  Closes #647769
+
+2011-05-09 13:11:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/playback/gstplaysink.c:
+	  playsink: Use new ghostpad/proxypad API to get the internal pad
+
+2011-05-09 12:59:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/playback/gstplaysinkaudioconvert.c:
+	* gst/playback/gstplaysinkaudioconvert.h:
+	* gst/playback/gstplaysinkvideoconvert.c:
+	* gst/playback/gstplaysinkvideoconvert.h:
+	  playsink: Use new ghostpad/proxypad API
+
+2011-05-09 12:50:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* tests/check/elements/playbin2-compressed.c:
+	  playbin2: Disable some compressed stream tests that are racy without a stream-activate event
+
+2011-03-29 19:15:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* tests/check/elements/playbin2-compressed.c:
+	  playbin2: Reset buffer counter in playbin2-compressed tests every time when going to READY
+
+2011-03-25 08:26:00 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/playback/Makefile.am:
+	* gst/playback/gstplaysink.c:
+	* gst/playback/gstplaysinkaudioconvert.c:
+	* gst/playback/gstplaysinkaudioconvert.h:
+	* gst/playback/gstplaysinkvideoconvert.c:
+	* gst/playback/gstplaysinkvideoconvert.h:
+	  playsink: Add audio and video converter convenience bins
+	  These reconfigure based on the caps and plugin in converters if
+	  necessary. This also makes switching between compressed and raw
+	  streams work flawlessly without loosing the states of any element
+	  somewhere or having running time problems.
+
+2011-03-15 12:51:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/playback/gstplaybin2.c:
+	* gst/playback/gstplaysink.c:
+	* gst/playback/gstplaysink.h:
+	  playbin2/playsink: Decide if A/V caps are raw only inside playsink
+	  Before playbin2 would use different selectors for raw audio and
+	  compressed audio (and the same for video) and used different
+	  pads from playsink. This made the involved logic much more
+	  complex and was not implemented completely in playsink, which
+	  made it impossible to support files with a compressed and
+	  uncompressed stream that is support by the sink.
+	  playbin2 handles raw/non-raw streams the same now and the
+	  decision is left to playsink, which now can also handle
+	  caps changes from raw to non-raw and the other way around.
+	  Fixes bug #632788.
+
+2011-03-15 11:41:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/playbin2-compressed.c:
+	  playbin2: Add unit test for compressed stream support in playbin2/playsink
+
+2011-05-09 12:56:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/alsa/gstalsasrc.c:
+	  alsasrc: Fix some compilation errors
+
+2011-05-09 11:50:05 +0200  Pontus Oldberg <pontus.oldberg@invector.se>
+
+	* ext/alsa/gstalsasrc.c:
+	* ext/alsa/gstalsasrc.h:
+	  alsasrc: Improve timestamp accuracy
+	  Fixes bug #635256.
+
+2011-05-06 17:01:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/playback/gstsubtitleoverlay.c:
+	* gst/playback/gstsubtitleoverlay.h:
+	  subtitleoverlay: Use new, public ghostpad functions
+
+2011-05-03 11:26:32 +0300  Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
+
+	* sys/xvimage/xvimagesink.c:
+	  xvimagesink: Use GST_BOILERPLATE
+
+2011-05-14 09:41:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	* docs/plugins/gst-plugins-base-plugins.hierarchy:
+	* 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:
+	* win32/common/_stdint.h:
+	* win32/common/config.h:
+	  Back to development
+
 === release 0.10.34 ===
 
