=== release 0.10.18 ===

2010-02-10  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  releasing 0.10.18, "Short Circuit"

2010-02-10 20:36:56 +0000  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: temporary safety check to avoid crashes with a certain file
	  Add temporary check to avoid crashes with a certain file when seeking
	  until the real cause of this is figured out. See #609405.

2010-02-05 18:05:39 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: skip unknown atoms when looking for moov
	  Fixes bug #609107

2010-02-05 02:13:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-02-04 19:10:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/bg.po:
	* po/hu.po:
	  po: update translations

2010-02-04 14:46:56 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: Set the segment start time to the requested seek time for non-keyframe seeks

2010-02-04 12:00:03 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix time returned for index at a byte offset
	  The logic for searching forwards/backwards was swapped

2010-02-01 19:22:24 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: initialize stereo decoding state

2010-01-28 18:58:08 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: improve stream synchronization
	  In particular, do not make it send newsegment updates that
	  sort-of contradict the indented playback segment (e.g. start time).

2010-01-28 18:53:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix bridging (time) gaps in streams
	  As a side effect, avoid sending newsegment updates with start times
	  that go back and forth, which leads to bogus downstream running_time.
	  Also fixes seeking in bug #606744.

2010-01-28 18:49:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix stream synchronization
	  .. by initializing streams starting at 0, as that is basically
	  where we 'seek to' at the start and assume streams to start elsewhere.
	  Also enables newsegment update events for subtitle streams.

2010-02-02 13:41:03 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegdec.c:
	  jpeg: don't directly access message, some message have args
	  This caused bogus messages, such as reported in bug #607471.

2010-02-02 00:02:34 +0000  David Hoyt <dhoyt@llnl.gov>

	* ext/libpng/gstpngdec.c:
	  png: fix compilation with libpng 1.4
	  png_set_gray_1_2_4_to_8() has been deprecated for a while and was
	  finally removed in libpng 1.4.x. Use png_set_expand_gray_1_2_4_to_8()
	  instead.
	  Fixes #608629.

2010-02-01 16:46:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: free transports on errors
	  See #608564

2010-02-01 09:18:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/v4l2_calls.c:
	  v4l2: fix unportable printf format

2010-01-30 15:18:48 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 15d47a6 to 96dc793

2010-01-27 17:53:07 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: index timestamps should be in seconds, not milliseconds

2010-01-27 15:24:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	  speexdec: free some more when resetting
	  Fixes #608255.

2010-01-27 15:24:24 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpspeexpay.c:
	  rtpspeexpay: fix occasional buffer leak
	  Fixes #608255.

2010-01-27 15:22:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: prevent invalid arithmetic if not setup yet
	  Fixes #608255.

2010-01-27 16:34:21 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_mmx.h:
	  videomixer: Fix assembly register constraints
	  Fixes bug #608209.

2010-01-27 01:56:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-01-27 01:52:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/LINGUAS:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ru.po:
	* po/sk.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  po: update translations

2010-01-27 01:49:49 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/.gitignore:
	  checks: ignore deinterlace check binary

2010-01-27 01:18:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: purge all mention of CVS

2010-01-26 11:18:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: ignore streams that finished
	  When we receive an UNEXPECTED from a stream, move to the next stream and only go
	  EOS when all streams are EOS. When selecting a stream to push, ignore streams
	  that went EOS.
	  Fixes #607949

2010-01-25 17:23:43 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: don't deref NULL
	  Error out when the pool gets shutdown.

2010-01-25 17:21:13 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	* sys/v4l2/v4l2src_calls.c:
	* tests/check/Makefile.am:
	  Revert "v4l2src: don't deref NULL"
	  This reverts commit 3d9d34bd60faeb940b36d992a47168fc895036ba.

2010-01-25 14:16:22 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	* sys/v4l2/v4l2src_calls.c:
	* tests/check/Makefile.am:
	  v4l2src: don't deref NULL
	  Error out when the pool gets shutdown.

2010-01-23 15:32:48 -0800  Michael Smith <msmith@xiph.org>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: when creating an overflow buffer, copy timestamps.

2010-01-23 14:47:55 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: dmb1 is a valid fourcc for Motion-JPEG

2010-01-23 14:20:02 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdeux: IV32 is also used for Indeo 3 video streams

2010-01-22 16:48:01 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/icles/ximagesrc-test.c:
	  build: no unused variables when disabling asserts

2010-01-21 23:17:40 -0300  Roland Krikava <rkrikava@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Avoid negative overflow on keyframe search
	  Do not overflow negatively when searching a previous
	  "keyframe" on audio streams. Could cause infinite loops
	  on backwards playback
	  Fixes #607718

2010-01-21 17:22:38 -0800  Peter van Hardenberg <pvh@songbirdnest.com>

	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstjpegenc.h:
	  jpegenc: enlarge buffer if libjpeg tells us it's out of space. Fixes buffer overflow on some high-quality, low-resolution jpeg encodes.

2010-01-21 19:24:22 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix compiler warnings under OS X.

2010-01-21 17:57:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: don't parse NULL indexes
	  for some streams we might fail to fetch the index offsets. Don't try to parse
	  NULL indexes in those cases.

