=== release 0.10.31 ===

2012-02-21  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  releasing 0.10.31, "Faster"

2012-02-20 12:22:12 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: force baseline profile is profile-level-id is unspecified
	  If profile-level-id is missing or invalid, we want any upstream
	  encoder to default to baseline profile, so specify that in the
	  caps we pass upstream. If the caps contain no profile restriction,
	  an encoder may default to high or main profile.

2012-02-17 17:21:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: fix switching from passthrough to non-passthrough when parameters change
	  commit b5bf0294 moved the if(need_new_coefficients) set_passthrough(equ)
	  after the if(is_passthrough) return FLOW_OK shortcut, so the passthrough
	  mode would never get updated even if the coefficients change.
	  Fixes equalizer-test doing .. nothing.

2012-02-16 17:14:20 +0800  Gary Ching-Pang Lin <chingpang@gmail.com>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: failure to query some optional controls is not a fatal error
	  Don't post a (fatal) error message on the bus just because we
	  failed to query some control. Fixes issue with built-in
	  Suyin Corp webcam for HP notebook (usbid 064e:e28a) on
	  OpenSuse 12.1, where querying red/blue balance fails.
	  https://bugzilla.gnome.org/show_bug.cgi?id=670197

2012-02-16 12:59:10 +0000  Tuukka Pasanen <tuukka.pasanen@ilmi.fi>

	* sys/v4l2/v4l2_calls.c:
	  v4l2src: fix for webcamstudio vloopback
	  Because vlooback emits 25 - ENOTTY and no EINVAL v4l2src thought it
	  can't handle this and does not work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=669455

2012-02-13 12:06:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/flacparse.c:
	  tests: flacparse: check and compare intended data

2012-02-09 22:12:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/mpegaudioparse.c:
	  tests: mpegaudioparse: remove stray declaration

2012-02-09 10:11:48 +0100  Marc Leeman <marc.leeman@gmail.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: typo fix (bytes send -> bytes sent)

2012-02-07 14:10:44 -0800  Ralph Giles <giles@mozilla.com>

	* ext/shout2/gstshout2.c:
	  shout2send: send video/webm through libshout.
	  This requires SHOUT_FORMAT_WEBM, added in libshout 2.3.0,
	  so video/webm support is contingent on that symbol being
	  defined.
	  Also an indentation change required by the pre-commit hook.
	  https://bugzilla.gnome.org/show_bug.cgi?id=669590

2012-01-28 11:13:16 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: avoid posting invalid duration for each frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=666583

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

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

2012-02-03 22:05:59 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/plugin.c:
	  pulseaudiosink: Lower rank to prevent autoplugging
	  pulseaudiosink breaks visualisations in its current form, so let's
	  prevent it from being autoplugged for the time being.
	  The best we can hope to do in the 0.10 series is query the list of
	  available sinks and their formats, and expose these as the bin's sinkpad
	  caps. While this is not a comprehensive solution, it will make sure that
	  we're only trying to support compressed formats if we're certain that
	  one exists.
	  The long-term fix for this will be in the form of proper upstream
	  renegotiation support in the 0.11/1.0 series.
	  https://bugzilla.gnome.org/show_bug.cgi?id=666361

2012-02-03 14:53:31 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: fix event leak when there is no peer on the src pad

2012-02-02 12:27:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: specify we only accept raw AAC in template caps
	  No header seems to be added, and the codec ID is the same as used
	  for raw by flvdemux, so raw seems the only supported case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665394

2012-02-02 12:25:21 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: specify we only output raw AAC in template caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=665394

2012-01-30 14:52:37 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtpmp2tpay.c:
	  rtpmp2tpay: do not try to flush a packet when no data is available
	  https://bugzilla.gnome.org/show_bug.cgi?id=668874

2010-06-11 08:36:33 +0200  Pascal Buhler <pascal.buhler@tandberg.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Exclude NALu size from payload length on truncated packets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667846

2012-01-28 13:05:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videobox/gstvideobox.c:
	  videobox: avoid wrapping opaque to transparent

2012-01-25 15:21:44 +0000  Jayakrishnan M <jay.krishnanm@gmail.com>

	* ext/cairo/Makefile.am:
	  cairo: fix build, make sure libgstvideo can be found
	  https://bugzilla.gnome.org/show_bug.cgi?id=668648

2012-01-25 13:19:12 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: don't pretend our random hostnames are fully-qualified domain names

2012-01-23 13:15:46 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: don't reveal the user's username, hostname or real name by default
	  Send a randomly made-up user@hostname as CNAME and don't
	  send a NAME at all by default.
	  https://bugzilla.gnome.org/show_bug.cgi?id=668320

2012-01-20 17:06:42 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: simplify internal src event debug logging
	  ... which avoids almost superfluous obtaining of rtsp element.