 2011-05-14 01:00:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
diff --git a/NEWS b/NEWS
index 6ee1d4f9e..94583e827 100644
--- a/NEWS
+++ b/NEWS
@@ -1,4 +1,322 @@
-This is GStreamer Base Plug-ins 0.10.35, "Short Notice"
+This is GStreamer Base Plug-ins 0.10.36, "Better"
+
+Changes since 0.10.35:
+
+      * audio: new IEC 61937 payloading library
+      * audio: new GstAudioFormat, GstAudioFormatInfo and GstAudioInfo API
+      * audio: new GstAudioDecoder and GstAudioEncoder base classes
+      * audio: baseaudiosink: allow subclasses to provide payloaders
+      * audio: baseaudiosink: fix latency calculation for live elements
+      * audio: baseaudiosink: make discont-wait configurable
+      * audio: baseaudiosink: split "drift-tolerance" into "alignment-threshold" 
+      * codec-utils: Add method to convert H.264 text level in a level_idc
+      * discoverer: add support for subtitles; try harder to extract language and duration
+      * encoding-profile: add function to create a profile from a discoverer info
+      * ringbuffer: add support for AAC, DTS, E-AC3 and MPEG audio buffers 
+      * rtcpbuffer: Add feedback message types from RFC 510
+      * rtcpbuffer: prevent overflow of 16bit header length
+      * rtspconnection: make hostname lookup thread-safe; OSX portability fixes
+      * rtspconnection: only send new data immediately if there are no queued messages
+      * tags: add new GstTagMux base class
+      * tags: add convenience API to handle creative commons licenses
+      * tags: add API to parse ID3v2 tags
+      * tags: various exif and xmp tag writing fixes
+      * tags: xmp: add Iptc4xmpExt schema support
+      * tags: gstvorbistag: map ENCODER Vorbis comment to application-name
+      * video: add video overlay composition API for subtitles
+      * video: fix a RGB ordering mixup in colorspace conversion code
+      * alsasink: fix high sample rates being rejected, and negotiation to "nearest" rate
+      * audioresample: don't emit DISCONT buffers if no discontinuity happened
+      * audioresample: fix quality setting being ignored; use SSE/SSE2 when possible
+      * audiotestsrc: add red (brownian) and blue/violet noise generator
+      * cdparanoiasrc: fix build issue on OSX (caused by broken cdparanoia port and broken system headers)
+      * decodebin2: improve handling of multi-stream chains (e.g. mpeg-ts)
+      * decodebin2: Cache the upstream seekability for demuxer decode chains and use it for the non-preroll multiqueue limits
+      * decodebin2: add support for autoplugging parsers and parser-converters, and negotiate stream-format conversions properly as needed
+      * decodebin2: link elements before testing if they can reach the READY state, so allow hw-accelerated elements can query the video context
+      * decodebin2: use a TIME limit for pre-rolling in live streams and not in non-live streams
+      * decodebin2: fix preroll for HLS streams at low bitrates
+      * decodebin2: add source pads to stream-topology element messages
+      * decodebin, decodebin2: don't plug the same parser multiple times in a row, so we can make parsers accept parsed input as well (and use them to convert to different stream formats)
+      * encodebin: add flags to disable conversion elements
+      * encodebin: autoplug formatters; re-enable parsers
+      * gnomevfssrc: add support for cancelling read operations
+      * oggdemux, oggmux: add support for new Opus audio codec
+      * oggdemux: implement push mode seeking (e.g. for http)
+      * oggdemux: assume input is live stream if byte size cannot be determined
+      * oggdemux: fix hang on small truncated files
+      * oggmux: add skeleton write support
+      * oggmux: sync  input streams and select input buffers based on running time
+      * oggmux: headers should always have granpos 0
+      * oggmux: refactor how EOS is determined
+      * oggmux: support sparse streams as input (e.g. kate subtitle streams)
+      * playbin2: fix decoder-sink compatibility check for raw audio/video formats
+      * playbin2: make sure that the decoders we plug are compatible with the fixed sink
+      * playsink: Add audio and video converter convenience bins
+      * playbin2: improve stream switching
+      * playbin2/playsink: Decide if A/V caps are raw only inside playsink
+      * playbin2/playsink: better support for raw + compressed streams (audio passthrough)
+      * playbin2/playsink: improve handling of "non-raw" formats (for hw-accelerated video decoding)
+      * playsinkconvertbin: Reconfigure if we switch from raw to incompatible raw caps
+      * subparse: subtitle format typefinding improvements
+      * subtitleoverlay: handle non raw video streams (add suport for hardware accelerated videos)
+      * textoverlay: support more video formats
+      * textoverlay: add "outline-color" and "shadow" properties
+      * textoverlay: attach GstVideoOverlayComposition to buffers if input is not raw video
+      * theoraenc: do not automatically override quality when using target bitrate
+      * theoraenc: proxy downstream caps restrictions upstream
+      * typefinding: extract SOF marker in jpeg typefinder (to distinguish lossless JPEG)
+      * typefinding: add typefinder for WAP WBMP bitmaps (mostly to avoid false positives)
+      * typefinding: typefind UTF-16 and UTF-32 with BOMs (to avoid false positives)
+      * typefinding: recognize Asylum modules
+      * videorate: add a "max-rate" property; optionally ensure maximum average output frame rate
+      * videorate: add "force-fps" property to force an output framerate or change it on the fly
+      * videorate: optionally only drop frames to ensure maximum frame rate
+      * videoscale: add modified Lanczos scaling method
+      * volume: Fix handling of volume>=4.0 for 8 and 16 bit integer formats
+      * vorbisenc: relax overly tight jitter tolerances (make it work better with non-perfect input streams)
+      * xvimagesink, ximagesink: fall back to non-XShm mode if allocating the XShm image failed
+
+Bugs fixed since 0.10.35:
+
+      * 643202 : [encodebin] streamcombiner not completely implemented
+      * 654270 : oggmux unit test fails after latest changes
+      * 658984 : Fix typos in gst-plugins-base
+      * 555437 : [tag] add GstTagMux base class
+      * 556648 : [typefind] detect lossless jpeg
+      * 563251 : oggmux should have option to create Ogg Skeleton stream
+      * 584811 : playbin2's get-text-tags sometimes fails in text stream 0
+      * 607619 : [typefind] utf-16 text file mistakenly identified as layer 1 mpeg audio
+      * 607742 : API: add gst_event_new_{upstream,downstream}_force_key_unit() etc.
+      * 609918 : [OS X] configure: cdda_interface.h: present but cannot be compiled (if VERSION is defined)