2010-01-18 21:15:51 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: ptime should is in nanoseconds
	  https://bugzilla.gnome.org/show_bug.cgi?id=607403

2010-01-20 15:11:15 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/wavenc/gstwavenc.c:
	* gst/wavenc/gstwavenc.h:
	  wavenc: Post warning if file isnt finished properly
	  When the pipeline is shut down and the file isn't
	  finished properly, wavenc should post a warning.
	  Fixes #607440

2009-05-27 13:51:44 +0200  Arnout Vandecappelle <arnout@mind.be>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: make index size configurable.
	  Added the 'min-index-interval' property to matroskamux,
	  which determines how much time (nanoseconds) is left
	  between keyframes stored in the index.
	  Fixes #583985.

2010-01-20 16:28:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: scale spspps_interval to milliseconds
	  The spspps_interval is kept in seconds. Convert it to milliseconds before
	  comparing it to another value in milliseconds.

2010-01-20 15:18:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: always keep media segments within total duration
	  ... as opposed to only doing so following a seek.

2010-01-20 15:44:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: rename spspps-interval property
	  Rename the spspps-interval property to config-interval because it is nicer.

2010-01-19 18:37:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: skip RIFF and index in push mode
	  When we are in push mode, we can encounter RIFF and idx tags in the data chunk
	  when we are dealing with ODML files. In these cases, simply skip the chunks and
	  continue streaming instead of going EOS.

2010-01-20 11:27:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: more DISCONT handling
	  Add some debug in the DISCONT handling code.
	  When we receive a DISCONT in push mode, mark all streams as DISCONT.

2010-01-20 11:26:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: reset on flush events
	  When we receive a flush event on the sinkpad, reset the EOS state and the
	  flowreturn of all streams. Also mark the streams with a DISCONT.

2010-01-20 11:22:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: rename some variable
	  Rename the seek_event variable to seg_event because it really contains the
	  newsegment event that needs to be pushed.

2010-01-20 00:54:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 14cec89 to 15d47a6

2010-01-18 14:49:26 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Don't set profile-level-id in out caps
	  The profile-level-id represents restrictions on what can be sent, it does not
	  describe the stream. So it should be reflected in the sink caps of the
	  payloader, not the src caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=607353

2010-01-18 14:41:10 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Don't ignore the return value from set_outcaps
	  https://bugzilla.gnome.org/show_bug.cgi?id=607353

2010-01-18 17:43:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/deinterlace/tvtime/greedyhmacros.h:
	* gst/deinterlace/tvtime/linear.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/tomsmocomp.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: Fix license and copyright headers

2010-01-18 14:57:42 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: move G_END_DECLS to the end

2010-01-18 14:55:38 +0200  Stefan Kost <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: fix bufferpool file names in header comment

2010-01-15 18:15:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: avoid some typecasting

2010-01-15 18:13:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: avoid some type checks

2010-01-15 18:09:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: fallback to avih duration
	  when we have not yet parsed the indexes (in push mode, for example) use
	  the duration as given in the avih header instead of -1.

2010-01-15 13:32:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: g_free is NULL safe

2010-01-15 13:27:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use DEMUX errors, instead of DECODE
	  qtdemux should use DEMUX errors, and not DECODE
	  Conflicts:
	  gst/qtdemux/qtdemux.c

2010-01-14 19:16:19 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Minor refactor
	  Replace repeated code with a function call

2010-01-14 17:11:13 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: Handle another kind of redirect trak
	  Some traks might contain a redirect rtsp uri inside
	  hndl atom (which is a dref atom entry). This commit makes qtdemux
	  post a message when it finds one of these traks and there are
	  no other traks.
	  Fixes #597497

2010-01-14 16:13:08 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: Post error when reaching EOS without pads
	  Post an error when EOS is reached and there are no src pads

2010-01-14 14:13:50 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Do not post empty redirect messages
	  Some misinterpreted data could result in posting redirect messages
	  with empty redirect strings. It is better not to post them.
	  An example is the file on bug #597497

2010-01-14 18:19:25 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: polish last buffer end time usage
	  That is, reset it upon seek, and note that (rarely) last pushed buffer
	  time might precede segment start.

2010-01-13 16:48:46 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/videomixer/blend_mmx.h:
	  videomixer: use 'q' constraint instead of 'r'
	  This avoids the "bad register name `%dil'" compilation errors on 32bit where
	  because of 'r' gcc puts the value in a general purpose register and then tries
	  to access the lower part as %dil/%sil which is not existing on 32bit. 'q' requests
	  a-d registers

2010-01-13 16:44:58 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  avi: add missing include for sscanf

2010-01-13 09:36:03 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/equalizer/gstiirequalizer10bands.c:
	  equalizer: Fix property description for the 3rd band of the 10band equalizer
	  The frequency is actually 237 Hz, not 227 Hz.
	  Fixes bug #606692.

2010-01-13 09:22:20 +0100  Kipp Cannon <kcannon@ligo.caltech.edu>

	* gst/audiofx/audioamplify.c:
	  audioamplify: Allow negative amplifications
	  Fixes bug #606807.

2010-01-13 09:17:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/taglib/gstapev2mux.cc:
	  apev2mux: Don't call constructors directly, this leads to compiler errors with gcc 4.5

2010-01-12 17:39:05 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: use G_GSIZE_FORMAT for platform independent gsize qualifier
	  Fixes build on macosx

2010-01-11 19:02:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: refactor eos sending when pausing loop
	  Also, prevent hanging if no pads yet on which to send eos by
	  posting a message instead.

2010-01-11 17:50:35 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: standardize seek handling
	  ... which implies fixing some corner cases.