2012-01-20 17:03:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid NULL string comparison

2012-01-20 17:02:15 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	  rtpmp4adepay: prevent out-of-bound array access

2012-01-20 17:01:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/atomsrecovery.c:
	  isomp4: recovery: add sanity check
	  ... on possibly bogus/corrupt input data.

2012-01-20 16:58:28 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: remove redundant variable

2012-01-20 16:57:52 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix arithmetic for unsigned comparison

2012-01-20 16:55:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/imagefreeze/gstimagefreeze.c:
	  imagefreeze: add various missing break

2012-01-20 16:49:14 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/alpha/gstalphacolor.c:
	  alphacolor: remove redundant statement

2012-01-20 16:48:49 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: improve upstream peer duration querying
	  ... to avoid accepting unhandled duration query result.

2012-01-20 16:47:36 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: additional error condition checking

2012-01-20 16:46:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: additional error condition checking

2012-01-20 16:44:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: check _alloc_buffer result and perform fallback alloc if needed
	  ... rather than carrying on with NULL buffer.

2012-01-13 18:11:36 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: fix wrong error check
	  pa_stream_* functions return negative on error, despite the defines
	  for error codes being positive.
	  I only got to repro the error twice, so I'm not sure 100% sure this
	  fixes the issue (the negative var being uninitialized after returning
	  from pa_stream_get_latency).

2012-01-16 17:51:18 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/cutter/gstcutter.c:
	  cutter: fix leak of unused GValue

2012-01-16 16:10:08 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/elements/autodetect.c:
	  tests: fix autodetect test not testing correctly for state change success
	  State change to PAUSED can be done async, so if this happens, we need
	  to wait for the change to be done (or failed).

2012-01-16 15:42:46 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: fix caps leak

2012-01-16 12:13:50 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: make interlacedness test deterministic
	  If the interlaced flag is not present in the caps, we assume the
	  data is not interlaced, instead of leaving the boolean uninitialized.

2012-01-13 17:43:49 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	  oss4: fix caps leaks

2012-01-13 17:25:59 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: fix caps leak

2012-01-13 15:57:20 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

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

2012-01-13 10:32:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: plug pad leak in error code path
	  Based on patch by: Stig Sandnes <stig.sandnes@cisco.com>
	  Don't leak srcpad if there are no caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667820

2011-10-04 10:00:02 +0200  Stig Sandnes <stigsand@cisco.com>

	* sys/osxvideo/cocoawindow.m:
	  osxvideo: Fix leak of NSOpenGLPixelFormat object
	  https://bugzilla.gnome.org/show_bug.cgi?id=667818

2011-09-05 10:43:19 +0200  Havard Graff <havard.graff@tandberg.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Don't assert when the interface is not implemented.
	  Simply return FALSE instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=667817

2012-01-12 00:18:39 +0200  Raimo Järvi <raimo.jarvi@gmail.com>

	* sys/waveform/gstwaveformsink.c:
	* sys/waveform/gstwaveformsink.h:
	  waveformsink: Fix mingw warnings
	  https://bugzilla.gnome.org/show_bug.cgi?id=667719

2012-01-12 18:23:42 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  gstrtpssrcdemux: fix element leak

2012-01-12 14:19:22 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  matroska: do not leak attachment buffers

2012-01-12 10:30:11 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: do not drop the first data buffer on the floor (and leak it either)

2012-01-11 18:45:33 -0300  Reynaldo H. Verdejo Pinochet <reynaldo@collabora.com>

	* Android.mk:
	  Temporarily disabling multifile for the Android build
	  There is a hard dependency on inotify comming from gio. We
	  are not currently bundling inotify with the Android dist so
	  I'm disabling multifile for now until someone gets around
	  to sort this out.
	  This change fixes building on Android

2012-01-11 01:45:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/pipelines/wavenc.c:
	  tests: fix wavenc test on big endian
	  wavenc only accepts little-endian PCM, but most of our
	  elements such as audiotestsrc only produce or process
	  audio in native endianness, so we need to plug a
	  converter before wavenc on big endian systems.