+      * 610443 : baseaudiosink: clock can jump on setcaps
+      * 612443 : oggdemux: only use information from skeleton if we have nothing better
+      * 615131 : playing an ogg over http does not report duration correctly
+      * 615342 : [gstalsamixer] leaks
+      * 621897 : [oggdemux] reports wrong duration, and push mode seeking support
+      * 628337 : [gnomevfssrc] Add support for cancelling read operations
+      * 628764 : [videorate] add new option for max frame rate
+      * 629212 : [oggdemux] Improve support for push mode (seeking, duration)
+      * 630322 : make seek example work with windows
+      * 630442 : xvimagesink, ximagesink: fallback to X*CreateImage() if X*ShmCreateImage() fails
+      * 630497 : [seek] sprinkle GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS to dump pipeline to dot file
+      * 632788 : [playbin2] Doesn't support files with a streams that are supported compressed by a sink and streams that need decoding
+      * 635556 : [oggdemux] bad duration estimate in streaming mode with vertical-overview.ogg
+      * 637812 : vorbisenc: choppy sound due to input timestamp jitter
+      * 638897 : [textrender] allow setting the canvas size using peer caps + bugfixes
+      * 639055 : discoverer: add support for subtitle streams
+      * 640041 : textoverlay: Added parameters to control text outline color and whether shadowing is enabled
+      * 640564 : Remuxing a Theora stream generates a stream that oggz-validate complains about
+      * 640859 : basesink incorrectly categorizes timestamp jitter as drift
+      * 642690 : [baseaudio] GstBaseAudioEncoder and GstBaseAudioDecoder class
+      * 642878 : encoding-profile: add a function to create a profile from a discoverer info
+      * 643578 : [encodebin] - broken remuxing
+      * 644284 : Suspicious max_latency computation in gstbaseaudiosink.c
+      * 647648 : videorate: support for caps modifications in a running pipeline
+      * 647769 : [decodebin2] Fix preroll for streams at low bitrate
+      * 649319 : Add boiler plate code to xvimagesink
+      * 649642 : [volume] Overflows with volume > =4.0 and 8/16 bit integer formats
+      * 649969 : [audiotestsrc] Add more noise variants
+      * 650406 : vorbisdec does not handle headers in caps
+      * 651089 : [xvimagesink/ximagesink] Remove g_assert from Interface query
+      * 651294 : WBMP images are not supported by typefind
+      * 651496 : encodebin seems to fail to pick up container variant
+      * 651615 : [vorbisenc] Too small jitter tolerance
+      * 651788 : [theoraenc] separate encode and push block in theora_enc_chain
+      * 651855 : elements/volume unit test fails
+      * 652342 : encoding-target: set name on audio and video profiles when reading from keyfile
+      * 652642 : typefind: NULL check in degas_type_find
+      * 652838 : gst_discoverer_discover_uri Allow NULL GError* argument
+      * 653461 : [theoraenc] element causes encoder to drop frames?
+      * 654295 : [typefind] audio/x-sap detection doesn't work
+      * 654434 : [basertppayload] RTP timestamps not longer reproducible
+      * 654959 : textoverlay would flicker if it receives not timestampes text on input
+      * 655244 : encodebin has to provide the downstream possible caps to h264parse
+      * 655268 : decodebin2: deadlock after multi-stream chains change
+      * 655279 : [playbin2] Don't reset sinks when not needed
+      * 655347 : theoradec: segfault on 0-byte ogg_packet in _chain_reverse
+      * 655503 : pbutils: Add MPEG-4 SP levels 4a, 5 and 6
+      * 655574 : ogg: crash determining duration of empty vorbis packet
+      * 656022 : volume: fix sample depth typo
+      * 656034 : gstvorbistag: map ENCODER Vorbis comment to application-name
+      * 656392 : audioresample: add FFT based checks
+      * 656715 : playbin2, playsink: reference count ts_offset to avoid crashes
+      * 656775 : oggmux: various cleanups
+      * 656781 : resample.c has warnings treated as errors that prevent compilation
+      * 657049 : textoverlay: buffer leaks
+      * 657062 : oggdemux: do not skip sparse streams when determining start times
+      * 657151 : ogg: another cleanup round
+      * 657257 : discoverer: retrieve audio track language from tags too
+      * 657261 : resindvd: regression in git: no more button highlights in menus
+      * 657319 : videorate should use basetransform
+      * 657333 : theoraenc: fix caps leak
+      * 657504 : gtk-doc distcheck failure: files left in build directory after distclean:
+      * 657872 : [subparse] Doesn't detect some SRT subtitle files
+      * 658294 : gst-inspect videorate hangs
+      * 658416 : decodebin2: refcounting bugs causing criticals
+      * 658443 : theoraenc: do not automatically override quality when using target bitrate
+      * 658514 : typefinding: recognize .amf (Asylum Music File) files for modplug
+      * 658609 : Handle subtitles with non raw caps video streams in subtitle overlay
+      * 658846 : Playbin2 pipeline stuck while prerolling if decoder is missing
+      * 658901 : textoverlay: crash when the video sink pad has no parent
+      * 659562 : videorate: gst_mini_object_unref: assertion `GST_IS_MINI_OBJECT (mini_object)' failed
+      * 660150 : baseaudio: compiler warnings if debugging system is disabled
+      * 660170 : alsasrc: broken timestamps lead to alsasrc ! audiorate endless loop
+      * 660301 : playbin2: Fix mingw compiler warnings
+      * 660304 : videotestsrc: Fix mingw compiler warning
+      * 660598 : playbin2: Make sure that elements that are plugged are compatible with the fixed sink
+      * 660604 : textoverlay: add YV12 support
+      * 660816 : dvd menus got broken 
+      * 661105 : audiotestsrc: add missing break
+      * 661106 : tests: actually test what we said we would
+      * 661122 : videotestsrc does not build on Solaris
+      * 661202 : decodebin2: fire drained signal where appropriate
+      * 661738 : Deadlock between threads in gstaudiosink and gstringbuffer
+      * 661897 : oggdemux: do not retry seeking indefinitely 
+      * 661983 : Regression: Reverse playback does not work for vorbis
+      * 662049 : oggdemux/oggmux in push mode cause preroll to wedge
+      * 662108 : Assertion in base audio decoder when decoding vorbis
+      * 662330 : [decodebin2] Should link and add elements to the bin before checking if they can reach READY state
+      * 662475 : oggdemux: Improvements on the push mode seeking algorithm.
+      * 662829 : [textoverlay] - silent property looks not well implemented
+      * 663174 : oggmux: set collectpads2 not to wait on sparse streams
+      * 663312 : decodebin2: Post all source pads in stream-topology messages as " element-srcpad " values
+      * 663390 : theoraenc: fix speed level failure test
+      * 663391 : theoraenc: misc small tweaks