2010-01-11 15:14:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: use more generic xiphN_streamheader_to_codecdata helper

2010-01-11 17:50:04 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: reflow audio and video setcaps and improve logging
	  Also ensure width and height are available as they are mandatory
	  in matroska specs.

2010-01-11 11:42:43 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix offset for type 2 mp4a sound sample descriptions.
	  Allows us to correctly find the esds (and thus the codec data) for such
	  mp4a files.

2010-01-11 15:45:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	  rtpmp4g(de)pay: Only handle raw aac
	  rtpmp4g(de)pay should only handle raw AAC streams

2010-01-11 18:59:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Implement basic QoS
	  This drops frames if they're too late anyway before blending and all
	  that starts but QoS events are not forwarded upstream. In the future
	  the QoS events should be transformed somehow and forwarded upstream.

2010-01-11 14:48:26 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	  rtpmp4a(de)pay: Only accept raw aac
	  rtpmp4a(de)pay should only handle raw aac to conform to the RFC

2010-01-11 18:35:47 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend.c:
	* gst/videomixer/blend_mmx.h:
	  videomixer: Add MMX implementations for I420 and all non-alpha RGB formats

2010-01-04 10:24:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend.c:
	* gst/videomixer/blend.h:
	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_i420.c:
	* gst/videomixer/blend_mmx.h:
	* gst/videomixer/blend_rgb.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer.h:
	  videomixer: Refactor processing functions
	  This allows easier plugging of optimized processing functions
	  in the future, like for SSE or AltiVec.

2010-01-11 13:26:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/matroska/matroska-mux.c:
	  avimux: matroskamux: rename aac's stream-format to raw
	  AAC's none stream-format has been renamed to raw, rename
	  on avimux and matroskamux as well

2010-01-11 12:07:29 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Only accept raw aac
	  makes matroskamux reject aac streams that are not
	  in raw format (stream-format=none)
	  Fixes #598350

2010-01-11 12:08:55 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: Only accept raw aac
	  makes avimux reject aac streams that are not
	  in raw format (stream-format=none)
	  Fixes #598350

2010-01-11 10:38:10 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Oops. The gpointer cast is needed because of the const qualifiers on the data elements

2010-01-11 10:17:54 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Debug -> info level for a message for benchmarking index parsing
	  The extra message output at higher levels affects the accuracy of the
	  benchmark.

2010-01-11 10:05:10 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Don't check for NULL pointers or cast to gpointer as this is not needed

2010-01-08 13:55:05 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Refactor stbl sub-atom freeing. Free when index has been completely parsed.

2010-01-08 14:32:06 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Avoid whitespace commits due to inconsistent GNU indent behaviour

2010-01-11 00:10:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: remove newline at end of debug statement

2010-01-08 19:26:21 +0100  Havard Graff <havard.graff@tandberg.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Compiler warning fixes for Windows
	  Just simple missing casts
	  Fixes bug #606438.

2010-01-08 18:04:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: fix seekpoints property copy-and-paste documentation

2010-01-06 17:06:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flacenc: optionally add a seek table
	  API: GstFlacEnc:seekpoints
	  Fixes #351595.

2010-01-08 11:33:02 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Use more glib and be safer
	  Be safer on sscanf by limiting string format sizes.
	  Remove useless parameter and use g_strndup.

2010-01-08 10:44:44 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Simplifying code
	  Greatly simplify the IDIT chunk handling by using sscanf
	  instead of 'manually' parsing. Also replaces strncasecmp and
	  is_alpha/is_digit with glib versions.

2010-01-08 10:18:30 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: it's feb for february
	  Fix typo in last commit.

2010-01-08 09:17:22 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: Parse and post IDIT dates
	  Parses and post date tags contained in IDIT chunks.
	  Fixes #503582

2010-01-07 17:25:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Add property for not draining the history on kernel changes
	  Currently this only works if the kernel size doesn't change, in the future
	  it will be possible to change the kernel size too without draining
	  the complete history and without loosing anything.
	  Partially based on a patch by
	  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

2010-01-07 16:58:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: remove weird memcmp code
	  Use plain memcmp for comparing memory instead of the custom buggy one.
	  Fixes #606198

2010-01-07 15:38:36 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/level/gstlevel.c:
	  level: fix typo in 'message' property description

2010-01-06 14:06:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: really use upstream timestamp if there is one
	  See/fixes #603471.

2010-01-06 13:45:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg728pay: remove unused adapter peek

2010-01-05 19:00:35 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Improve passthrough tests
	  Improve passthrough tests by forcing more specific
	  interlaced/deinterlaced caps to be tested

2010-01-05 18:22:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Adds some docs to the new tests
	  Adds some docs explaining the utility functions of the check
	  tests of deinterlace

2010-01-05 18:14:08 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/deinterlace.c:
	  deinterlace: Adds tests for passthrough
	  Adds tests for checking if the element really does
	  passthrough in disabled mode and in auto (if the input is
	  not interlaced)

2010-01-05 07:50:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/deinterlace.c:
	  deinterlace: Adds tests for caps acceptance
	  Adds check unit tests for deinterlace for validating
	  caps accepting and the expected caps output on the
	  other pad

2010-01-04 13:43:00 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/deinterlace.c:
	  deinterlace: Adds basic check test
	  Adds a basic check test for deinterlace element

2010-01-04 15:44:28 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/qtdemux/Makefile.am:
	* gst/qtdemux/qtdemux.c:
	  qtdemux: Add support for wave-style audio in qt.
	  Uses gstriff to parse the wave headers appropriately. Tested with MS-ADPCM
	  content.