2012-01-05 19:25:33 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  isomp4: fix caps leak

2012-01-05 19:08:03 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  isomp4: remove dead assignment

2012-01-04 19:40:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 11f0cd5 to cb5da59

2012-01-04 17:59:55 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: fix some leaks and remove files when done in qtmux test

2011-12-14 10:14:20 +0100  Peter Seiderer <ps.report@gmx.net>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: post better error message when we run out of disk space
	  Map write errno ENOSPC to GST_RESOURCE_ERROR_NO_SPACE_LEFT.

2011-12-27 11:50:03 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix valgrind warning
	  https://bugzilla.gnome.org/show_bug.cgi?id=666644

2011-12-21 13:22:03 +0100  John Ogness <john.ogness@linutronix.de>

	* gst/udp/gstudpsrc.c:
	  udpsrc: drop dataless UDP packets
	  It is allowed to send/receive UDP packets with no data. When such
	  a packet is available, select() will return with success but
	  ioctl(FIONREAD) will return 0. But a read() must still occur in
	  order to clear off the UDP packet from the queue.
	  This patch will read the dataless packet from the socket. If
	  select() was woken for other reasons (and FIONREAD returns 0),
	  this may result in a UDP packet getting accidentally dropped.
	  But since UDP is not reliable, this is acceptable.
	  NOTE: This patch fixes a nasty bug where sending a dataless
	  UDP packet to a udpsrc instance will cause an infinite
	  loop.
	  https://bugzilla.gnome.org/show_bug.cgi?id=666644
	  Signed-off-by: John Ogness <john.ogness@linutronix.de>

2011-12-21 20:50:21 +0100  Nicola Murino <nicola.murino@gmail.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix peer_caps leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=666688

2011-12-25 14:23:29 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: don't try to push already-freed buffers
	  Fixes unit test.

2011-09-09 11:42:09 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: let bsid 9 and 10 through
	  Files with 9 and 10 happen, and seem to comply with the <= 8
	  format, so let them through.
	  The spec says nothing about 9 and 10.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658546

2011-12-16 19:15:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: properly determine final duration
	  ... which can be authoratively obtained from our own written timestamps.

2011-12-19 13:56:30 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: only write full metadata at start
	  ... rather than having (potentially) unnecessary duplicates written all over,
	  or even contradictory varying filesize info, or duration info that will not
	  be rewritten upon header rewrite.

2011-12-21 17:43:10 +0100  Branko Subasic <branko@axis.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: do not consider duration of non-finalized file
	  ... to avoid it clamping requested seek position.
	  Non-finalized file case, determined by whether
	  _parse_blockgroup_or_simpleblock ever updates the segment duration.
	  Fixes #652195.

2011-12-21 15:06:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: improve decision to fall back to scanning when seeking
	  ... which is basically iff not streaming and no entry found in index

2011-12-13 18:18:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: filter bogus index entries with missing block number
	  ... to avoid contradictory information resulting in seeks sending more
	  downstream than needed for the corresponding segment.

2011-12-13 18:15:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: cater for safer arithmetic with global start time

2011-12-13 17:02:01 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: tweak final closing segment sending
	  ... to avoid it interfering with (sparse) stream syncing.

2011-12-12 11:54:56 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/glib-compat-private.h:
	  glib-compat: Add license boilerplate for LGPL

2011-12-12 15:15:46 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: mind (un)signed in some timestamp arithmetic
	  ... to avoid ending up with invalid (negative) duration.

2011-02-09 15:31:22 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: increase parse tolerance for fuzzy file cases

2011-12-12 10:38:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	  build: dist glib-compat-private.h properly
	  Add missing slash.

2011-12-12 10:18:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/souphttpsrc.c:
	  tests: use atexit, g_atexit has been deprecated in glib master

2011-12-12 02:52:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/dv/gstdvdemux.c:
	* ext/flac/gstflacdec.c:
	* ext/wavpack/gstwavpackparse.c:
	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/isomp4/gstqtmoovrecover.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/videomixer/videomixer2.c:
	* gst/wavparse/gstwavparse.c:
	  Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly
	  GStaticRecMutex is part of our API/ABI, not much we can do here
	  in 0.10 for most of these.