+      * 663465 : baseaudiosink: fix late buffers leaking
+      * 663766 : [0.11] oggmux: split request pad templates into audio/video/subtitle
+      * 663892 : [playbin2] visualisation leads to not-negotiated error
+      * 663893 : playbin2: g_object_set_valist: construct property " use-volume " for object `GstPlaySinkAudioConvert' can't be set after construction
+      * 664818 : Autoplugger sink bin receives strange caps while it gets the correct ones in 0.10.35 and earlier
+      * 665004 : audioresample emits spurious disconts
+      * 665074 : [gstfft] headers are not bracketed
+      * 665080 : API: subtitle overlays for raw and non-raw video buffers
+      * 665120 : playbin2: decoder not selected for audio-sink=autoaudiosink
+      * 666395 : playbin2: set uri to a non-existed file in " about-to-finish " causes a CRITICAL warning
+      * 667210 : videotestsrc/generate_sine_table needs to link against glib
+      * 667306 : discoverer: don't use unportable vararg macro
+      * 667311 : fix various unlikely, but still potential memoryleaks
+      * 667312 : appsrc: implement get_caps
+      * 667313 : rtcpbuffer: prevent overflow of 16bit header length.
+      * 667315 : videotestsrc: keep the calculation fixed-point
+      * 667316 : pango: Changes includes from brackets to quotes for local files
+      * 667917 : alsasink: Rate doesn't match (requested 88200Hz, get 0Hz)
+      * 668097 : [subtitleoverlay] fix state change stall on PAUSED- > READY- > PAUSED (patch)
+      * 669039 : gstrtspconnection: new data may get sent even-though there is a queued message in the GstRTSPWatch
+      * 669164 : oggdemux generates invalid granpos which causes asserts in theoraparse
+      * 669167 : vorbisparse drops certain data buffers on the floor mistakenly thinking they're headers
+      * 669203 : playbin2: totem segfaults in gst_stream_get_other_pad_from_pad()
+      * 646868 : tag: Provide Creative Commons helper functions
+      * 654388 : [tags] API: move id3 parsing from id3demux to tag lib
+      * 311486 : [oggmux] theora bos must come before any audio bos pages
+
+API additions since 0.10.35:
+
+      * gst_audio_decoder_finish_frame() 
+      * gst_audio_decoder_get_audio_info()
+      * gst_audio_decoder_get_byte_time()
+      * gst_audio_decoder_get_delay()
+      * gst_audio_decoder_get_drainable()
+      * gst_audio_decoder_get_latency()
+      * gst_audio_decoder_get_max_errors() 
+      * gst_audio_decoder_get_min_latency()
+      * gst_audio_decoder_get_needs_format()
+      * gst_audio_decoder_get_parse_state()
+      * gst_audio_decoder_get_plc()
+      * gst_audio_decoder_get_plc_aware()
+      * gst_audio_decoder_get_tolerance()
+      * gst_audio_decoder_get_type()
+      * gst_audio_decoder_set_byte_time()
+      * gst_audio_decoder_set_drainable()
+      * gst_audio_decoder_set_latency()
+      * gst_audio_decoder_set_max_errors()
+      * gst_audio_decoder_set_min_latency()
+      * gst_audio_decoder_set_needs_format()
+      * gst_audio_decoder_set_plc()
+      * gst_audio_decoder_set_plc_aware()
+      * gst_audio_decoder_set_tolerance()
+      * gst_audio_encoder_finish_frame()
+      * gst_audio_encoder_get_audio_info()
+      * gst_audio_encoder_get_drainable()
+      * gst_audio_encoder_get_frame_max()
+      * gst_audio_encoder_get_frame_samples_max()
+      * gst_audio_encoder_get_frame_samples_min()
+      * gst_audio_encoder_get_hard_min()
+      * gst_audio_encoder_get_hard_resync()
+      * gst_audio_encoder_get_latency()
+      * gst_audio_encoder_get_lookahead() 
+      * gst_audio_encoder_get_mark_granule()
+      * gst_audio_encoder_get_perfect_timestamp()
+      * gst_audio_encoder_get_tolerance()
+      * gst_audio_encoder_get_type()
+      * gst_audio_encoder_merge_tags()
+      * gst_audio_encoder_proxy_getcaps()
+      * gst_audio_encoder_set_drainable()
+      * gst_audio_encoder_set_frame_max()
+      * gst_audio_encoder_set_frame_samples_max()
+      * gst_audio_encoder_set_frame_samples_min()
+      * gst_audio_encoder_set_hard_min()
+      * gst_audio_encoder_set_hard_resync()
+      * gst_audio_encoder_set_latency()
+      * gst_audio_encoder_set_lookahead()
+      * gst_audio_encoder_set_mark_granule()
+      * gst_audio_encoder_set_perfect_timestamp()
+      * gst_audio_encoder_set_tolerance()
+      * gst_audio_iec61937_frame_size()
+      * gst_audio_iec61937_payload()
+      * gst_audio_info_clear()
+      * gst_audio_info_convert()
+      * gst_audio_info_copy()
+      * gst_audio_info_free()
+      * gst_audio_info_from_caps()
+      * gst_audio_info_init()
+      * gst_audio_info_to_caps()
+      * gst_base_audio_sink_get_alignment_threshold()
+      * gst_base_audio_sink_get_discont_wait()
+      * gst_base_audio_sink_set_alignment_threshold()
+      * gst_base_audio_sink_set_discont_wait()
+      * gst_codec_utils_h264_get_level_idc()
+      * gst_discoverer_audio_info_get_language()
+      * gst_discoverer_info_get_subtitle_streams()
+      * gst_discoverer_subtitle_info_get_language()
+      * gst_discoverer_subtitle_info_get_type()
+      * gst_encoding_profile_from_discoverer()
+      * gst_tag_get_license_description()
+      * gst_tag_get_license_flags()
+      * gst_tag_get_license_jurisdiction()
+      * gst_tag_get_license_nick()
+      * gst_tag_get_license_title()
+      * gst_tag_get_license_version()
+      * gst_tag_get_licenses()
+      * gst_tag_license_flags_get_type()
+      * gst_tag_get_id3v2_tag_size()
+      * gst_tag_list_from_id3v2_tag()
+      * gst_tag_mux_get_type()
+      * gst_video_buffer_get_overlay_composition()
+      * gst_video_buffer_set_overlay_composition()
+      * gst_video_event_is_force_key_unit()
+      * gst_video_event_new_downstream_force_key_unit()
+      * gst_video_event_new_upstream_force_key_unit()
+      * gst_video_event_parse_downstream_force_key_unit()
+      * gst_video_event_parse_upstream_force_key_unit()
+      * gst_video_get_size_from_caps()
+      * gst_video_overlay_composition_add_rectangle()
+      * gst_video_overlay_composition_blend()
+      * gst_video_overlay_composition_copy()
+      * gst_video_overlay_composition_get_rectangle()
+      * gst_video_overlay_composition_get_seqnum()
+      * gst_video_overlay_composition_get_type()
+      * gst_video_overlay_composition_make_writable()
+      * gst_video_overlay_composition_n_rectangles()
+      * gst_video_overlay_composition_new()
+      * gst_video_overlay_rectangle_copy()
+      * gst_video_overlay_rectangle_get_pixels_argb()
+      * gst_video_overlay_rectangle_get_pixels_unscaled_argb()
+      * gst_video_overlay_rectangle_get_render_rectangle()
+      * gst_video_overlay_rectangle_get_seqnum()
+      * gst_video_overlay_rectangle_get_type()
+      * gst_video_overlay_rectangle_new_argb()
+      * gst_video_overlay_rectangle_set_render_rectangle()
 