2009-12-31 17:09:03 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* tests/check/elements/rtp-payloading.c:
	  tests: Add G.729 RTP payloader/depayloader test
	  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-31 16:52:30 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Simplify adapter usage
	  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-31 16:27:30 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Support ptime from caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-02 19:35:21 +0530  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/README:
	  rtp: Add maxptime to the README
	  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2010-01-05 19:03:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723depay.h:
	  rtpg723depay: add G723 depayloader

2010-01-05 19:02:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729depay.h:
	  rtpg729depay: remove unused variable

2010-01-05 18:33:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg723pay.h:
	  rtpg723pay: rewrite payloader
	  Handle all 3 packet sizes according to RFC 3551.
	  Totally untested, we don't have a G723 encoder.
	  Fixes #605882

2010-01-05 11:47:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix chunk counter

2010-01-04 19:44:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: more work at reducing loop overhead
	  Try to avoid derefs when parsing the index. Save the state into the structures
	  when we exit the loop instead of for each iteration.

2010-01-04 16:33:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: cleanups and make duration more accurate
	  Make the QtDemuxSample struct smaller by keeping the duration and the pts_offset
	  as their 32 bit values.
	  Make some macros to calculate PTS, DTS and duration of a sample.
	  Deref the sample index less often by keeping a ref to the sample we're dealing
	  with.

2010-01-04 13:41:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: simplify logic to calculate duration
	  Since we no longer store the timestamp and duration in nanoseconds, we can now
	  simply store the duration as-is.

2010-01-01 16:42:57 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Store timestamps in mov format in the index
	  This allows faster building of the index upon seeks so that scaling of
	  timestamps only occurs when actually needed.

2009-12-18 13:54:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: make seeking in push mode work
	  Move sample position checks into qtdemux_parse_samples where we can protect it
	  with a lock.
	  Refactor and make an qtdemux_ensure_index function.
	  Rename qtdemux_do_push_seek to qtdemux_seek_offset in order to avoid confusion
	  with gst_qtdemux_do_push_seek.

2009-12-18 12:44:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: move error code out of normal flow

2009-11-24 16:27:26 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux.h:
	  qtdemux: Add push mode seek support for seeking to obtain the moov atom

2010-01-05 12:22:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix on-npt-stop signal warnings for RDT
	  The RDT manager does not implement this signal so we need to check for it before
	  trying to connect to it.

2010-01-05 09:47:00 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: fix memory leak in new uri handler code
	  Don't leak a string everytime get_uri() is called and a device
	  has been set. There's a limited number of devices, so just
	  intern the string instead of doing more elaborate housekeeping
	  and storing it in the instance struct or so.

2010-01-01 14:10:49 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	  avimux: fix typo in warning message

2010-01-04 09:28:36 -0300  Robert Weidlich <gnomebugzilla@robert.weidlich.cc>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2send: Add 'public' property
	  Adds a property to set 'public' flag on libshout, making
	  the stream listed on the server's stream directory.
	  Fixes #605269

2009-12-30 14:14:55 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Add tags for average and maximum bitrate
	  Fixes #599300.

2009-12-26 16:59:14 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: do not try to alloc really large buffers
	  When nsamples_out is larger than nsamples_in, using unsigned
	  ints lead to a overflow and the resulting value is wrong and
	  way too large for allocating a buffer. Use signed integers
	  and returning immediatelly when that happens.

2009-12-25 12:38:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	  videomixer: optimize blend code some more
	  Use more efficient formula that uses less multiplies.
	  Reduce the amount of scalar code, use MMX to calculate the desired
	  alpha value.
	  Unroll and handle 2 pixels in one iteration for improved pairing.

2009-12-24 22:59:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_i420.c:
	* gst/videomixer/blend_rgb.c:
	  videomixer: scale and clamp
	  Scale and clamp to the max alpha values.

2009-12-24 22:50:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	  alpha: scale and clamp alpha to its full extend
	  Convert the alpha value to 0->255 when setting and to 0->256 when using as
	  a scaling factor. This makes sure we can reach the full opacity value of 0xff in
	  all cases.

2009-12-24 22:23:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix some comments, remove property check
	  Fix some comments, clarify some FIXMEs
	  Remove the on-ntp-stop signal check now that the jitterbuffer is in
	  -good and we know that it supports this signal.

2009-12-24 20:27:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: some trivial cleanups

2009-12-24 17:04:28 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Parse all rtpinfo entries
	  Do not forget to parse all rtp-info entries, instead of
	  parsing the first one only.
	  Fixes #605222

2009-12-22 12:44:50 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: perf tag should map to GST_TAG_ARTIST

2009-12-24 17:03:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/interleave/interleave.c:
	  interleave: fix weird indentation

2009-12-24 17:01:54 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: use faster _adapter_copy() whem possible

2009-12-24 17:01:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/audiofx/firfilter-example.c:
	  tests: use right type when passing vararg value

2009-12-23 17:50:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacdec.h:
	  flacdec: use a single decoder field for both push and pull mode

2009-12-23 17:03:32 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix possible hanging in pull mode seeking
	  A seek in multi-sink pipeline typically leads to several seek events in a row,
	  which could lead to sending several newsegments in a row without intermediate
	  flushing.  These would then accumulate, distort rendering times and as such
	  lead to 'hanging'.