 Changes since 0.10.34:
 
diff --git a/RELEASE b/RELEASE
index 7c4682f21..8d94c90f9 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,5 +1,5 @@
 
-Release notes for GStreamer Base Plug-ins 0.10.35 "Short Notice"
+Release notes for GStreamer Base Plug-ins 0.10.36 "Better"
         
 
 
@@ -55,13 +55,323 @@ contains a set of less supported plug-ins that haven't passed the
 
 Features of this release
     
-      * work around GLib atomic ops API change
-      * don't use G_CONST_RETURN in public headers
-      * subparse: typefinding fixes for subtitles in non-UTF8 charsets
+      * audio: new IEC 61937 payloading library
+      * audio: new GstAudioFormat, GstAudioFormatInfo and GstAudioInfo API
+      * audio: new GstAudioDecoder and GstAudioEncoder base classes
+      * audio: baseaudiosink: allow subclasses to provide payloaders
+      * audio: baseaudiosink: fix latency calculation for live elements
+      * audio: baseaudiosink: make discont-wait configurable
+      * audio: baseaudiosink: split "drift-tolerance" into "alignment-threshold"
+      * codec-utils: Add method to convert H.264 text level in a level_idc
+      * discoverer: add support for subtitles; try harder to extract language and duration
+      * encoding-profile: add function to create a profile from a discoverer info
+      * ringbuffer: add support for AAC, DTS, E-AC3 and MPEG audio buffers
+      * rtcpbuffer: Add feedback message types from RFC 510
+      * rtcpbuffer: prevent overflow of 16bit header length
+      * rtspconnection: make hostname lookup thread-safe; OSX portability fixes
+      * rtspconnection: only send new data immediately if there are no queued messages
+      * tags: add new GstTagMux base class
+      * tags: add convenience API to handle creative commons licenses
+      * tags: add API to parse ID3v2 tags
+      * tags: various exif and xmp tag writing fixes
+      * tags: xmp: add Iptc4xmpExt schema support
+      * tags: gstvorbistag: map ENCODER Vorbis comment to application-name
+      * video: add video overlay composition API for subtitles
+      * video: fix a RGB ordering mixup in colorspace conversion code
+      * alsasink: fix high sample rates being rejected, and negotiation to "nearest" rate
+      * audioresample: don't emit DISCONT buffers if no discontinuity happened
+      * audioresample: fix quality setting being ignored; use SSE/SSE2 when possible
+      * audiotestsrc: add red (brownian) and blue/violet noise generator
+      * cdparanoiasrc: fix build issue on OSX (caused by broken cdparanoia port and broken system headers)
+      * decodebin2: improve handling of multi-stream chains (e.g. mpeg-ts)
+      * decodebin2: Cache the upstream seekability for demuxer decode chains and use it for the non-preroll multiqueue limits
+      * decodebin2: add support for autoplugging parsers and parser-converters, and negotiate stream-format conversions properly as needed
+      * decodebin2: link elements before testing if they can reach the READY state, so allow hw-accelerated elements can query the video context
+      * decodebin2: use a TIME limit for pre-rolling in live streams and not in non-live streams
+      * decodebin2: fix preroll for HLS streams at low bitrates
+      * decodebin2: add source pads to stream-topology element messages
+      * decodebin, decodebin2: don't plug the same parser multiple times in a row, so we can make parsers accept parsed input as well (and use them to convert to different stream formats)
+      * encodebin: add flags to disable conversion elements
+      * encodebin: autoplug formatters; re-enable parsers
+      * gnomevfssrc: add support for cancelling read operations
+      * oggdemux, oggmux: add support for new Opus audio codec
+      * oggdemux: implement push mode seeking (e.g. for http)
+      * oggdemux: assume input is live stream if byte size cannot be determined
+      * oggdemux: fix hang on small truncated files
+      * oggmux: add skeleton write support
+      * oggmux: sync  input streams and select input buffers based on running time
+      * oggmux: headers should always have granpos 0
+      * oggmux: refactor how EOS is determined
+      * oggmux: support sparse streams as input (e.g. kate subtitle streams)
+      * playbin2: fix decoder-sink compatibility check for raw audio/video formats
+      * playbin2: make sure that the decoders we plug are compatible with the fixed sink
+      * playsink: Add audio and video converter convenience bins
+      * playbin2: improve stream switching
+      * playbin2/playsink: Decide if A/V caps are raw only inside playsink
+      * playbin2/playsink: better support for raw + compressed streams (audio passthrough)
+      * playbin2/playsink: improve handling of "non-raw" formats (for hw-accelerated video decoding)
+      * playsinkconvertbin: Reconfigure if we switch from raw to incompatible raw caps
+      * subparse: subtitle format typefinding improvements
+      * subtitleoverlay: handle non raw video streams (add suport for hardware accelerated videos)
+      * textoverlay: support more video formats
+      * textoverlay: add "outline-color" and "shadow" properties
+      * textoverlay: attach GstVideoOverlayComposition to buffers if input is not raw video
+      * theoraenc: do not automatically override quality when using target bitrate
+      * theoraenc: proxy downstream caps restrictions upstream 
+      * typefinding: extract SOF marker in jpeg typefinder (to distinguish lossless JPEG)
+      * typefinding: add typefinder for WAP WBMP bitmaps (mostly to avoid false positives)
+      * typefinding: typefind UTF-16 and UTF-32 with BOMs (to avoid false positives)
+      * typefinding: recognize Asylum modules
+      * videorate: add a "max-rate" property; optionally ensure maximum average output frame rate
+      * videorate: add "force-fps" property to force an output framerate or change it on the fly
+      * videorate: optionally only drop frames to ensure maximum frame rate
+      * videoscale: add modified Lanczos scaling method
+      * volume: Fix handling of volume>=4.0 for 8 and 16 bit integer formats
+      * vorbisenc: relax overly tight jitter tolerances (make it work better with non-perfect input streams)
+      * xvimagesink, ximagesink: fall back to non-XShm mode if allocating the XShm image failed
 
 Bugs fixed in this release
      
-      * 600043 : subparse: fails to recognise Cyrillic subtitles in windows-1251 encoding
+      * 643202 : [encodebin] streamcombiner not completely implemented
+      * 654270 : oggmux unit test fails after latest changes
+      * 658984 : Fix typos in gst-plugins-base
+      * 555437 : [tag] add GstTagMux base class
+      * 556648 : [typefind] detect lossless jpeg
+      * 563251 : oggmux should have option to create Ogg Skeleton stream
+      * 584811 : playbin2's get-text-tags sometimes fails in text stream 0
+      * 607619 : [typefind] utf-16 text file mistakenly identified as layer 1 mpeg audio
+      * 607742 : API: add gst_event_new_{upstream,downstream}_force_key_unit() etc.
+      * 609918 : [OS X] configure: cdda_interface.h: present but cannot be compiled (if VERSION is defined)
+      * 610443 : baseaudiosink: clock can jump on setcaps
+      * 612443 : oggdemux: only use information from skeleton if we have nothing better
+      * 615131 : playing an ogg over http does not report duration correctly
+      * 615342 : [gstalsamixer] leaks
+      * 621897 : [oggdemux] reports wrong duration, and push mode seeking support
+      * 628337 : [gnomevfssrc] Add support for cancelling read operations
+      * 628764 : [videorate] add new option for max frame rate
+      * 629212 : [oggdemux] Improve support for push mode (seeking, duration)
+      * 630322 : make seek example work with windows
+      * 630442 : xvimagesink, ximagesink: fallback to X*CreateImage() if X*ShmCreateImage() fails
+      * 630497 : [seek] sprinkle GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS to dump pipeline to dot file
+      * 632788 : [playbin2] Doesn't support files with a streams that are supported compressed by a sink and streams that need decoding
+      * 635556 : [oggdemux] bad duration estimate in streaming mode with vertical-overview.ogg
+      * 637812 : vorbisenc: choppy sound due to input timestamp jitter
+      * 638897 : [textrender] allow setting the canvas size using peer caps + bugfixes
+      * 639055 : discoverer: add support for subtitle streams
+      * 640041 : textoverlay: Added parameters to control text outline color and whether shadowing is enabled
+      * 640564 : Remuxing a Theora stream generates a stream that oggz-validate complains about
+      * 640859 : basesink incorrectly categorizes timestamp jitter as drift
+      * 642690 : [baseaudio] GstBaseAudioEncoder and GstBaseAudioDecoder class
+      * 642878 : encoding-profile: add a function to create a profile from a discoverer info
+      * 643578 : [encodebin] - broken remuxing
+      * 644284 : Suspicious max_latency computation in gstbaseaudiosink.c
+      * 647648 : videorate: support for caps modifications in a running pipeline
+      * 647769 : [decodebin2] Fix preroll for streams at low bitrate
+      * 649319 : Add boiler plate code to xvimagesink
+      * 649642 : [volume] Overflows with volume > =4.0 and 8/16 bit integer formats
+      * 649969 : [audiotestsrc] Add more noise variants
+      * 650406 : vorbisdec does not handle headers in caps
+      * 651089 : [xvimagesink/ximagesink] Remove g_assert from Interface query
+      * 651294 : WBMP images are not supported by typefind
+      * 651496 : encodebin seems to fail to pick up container variant
+      * 651615 : [vorbisenc] Too small jitter tolerance
+      * 651788 : [theoraenc] separate encode and push block in theora_enc_chain
+      * 651855 : elements/volume unit test fails
+      * 652342 : encoding-target: set name on audio and video profiles when reading from keyfile
+      * 652642 : typefind: NULL check in degas_type_find
+      * 652838 : gst_discoverer_discover_uri Allow NULL GError* argument
+      * 653461 : [theoraenc] element causes encoder to drop frames?
+      * 654295 : [typefind] audio/x-sap detection doesn't work
+      * 654434 : [basertppayload] RTP timestamps not longer reproducible
+      * 654959 : textoverlay would flicker if it receives not timestampes text on input
+      * 655244 : encodebin has to provide the downstream possible caps to h264parse
+      * 655268 : decodebin2: deadlock after multi-stream chains change
+      * 655279 : [playbin2] Don't reset sinks when not needed
+      * 655347 : theoradec: segfault on 0-byte ogg_packet in _chain_reverse
+      * 655503 : pbutils: Add MPEG-4 SP levels 4a, 5 and 6
+      * 655574 : ogg: crash determining duration of empty vorbis packet
+      * 656022 : volume: fix sample depth typo
+      * 656034 : gstvorbistag: map ENCODER Vorbis comment to application-name
+      * 656392 : audioresample: add FFT based checks
+      * 656715 : playbin2, playsink: reference count ts_offset to avoid crashes
+      * 656775 : oggmux: various cleanups
+      * 656781 : resample.c has warnings treated as errors that prevent compilation
+      * 657049 : textoverlay: buffer leaks
+      * 657062 : oggdemux: do not skip sparse streams when determining start times
+      * 657151 : ogg: another cleanup round
+      * 657257 : discoverer: retrieve audio track language from tags too
+      * 657261 : resindvd: regression in git: no more button highlights in menus
+      * 657319 : videorate should use basetransform
+      * 657333 : theoraenc: fix caps leak
+      * 657504 : gtk-doc distcheck failure: files left in build directory after distclean:
+      * 657872 : [subparse] Doesn't detect some SRT subtitle files
+      * 658294 : gst-inspect videorate hangs
+      * 658416 : decodebin2: refcounting bugs causing criticals
+      * 658443 : theoraenc: do not automatically override quality when using target bitrate
+      * 658514 : typefinding: recognize .amf (Asylum Music File) files for modplug
+      * 658609 : Handle subtitles with non raw caps video streams in subtitle overlay
+      * 658846 : Playbin2 pipeline stuck while prerolling if decoder is missing
+      * 658901 : textoverlay: crash when the video sink pad has no parent
+      * 659562 : videorate: gst_mini_object_unref: assertion `GST_IS_MINI_OBJECT (mini_object)' failed
+      * 660150 : baseaudio: compiler warnings if debugging system is disabled
+      * 660170 : alsasrc: broken timestamps lead to alsasrc ! audiorate endless loop
+      * 660301 : playbin2: Fix mingw compiler warnings
+      * 660304 : videotestsrc: Fix mingw compiler warning
+      * 660598 : playbin2: Make sure that elements that are plugged are compatible with the fixed sink
+      * 660604 : textoverlay: add YV12 support
+      * 660816 : dvd menus got broken
+      * 661105 : audiotestsrc: add missing break
+      * 661106 : tests: actually test what we said we would
+      * 661122 : videotestsrc does not build on Solaris
+      * 661202 : decodebin2: fire drained signal where appropriate
+      * 661738 : Deadlock between threads in gstaudiosink and gstringbuffer
+      * 661897 : oggdemux: do not retry seeking indefinitely
+      * 661983 : Regression: Reverse playback does not work for vorbis
+      * 662049 : oggdemux/oggmux in push mode cause preroll to wedge
+      * 662108 : Assertion in base audio decoder when decoding vorbis
+      * 662330 : [decodebin2] Should link and add elements to the bin before checking if they can reach READY state