2009-12-23 19:39:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: fix uninitialized variable

2009-12-23 13:09:54 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: use boilerplate

2009-12-23 00:38:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpL16pay.h:
	  rtpL16pay: convert to baseaudiopayload
	  Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes
	  a bunch of problems that were already solved in the base class.
	  Fixes #853367

2009-12-23 00:30:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtppcmapay.c:
	  rtppcmapay: the boilerplate macro sets parent_class

2009-12-22 22:27:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpbin: avoid some structure copies
	  Don't make copied in the getter and setter for SDES in the RTPSource. This
	  avoids a couple of copies of the SDES structure when generating RTCP
	  packets.

2009-08-31 18:42:25 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpmanager: improve SDES handling
	  Store SDES internally as a struct to support multiple PRIV values.
	  Include all values set in SDES struct when sending RTCP SDES.

2009-12-22 14:41:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: add some fixmes

2009-12-22 14:35:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: baseclass handles timestamps for us

2009-12-22 14:27:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph263depay.c:
	  rtph263depay: reset start variable properly

2009-05-29 15:49:27 +0300  Marco Ballesio <marco.ballesio@nokia.com>

	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263depay.h:
	  Drop the whole frame if a packet is lost.
	  Fixes #582575

2009-12-21 20:39:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: add option to insert PPS/SPS in streams
	  Add a new spspps-interval property to instruct the payloader to insert
	  SPS and PPS at periodic intervals in the stream.
	  Rework the SPS/PPS handling so that bytestream and AVC sample code both use the
	  same code paths to handle sprop-parameter-sets. This also allows to have the AVC
	  code to insert SPS/PPS like the bytestream code.
	  Fixes #604913

2009-12-21 19:12:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 47cb23a to 14cec89

2009-12-21 12:01:53 -0300  Jonathan Conder <j@skurvy.no-ip.org>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	* gst/qtdemux/qtdemux_types.c:
	  qtdemux: Adds new tags
	  Adds some new tags mapping to qtdemux.
	  Fixes #599759

2009-12-21 15:05:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: add property to remove pads automatically
	  Add a property called autoremove to automatically remove the pads of sources
	  that timed out.
	  Fixes #554839

2009-12-21 14:55:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  ssrcdemux: fix comparison
	  A NULL means no pad was found.

2009-11-08 11:49:14 +0100  Edward Hervey <bilboed@bilboed.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Add GstURIHandler interface. Fixes #601143
	  This allows using v4l2://[<device>]

2009-12-20 17:24:47 -0800  Michael Smith <msmith@xiph.org>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: pass length parameter to g_convert

2009-12-18 12:44:50 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/matroska/matroska-demux.c:
	  matroska: Fix unitialized variable.
	  Yes, it's stupid, but macosx compilers are even more stupid.

2009-12-17 16:01:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	  videomixer: Fix assembly compilation on x86
	  Fixes bug #604814.

2009-12-17 17:37:03 +0100  Branko Čibej <brane at xbc.nu>

	* gst/replaygain/rganalysis.c:
	  rganalysis: fix timestamp rounding
	  Use scaling function to round and avoid overflows.
	  Fixes #604352

2009-12-17 17:27:42 +0100  Tiago Katcipis <tiago.katcipis@digitro.com.br>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg723pay.h:
	  rtp: add G723 payloader
	  Fixes #597823

2009-12-17 16:22:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_types.c:
	  qtdemux: Fix ALAC codec_data parsing
	  Fixes #604611

2009-12-16 17:28:30 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Remove cpp style coments
	  Removes // comments and replace them with /* */ comments

2009-12-16 12:48:02 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: also consider BlockNumber indicated in index when seeking

2009-12-16 12:43:27 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-read.h:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: support push based mode
	  Fixes #598610.

2009-12-16 12:44:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/ebml-read.c:
	  matroskademux: fix ebml read cache usage

2009-12-16 10:50:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	  videomixer: Use movzbl instead of movzxb for moving one byte to a l register
	  For some reason latest gcc/binutils accept movzxb here while
	  movzbl would be correct and is the only thing accepted by older
	  gcc/binutils.
	  Fixes bug #604679.

2009-12-16 06:59:01 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	  videomixer: src/dest are input and output of the AYUV blending MMX assembler

2009-12-15 18:18:54 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiowsincband.c:
	  audiowsincband: Use the same upper length limit as audiowsinclimit

2009-12-12 17:00:50 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	  audiowsinc{limit,band}: Allow much larger filter lengths now

2009-12-11 12:27:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Fix frequency response calculation

2009-12-08 14:57:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Remove dead assignments

2009-12-06 16:58:51 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Add special processing functions for Mono/Stereo
	  This provides another 7% speedup for the time domain convolution and 1.5%
	  speedup for the FFT convolution on Mono input.
	  This optimization assumes that the compiler simplifies calculations
	  and conditions on constant numbers and unrolls loops with a constant
	  number of repeats.

2009-12-04 09:25:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Add a "low-latency" mode
	  This will always use time-domain convolution, which lowers the latency.
	  With FFT convolution it's always a multiple of the kernel length,
	  with time domain convolution it's only the pre-latency of the filter kernel.