+      * 662475 : oggdemux: Improvements on the push mode seeking algorithm.
+      * 662829 : [textoverlay] - silent property looks not well implemented
+      * 663174 : oggmux: set collectpads2 not to wait on sparse streams
+      * 663312 : decodebin2: Post all source pads in stream-topology messages as " element-srcpad " values
+      * 663390 : theoraenc: fix speed level failure test
+      * 663391 : theoraenc: misc small tweaks
+      * 663465 : baseaudiosink: fix late buffers leaking
+      * 663766 : [0.11] oggmux: split request pad templates into audio/video/subtitle
+      * 663892 : [playbin2] visualisation leads to not-negotiated error
+      * 663893 : playbin2: g_object_set_valist: construct property " use-volume " for object `GstPlaySinkAudioConvert' can't be set after construction
+      * 664818 : Autoplugger sink bin receives strange caps while it gets the correct ones in 0.10.35 and earlier
+      * 665004 : audioresample emits spurious disconts
+      * 665074 : [gstfft] headers are not bracketed
+      * 665080 : API: subtitle overlays for raw and non-raw video buffers
+      * 665120 : playbin2: decoder not selected for audio-sink=autoaudiosink
+      * 666395 : playbin2: set uri to a non-existed file in " about-to-finish " causes a CRITICAL warning
+      * 667210 : videotestsrc/generate_sine_table needs to link against glib
+      * 667306 : discoverer: don't use unportable vararg macro
+      * 667311 : fix various unlikely, but still potential memoryleaks
+      * 667312 : appsrc: implement get_caps
+      * 667313 : rtcpbuffer: prevent overflow of 16bit header length.
+      * 667315 : videotestsrc: keep the calculation fixed-point
+      * 667316 : pango: Changes includes from brackets to quotes for local files
+      * 667917 : alsasink: Rate doesn't match (requested 88200Hz, get 0Hz)
+      * 668097 : [subtitleoverlay] fix state change stall on PAUSED- > READY- > PAUSED (patch)
+      * 669039 : gstrtspconnection: new data may get sent even-though there is a queued message in the GstRTSPWatch
+      * 669164 : oggdemux generates invalid granpos which causes asserts in theoraparse
+      * 669167 : vorbisparse drops certain data buffers on the floor mistakenly thinking they're headers
+      * 669203 : playbin2: totem segfaults in gst_stream_get_other_pad_from_pad()
+      * 646868 : tag: Provide Creative Commons helper functions
+      * 654388 : [tags] API: move id3 parsing from id3demux to tag lib
+      * 311486 : [oggmux] theora bos must come before any audio bos pages
+
+API changed in this release
+     
+- API additions:
+    
+      * gst_audio_decoder_finish_frame()
+      * gst_audio_decoder_get_audio_info()
+      * gst_audio_decoder_get_byte_time()
+      * gst_audio_decoder_get_delay()
+      * gst_audio_decoder_get_drainable()
+      * gst_audio_decoder_get_latency()
+      * gst_audio_decoder_get_max_errors()
+      * gst_audio_decoder_get_min_latency()
+      * gst_audio_decoder_get_needs_format()
+      * gst_audio_decoder_get_parse_state()
+      * gst_audio_decoder_get_plc()
+      * gst_audio_decoder_get_plc_aware()
+      * gst_audio_decoder_get_tolerance()
+      * gst_audio_decoder_get_type()
+      * gst_audio_decoder_set_byte_time()
+      * gst_audio_decoder_set_drainable()
+      * gst_audio_decoder_set_latency()
+      * gst_audio_decoder_set_max_errors()
+      * gst_audio_decoder_set_min_latency()
+      * gst_audio_decoder_set_needs_format()
+      * gst_audio_decoder_set_plc()
+      * gst_audio_decoder_set_plc_aware()
+      * gst_audio_decoder_set_tolerance()
+      * gst_audio_encoder_finish_frame()
+      * gst_audio_encoder_get_audio_info()
+      * gst_audio_encoder_get_drainable()
+      * gst_audio_encoder_get_frame_max()
+      * gst_audio_encoder_get_frame_samples_max()
+      * gst_audio_encoder_get_frame_samples_min()
+      * gst_audio_encoder_get_hard_min()
+      * gst_audio_encoder_get_hard_resync()
+      * gst_audio_encoder_get_latency()
+      * gst_audio_encoder_get_lookahead()
+      * gst_audio_encoder_get_mark_granule()
+      * gst_audio_encoder_get_perfect_timestamp()
+      * gst_audio_encoder_get_tolerance()
+      * gst_audio_encoder_get_type()
+      * gst_audio_encoder_merge_tags()
+      * gst_audio_encoder_proxy_getcaps()
+      * gst_audio_encoder_set_drainable()
+      * gst_audio_encoder_set_frame_max()
+      * gst_audio_encoder_set_frame_samples_max()
+      * gst_audio_encoder_set_frame_samples_min()
+      * gst_audio_encoder_set_hard_min()
+      * gst_audio_encoder_set_hard_resync()
+      * gst_audio_encoder_set_latency()
+      * gst_audio_encoder_set_lookahead()
+      * gst_audio_encoder_set_mark_granule()
+      * gst_audio_encoder_set_perfect_timestamp()
+      * gst_audio_encoder_set_tolerance()
+      * gst_audio_iec61937_frame_size()
+      * gst_audio_iec61937_payload()
+      * gst_audio_info_clear()
+      * gst_audio_info_convert()
+      * gst_audio_info_copy()
+      * gst_audio_info_free()
+      * gst_audio_info_from_caps()
+      * gst_audio_info_init()
+      * gst_audio_info_to_caps()
+      * gst_base_audio_sink_get_alignment_threshold()
+      * gst_base_audio_sink_get_discont_wait()
+      * gst_base_audio_sink_set_alignment_threshold()
+      * gst_base_audio_sink_set_discont_wait()
+      * gst_codec_utils_h264_get_level_idc()
+      * gst_discoverer_audio_info_get_language()
+      * gst_discoverer_info_get_subtitle_streams()
+      * gst_discoverer_subtitle_info_get_language()
+      * gst_discoverer_subtitle_info_get_type()
+      * gst_encoding_profile_from_discoverer()
+      * gst_tag_get_license_description()
+      * gst_tag_get_license_flags()
+      * gst_tag_get_license_jurisdiction()
+      * gst_tag_get_license_nick()
+      * gst_tag_get_license_title()
+      * gst_tag_get_license_version()
+      * gst_tag_get_licenses()
+      * gst_tag_license_flags_get_type()
+      * gst_tag_get_id3v2_tag_size()
+      * gst_tag_list_from_id3v2_tag()
+      * gst_tag_mux_get_type()
+      * gst_video_buffer_get_overlay_composition()
+      * gst_video_buffer_set_overlay_composition()
+      * gst_video_event_is_force_key_unit()
+      * gst_video_event_new_downstream_force_key_unit()
+      * gst_video_event_new_upstream_force_key_unit()
+      * gst_video_event_parse_downstream_force_key_unit()
+      * gst_video_event_parse_upstream_force_key_unit()
+      * gst_video_get_size_from_caps()
+      * gst_video_overlay_composition_add_rectangle()
+      * gst_video_overlay_composition_blend()
+      * gst_video_overlay_composition_copy()
+      * gst_video_overlay_composition_get_rectangle()
+      * gst_video_overlay_composition_get_seqnum()
+      * gst_video_overlay_composition_get_type()
+      * gst_video_overlay_composition_make_writable()
+      * gst_video_overlay_composition_n_rectangles()
+      * gst_video_overlay_composition_new()
+      * gst_video_overlay_rectangle_copy()
+      * gst_video_overlay_rectangle_get_pixels_argb()
+      * gst_video_overlay_rectangle_get_pixels_unscaled_argb()
+      * gst_video_overlay_rectangle_get_render_rectangle()
+      * gst_video_overlay_rectangle_get_seqnum()
+      * gst_video_overlay_rectangle_get_type()
+      * gst_video_overlay_rectangle_new_argb()
+      * gst_video_overlay_rectangle_set_render_rectangle()
 
 Download
 
@@ -90,10 +400,65 @@ Applications
   
 Contributors to this release
     
+      * Age Bosma
+      * Alessandro Decina
+      * Alex Lancaster
+      * Alexey Fisher
+      * Andoni Morales Alastruey
+      * Anssi Hannula
+      * Benjamin Otte
+      * Brian Cameron
+      * Christian Fredrik Kalager Schaller
+      * Christophe Fergeau
+      * Colin Walters
       * David Schleef
-      * Debarshi Ray
+      * Edward Hervey
+      * Erich Schubert
+      * Felipe Contreras
+      * Havard Graff
+      * Iago Toral
+      * Idar Tollefsen
+      * James "Doc" Livingston
+      * James Doc Livingston
+      * Jan Schmidt
+      * Jason Kivlighn
+      * Jens Georg
+      * Jonathan Liu
+      * Jonathan Matthew
+      * Josep Torra
+      * Julien Moutte
+      * Kipp Cannon
+      * LoneStar
+      * Luis de Bethencourt
       * Mark Nauwelaerts
+      * Mart Raudsepp
+      * Mersad Jelacic
+      * Michael Smith
+      * Monty Montgomery
+      * Nicolas Dufresne
+      * Ognyan Tonchev
+      * Olivier Crête
+      * Pascal Buhler
+      * Philip Jägenstedt
+      * Philippe Normand
+      * Piotr Fusik
+      * Raimo Järvi
+      * René Stadler
+      * Reynaldo H. Verdejo Pinochet
+      * Robert Swain
       * Sebastian Dröge
+      * Sergey Scobich
+      * Sergey Scobich)
+      * Sjoerd Simons
       * Stefan Kost
+      * Stefan Sauer
+      * Sébastien Moutte
+      * Thiago Santos
+      * Thibault Saunier
+      * Thomas Vander Stichele
       * Tim-Philipp Müller
+      * Tommi Myöhänen
+      * Vincent Penquerc'h
+      * Wim Taymans
+      * Youness Alaoui
  
\ No newline at end of file
diff --git a/configure.ac b/configure.ac
index 02cc4a8b2..1901bcfb2 100644
--- a/configure.ac
+++ b/configure.ac
@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
 dnl initialize autoconf
 dnl releases only do -Wall, git and prerelease does -Werror too
 dnl use a three digit version number for releases, and four for git/prerelease
-AC_INIT(GStreamer Base Plug-ins, 0.10.35.2,
+AC_INIT(GStreamer Base Plug-ins, 0.10.36,
     http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer,
     gst-plugins-base)
 
@@ -60,7 +60,7 @@ AC_LIBTOOL_WIN32_DLL
 AM_PROG_LIBTOOL
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=0.10.35.3
+GST_REQ=0.10.36
 
 dnl *** autotools stuff ****
 
diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap
index 4f0ff3a64..08bd57e04 100644
--- a/gst-plugins-base.doap
+++ b/gst-plugins-base.doap
@@ -36,6 +36,18 @@ A wide range of video and audio decoders, encoders, and filters are included.
 
  <release>
   <Version>
+   <revision>0.10.36</revision>
+   <branch>0.10</branch>
+   <name>Better</name>
+   <created>2012-02-20</created>
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-0.10.36.tar.bz2" />
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-0.10.36.tar.gz" />
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-0.10.36.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>0.10.35</revision>
    <branch>0.10</branch>
    <name>Short Notice</name>
diff --git a/win32/common/_stdint.h b/win32/common/_stdint.h
index 8ccb22f55..34887b199 100644
--- a/win32/common/_stdint.h
+++ b/win32/common/_stdint.h
@@ -1,7 +1,7 @@
 #ifndef _GST_PLUGINS_BASE__STDINT_H
 #define _GST_PLUGINS_BASE__STDINT_H 1
 #ifndef _GENERATED_STDINT_H
-#define _GENERATED_STDINT_H "gst-plugins-base 0.10.35.2"
+#define _GENERATED_STDINT_H "gst-plugins-base 0.10.36"
 /* generated using gnu compiler gcc (Debian 4.6.2-12) 4.6.2 */
 #define _STDINT_HAVE_STDINT_H 1
 #include <stdint.h>
diff --git a/win32/common/config.h b/win32/common/config.h
index acccbb933..be21f96fe 100644
--- a/win32/common/config.h
+++ b/win32/common/config.h
@@ -81,13 +81,13 @@
 #define GST_MAJORMINOR "0.10"
 
 /* package name in plugins */
-#define GST_PACKAGE_NAME "GStreamer Base Plug-ins prerelease"
+#define GST_PACKAGE_NAME "GStreamer Base Plug-ins source release"
 
 /* package origin */
 #define GST_PACKAGE_ORIGIN "Unknown package origin"
 
 /* GStreamer package release date/time for plugins as YYYY-MM-DD */
-#define GST_PACKAGE_RELEASE_DATETIME "2012-02-05T13:47Z"
+#define GST_PACKAGE_RELEASE_DATETIME "2012-02-20"
 
 /* I know the API is subject to change. */
 #undef G_UDEV_API_IS_SUBJECT_TO_CHANGE
@@ -358,7 +358,7 @@
 #define PACKAGE_NAME "GStreamer Base Plug-ins"
 
 /* Define to the full name and version of this package. */
-#define PACKAGE_STRING "GStreamer Base Plug-ins 0.10.35.2"
+#define PACKAGE_STRING "GStreamer Base Plug-ins 0.10.36"
 
 /* Define to the one symbol short name of this package. */
 #define PACKAGE_TARNAME "gst-plugins-base"
@@ -367,7 +367,7 @@
 #undef PACKAGE_URL
 
 /* Define to the version of this package. */
-#define PACKAGE_VERSION "0.10.35.2"
+#define PACKAGE_VERSION "0.10.36"
 
 /* directory where plugins are located */
 #ifdef _DEBUG
@@ -398,7 +398,7 @@
 #undef USE_TREMOLO
 
 /* Version number of package */
-#define VERSION "0.10.35.2"
+#define VERSION "0.10.36"
 
 /* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most
    significant byte first (like Motorola and SPARC, unlike Intel). */