2009-12-04 09:00:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Remove obsolete TODO comments

2009-12-03 20:12:01 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Use samples everywhere instead of samples*channels sometimes

2009-12-03 17:27:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: FFT convolution implementation
	  This provides a great speedup, especially the relationship between kernel
	  length and processing size is now logarithmic instead of linear. Below a
	  kernel size of 32 it's a bit slower, afterwards it's much faster:
	  17     0.788000 -> 0.950000
	  33     1.208000 -> 1.146000
	  65     2.166000 -> 1.146000
	  ...
	  4097 107.444000 -> 1.508000
	  For sizes smaller 32 the normal time-domain convolution is chosen,
	  for larger sizes the FFT convolution is automatically used.
	  Fixes bug #594381.

2009-11-27 20:33:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Make most code parts independent of the processing functions and used convolution algorithm
	  Only remaining part is the residue pushing, which will be fixed later.

2009-11-26 15:17:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Optimize time-domain convolution
	  Remove some redundant calculations, move comparisions out of
	  inner loops, etc.
	  This makes the convolution about 3 (!) times faster but
	  processing time is of course still proportional to the
	  filter size.

2009-11-26 10:45:37 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	  audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once

2009-11-25 18:12:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Rewrite timestamp tracking
	  It's much simpler now and doesn't introduce accumulating rounding
	  errors.

2009-11-25 17:39:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Rename some variables and change comments

2009-11-24 20:06:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audiofx/audiofxbasefirfilter.c:
	* gst/audiofx/audiofxbasefirfilter.h:
	  audiofxbasefirfilter: Add const qualifier to the source data array

2009-12-14 20:08:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/videomixer.c:
	  videomixer: Add MMX implementations of the AYUV blending and color filling functions
	  This provides a 20% speedup for blending and 100% for color filling.
	  The blending can probably be optimized even more.

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

	* gst/id3demux/id3v2frames.c:
	  id3demux: prefer two letter ISO 639-1 code for extended comment

2009-12-13 13:10:12 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix up language code extraction some more
	  Quicktime uses ISO 639-2 for language codes, but GST_TAG_LANGUAGE
	  is supposed to hold a ISO 639-1 code, so convert as needed using
	  the new API from -base.
	  See #602126.

2009-12-13 12:45:22 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	  matroska: fix language code writing and extraction
	  Matroska uses three-letter ISO 639-2B codes, but GST_TAG_LANGUAGE is
	  supposed to contain two-letter ISO 639-1 codes, so use new language
	  code mapping functions in -base to convert between those two as
	  needed.
	  Fixes #505823.

2009-12-07 20:54:07 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: minor debug message changes
	  Fix up a few debug messages so that it's clearer what they mean.

2009-12-12 17:44:04 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  Revert "qtdemux: Correctly parse classification tags"
	  This reverts commit cd883aa60c1133196a6ae052884d15c295c37dde.
	  Previous code was correct, 4 is due to table and language code,
	  not only language code

2009-12-12 16:28:36 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Correctly parse classification tags
	  In clsf atoms, the language code is 2 bytes long, not 4.

2009-12-12 16:55:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Dequeue current buffer on FLUSH_STOP and don't unref NULL buffers
	  ... NULL buffers shouldn't really happen anymore when popping the
	  buffer from GstCollectPads but better check for this and print a warning.

2009-12-11 13:11:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_i420.c:
	  videomixer: Fix stupid mistake in last commit

2009-12-11 12:35:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_i420.c:
	  videomixer: Don't do floating point math in the inner processing loop for I420 blending

2009-12-10 18:43:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle NULL and empty transport strings
	  When an RTSP extension returns NULL or an empty transport string, just ignore it
	  and try to get the next possible transport. Fixes playback of RealMedia streams.

2009-12-10 18:42:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: install event function on internal RTCP pad
	  Install a custom event function on the internal RTCP pad so that we can reply
	  TRUE to a latency event.

2009-12-10 10:48:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/blend_ayuv.c:
	* gst/videomixer/blend_bgra.c:
	* gst/videomixer/blend_rgb.c:
	  videomixer: Remove wrong comments, copied from the I420 blend function

2009-12-09 21:15:07 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: The queued duration is a signed integer
	  ...and it will really be negative sometimes.

2009-12-09 21:03:57 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Only pop buffers from collectpads after they're fully consumed
	  This decreases latency and memory usage because new buffers are only
	  accepted by collectpads if there's no queued buffer.

2009-12-09 20:42:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: Clean up position/duration handling
	  Also use the last end time for closing the segment, not the
	  start time of the last buffer.

2009-12-09 16:50:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Close the segment on EOS if the real duration is known

2009-12-09 16:46:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Update duration if current buffer is already after the old duration

2009-12-09 16:43:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Drop buffers that are after segment stop
	  ...and if this happened for all streams go EOS.

2009-12-09 16:41:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix position tracking and sending of filler segments

2009-12-09 16:15:09 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer.c:
	  videomixer: Use gst_util_uint64_scale_int() for fps to seconds per frame calculations

2009-12-08 17:34:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Keep the segment stop position for update newsegment events

2009-12-04 14:42:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	* ext/Makefile.am:
	* ext/ladspa/Makefile.am:
	* ext/ladspa/gstladspa.c:
	* ext/ladspa/gstladspa.h:
	* ext/ladspa/gstsignalprocessor.c:
	* ext/ladspa/gstsignalprocessor.h:
	* ext/ladspa/load.c:
	* ext/ladspa/search.c:
	* ext/ladspa/utils.h:
	  ladspa: Remove the sources from gst-plugins-good
	  It's disabled anyway and the latest version of it is in
	  gst-plugins-bad. Fixes bug #603779.

2009-12-04 13:50:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: init current_entry in push mode
	  Set the current_entry to 0 (instead of -1) in push mode so that we correctly
	  calculate the current frame number and timestamp.
	  Add some more debug info and fic the duration debug.

2009-12-04 11:14:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix major memory leak when playing back rtsp video streams
	  Don't forget to unref QoS, navigation and latency events when
	  dropping them.

2009-12-03 08:58:08 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: only send pending tags with newsegment events
	  Send pending tags only from the streaming thread, just after we've sent
	  the newsegment event, not with e.g. flush-start. This not only does the
	  right thing, but also makes sure we're not trampling over variables set
	  up in the streaming thread from the seeking thread in case someone tries
	  to issue a seek just as the demuxer is parsing the headers.
	  Fixes #601617. Spotted by Ognyan Tonchev.

2009-12-03 17:49:55 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix debug message printf args
	  Fixes debug message printf format to make it build in mac's gcc

2009-12-02 13:33:20 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/shout2/gstshout2.c:
	  shout2: Convert delay correctly
	  Use GST_MSECOND to convert delay in msecs to nanosecs
	  Fixes #603547

2009-12-01 19:24:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: reset segment info after flush
	  Reset the segment info after a flush. We use the segment for handling QoS and if
	  we don't reset the segment, QoS is basically disabled after a flushing seek.

2009-12-01 15:07:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 87bf428 to 47cb23a

2009-12-01 14:15:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From da4c75c to 87bf428

2009-11-30 15:59:50 +0100  Aurelien Grimaud <gstelzz at yahoo dot fr>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: avoid buffer ref/unref pairs for CSRCs
	  We ref the buffer before pushing it downstream in order to get the CSRCs of it
	  after pushing. This causes performance problems when downstream elements want to
	  change the metadata because the buffer needs to be subbuffered.
	  Instead, read and store the CSRCs of the buffer in an array before pushing it
	  and process the array after pushing the buffer. This allows us to remove the
	  ref/unref pair.
	  Fixes #603376

2009-11-28 19:23:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/shout2/gstshout2.c:
	* ext/shout2/gstshout2.h:
	  shout2: use gstpoll for timeouts
	  Use our own GstPoll based timeout instead of the shout sleep so that we can
	  interrupt when doing a state change and shutting down.
	  Fixes #602887

2009-11-28 12:25:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  check: fix jitterbuffer check
	  Make sure we set a base_time on the element.
	  Fix the timeout to at least twice the jitterbuffer latency.
	  Enable previously failing tests.
	  Remove impossible checks.

2009-11-27 18:55:20 +0100  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From 53a2485 to da4c75c

2009-11-26 16:14:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: optionally merge NALUs into Access Units
	  ... which may be expected/desired by some downstream decoders
	  (and spec-wise highly recommended for at least non-bytestream mode).

2009-11-26 17:29:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix timestamp datatype

2009-11-25 10:38:23 -0600  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: avoid using wrong clock-rate
	  Check for a valid clock-rate before attempting to estimate the npt
	  stop time.

2009-11-25 10:37:30 -0600  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: fix typo in comments

2009-11-25 16:05:10 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffertest: add one more test and file a bug now
	  CHange the backwards test to always send first buffer first to have a define
	  basetime. Add another test that sends buffers backwards to assert that only
	  first sent buffer is keep and used as basetime. Disabled those tests still,
	  as its not passing/failing consitently and file a bug for jitterbuffer.

2009-11-25 10:17:34 +0200  Stefan Kost <ensonic@users.sf.net>

	* tests/check/elements/rtpjitterbuffer.c:
	  jitterbuffertest: improve the test
	  the tests are a bit more solid now but still not produce reliable results.
	  Wonder if they are still flawky or if its a bug in jitterbuffer.

2009-11-24 11:13:06 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: return error message on windows too.

2009-11-24 10:58:49 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: first phase of fixing up error reporting for windows.

2009-10-30 03:13:54 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: also set the suggested buf size for audio
	  We were only setting the suggested buf size for video,
	  we can set it for audio as well.
	  This and 195e14529d80ef318ce3a778c1995efb11f266cd
	  fix an issue that prevented seeking on large avi files
	  on WMP (non-recent versions).

2009-11-04 16:10:23 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  avimux: fix indx duration for PCM audio
	  GstBuffers for PCM audio usually contains more than
	  1 sample, we need to get the total number of samples to set
	  the indx duration.

2009-11-04 16:04:10 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/avi/gstavimux.c:
	  avimux: Audio buffers should be picked earlier
	  Adds a 0.5s advantage for audio buffers to being
	  picked earlier for muxing.

2009-11-24 16:40:19 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix push mode by making sure stbl information is available in next_entry_size ()

2009-11-24 16:35:20 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix order of arguments in log message

2009-11-24 15:51:21 +0200  Stefan Kost <ensonic@users.sf.net>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: fix spelling in comment

2009-11-23 17:58:17 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* common:
	  build system: Fix wrongly committed change to common/

2009-11-10 10:26:07 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Ease debugging by removing a goto for an error message

2009-11-14 15:52:09 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* common:
	* gst/qtdemux/qtdemux.c:
	  qtdemux: Parse per sample rather than all at once but build complete index when seeking

2009-11-04 17:31:15 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Save atom data for later use so it doesn't get freed after initial parsing

2009-11-06 11:00:04 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Parse from the previously parsed sample up to sample n

2009-11-04 17:04:22 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Make qtdemux_parse_samples () parse up to n samples

2009-10-28 17:49:02 +0000  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Separate off stbl sub-atom initialisation

2009-10-26 22:42:36 +0000  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Move variables into context in preparation for refactorisation

2009-10-26 20:36:08 +0000  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Fix bug where stps is never parsed due to logic error

2009-11-04 17:31:15 +0100  Robert Swain <robert.swain@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: Port ctts from Gnode * to GstByteReader

2009-10-23 13:06:44 +0100  Robert Swain <robert.swain@gmail.com>

	* gst/qtdemux/qtatomparser.h:
	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_dump.c:
	* gst/qtdemux/qtdemux_dump.h:
	* gst/qtdemux/qtdemux_types.h:
	  qtdemux: Switch from QtAtomParser to GstByteReader

2009-11-23 12:53:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix typo and grammar

2009-11-20 10:30:00 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix typo in mode enum description

2009-11-20 11:25:49 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/rtpmanager/gstrtpbin.c:
	  docs: more links and better short description
	  Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change
	  the short description to be more meaningful.

2009-11-20 09:58:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/wavpackparse.c:
	  wavpackparse: Fix unit test for recent position reporting changes

2009-11-19 16:09:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/wavpack/gstwavpackparse.c:
	  wavpackparse: After pushing a frame, update last_stop to the end of the frame
	  This improves position reporting, especially because of the fact that
	  WavPack frames are usually 0.5-1.0 seconds long.

2009-11-19 16:08:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/wavpack/gstwavpackparse.c:
	  wavpackparse: Allow pulling the last WavPack frame of a file
	  Because of a >= instead of a >, that last frame of a WavPack file
	  would never be parsed in pull mode.

2009-11-19 10:30:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 0702fe1 to 53a2485

2009-10-29 08:29:38 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	* gst/qtdemux/qtdemux_fourcc.h:
	  qtdemux: Add more fields to SVQ3 caps
	  qtdemux only added the whole stsd atom as 'codec_data'
	  in its output caps for SVQ3. This patch makes it add
	  the SEQH (inside a SMI atom) and a gamma field (taken
	  from the gama atom) if available.
	  Fixes #587922

2009-11-18 17:55:42 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Raise rank of muxer to PRIMARY

2009-11-18 17:54:16 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/y4m/gsty4mencode.c:
	  y4m: Raise rank of encoder to PRIMARY

2009-11-18 17:54:02 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/law/alaw.c:
	* gst/law/mulaw.c:
	  law: Raise rank of encoders to PRIMARY

2009-11-12 19:11:18 +0000  Bastien Nocera <hadess@hadess.net>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  Add user-id and user-pw properties
	  So that one doesn't need to modify the URL to have access
	  to authenticated RTSP streams.
	  fixes #601728

2009-11-18 12:22:10 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: use acquired flag when checking valid state
	  Use the acquired field of the ringbuffer in get_time to know when we are in an
	  invalid state. We don't clear the rate flag when releasing the ringbuffer so
	  this values is not usable.
	  Avoids some error messages being posted because the pulseaudio connection is
	  down.

2009-11-18 10:17:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  configure: bump core requirement to 0.10.25.1 as well
	  Make implicit requirement explicit.

2009-11-18 12:53:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: fix bogus memory chunk size check

2009-11-18 12:01:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: implement some more callbacks
	  Implement some more callbacks for debugging purposes.

2009-11-11 15:50:19 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: release lock before emiting signals
	  Release the jbuf lock before emiting the request-pt-map signal to avoid
	  deadlocks. We also need to catch the shutdown case when locking again.
	  Fixes #593354

2009-11-11 11:59:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvdepay.h:
	  rtp: add BroadcomVoice depayloader

2009-11-11 11:38:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpbvpay.c:
	  rtpbvpay: add rfc reference

2009-11-11 11:37:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpbvpay.h:
	  rtp: add BroadcomVoice payloader

2009-11-09 12:17:34 +0100  Jan Urbański <wulczer@wulczer.org>

	* gst/flv/gstflvmux.c:
	  flvmux: properly finish the ECMA array
	  The ECMA array with the file index was missing a mandatory end marker.
	  Fixes bug #601242.

2009-11-18 02:15:15 +0000  Jan Schmidt <thaytan@noraisin.net>

	* gst/deinterlace/gstdeinterlace.c:
	  Use new still-frame API from gst-plugins-base

2009-11-18 02:14:46 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	  Bump gst-plugins-base requirement to 0.10.25.1

2009-11-17 17:59:13 -0800  Michael Smith <msmith@songbirdnest.com>

	* gst/qtdemux/qtdemux.c:
	  qtdemux: identify IMA adpcm in qt properly.

2009-11-18 01:27:37 +0000  Jan Schmidt <thaytan@noraisin.net>

	* configure.ac:
	* win32/common/config.h:
	  Back to development -> 0.10.17.1

2009-11-17 01:53:08 +0000  Jan Schmidt <thaytan@noraisin.net>

	* gst-plugins-good.doap:
	  Add release 0.10.17 to the doap file