2011-12-12 02:41:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/souphttpsrc.c:
	* tests/icles/equalizer-test.c:
	* tests/icles/gdkpixbufsink-test.c:
	* tests/icles/test-oss4.c:
	* tests/icles/videocrop-test.c:
	  tests: g_thread_init() is deprecated in glib master
	  It's not needed any longer.

2011-12-12 02:38:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpclientsink.c:
	* gst/rtpmanager/gstrtpsession.c:
	* sys/oss4/oss4-mixer.c:
	* tests/icles/v4l2src-test.c:
	  Use g_thread_try_new() instead of g_thread_crate() with newer glib versions

2011-12-12 02:31:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalpha.h:
	  alpha: use new glib API for static mutex if available

2011-12-12 02:30:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* Makefile.am:
	* ext/jack/gstjackaudioclient.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/soup/gstsouphttpclientsink.c:
	* gst-libs/gst/glib-compat-private.h:
	* gst/audiofx/audiochebband.c:
	* gst/audiofx/audiocheblimit.c:
	* gst/audiofx/audiofirfilter.c:
	* gst/audiofx/audioiirfilter.c:
	* gst/audiofx/audiowsincband.c:
	* gst/audiofx/audiowsinclimit.c:
	* gst/equalizer/gstiirequalizer.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer2.c:
	* sys/oss4/oss4-mixer.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/ximage/gstximagesrc.c:
	  Work around deprecated thread API in glib master
	  Add private replacements for deprecated functions such as
	  g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
	  to avoid the deprecation warnings. We'll change these
	  over to the new API once we depend on glib >= 2.32.

2011-12-12 10:24:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  configure: Require GLib >= 2.24
	  All other modules require this already and nobody is testing with
	  older versions anyway.

2011-12-11 18:40:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	  gdkpixbufsink: fix inverted pixel-aspect-ratio
	  Spotted by Mike Morrison.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665882

2011-12-11 17:55:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: don't leak pad template

2011-12-10 15:13:07 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* gst/deinterlace/tvtime-dist.c:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videomixer/blendorc-dist.c:
	* po/eo.po:
	* win32/common/config.h:
	  0.10.30.2 pre-release

2011-12-10 14:48:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpclientsink.c:
	  soup: fix start/stop race in souphttpclientsink
	  Fix crash or hang in generic/states unit test when doing stop()
	  right after start(). Create main loop in the start function already
	  and not just in the thread function, so that stop() always has a
	  valid main loop to quit on. Also, calling g_main_loop_quit() before
	  g_main_loop_run() won't work and result in the stop function waiting
	  for the thread to join forever. Therefore, wait for the thread to
	  be ready and get the main loop running in the start() function, to
	  be sure stop() always works.

2011-12-10 13:35:08 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/files/Makefile.am:
	  tests: dist test file used in matroskaparse unit test

2011-12-10 12:32:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/rgvolume.c:
	  tests: fix up rgvolume test for basetransform event caching
	  Some tests assumed that tag events would always pushed through
	  immediately, which isn't the case any longer, so push a newsegment
	  event and an empty buffer first.

2011-12-10 02:21:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* po/LINGUAS:
	* po/eo.po:
	* po/ja.po:
	* po/lv.po:
	* po/sr.po:
	  po: update translations

2011-12-09 15:50:28 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	  jack: don't leak client name when freeing the element
	  And add gtk-doc chunks for the new property.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665872

2011-12-09 15:45:03 +0000  Nicolas Baron <hoggins@radiom.fr>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jack/gstjackaudiosrc.h:
	  jack: add "client-name" property to jackaudiosink and jackaudiosrc
	  https://bugzilla.gnome.org/show_bug.cgi?id=665872

2011-12-08 11:00:45 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: stream-format=raw goes with aac caps, not mp3 caps

2011-12-02 12:07:24 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: do not ignore the highest frame interval
	  https://bugzilla.gnome.org/show_bug.cgi?id=665387

2011-12-02 11:59:03 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: do not ignore the largest resolution
	  The 'max' value isn't an STL style "one after the end" bound,
	  but the largest allowed value.
	  https://bugzilla.gnome.org/show_bug.cgi?id=665387

2011-12-06 16:47:25 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/multifile/gstmultifilesink.h:
	  docs: add add the two enum values that were just added too

2011-12-06 16:14:54 +0100  Stefan Sauer <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: expose the enum property docs for splitting mode.
	  Fixes #665666.

2011-12-05 12:15:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: replace deprecated GST_CLASS_LOCK

2011-11-24 13:58:01 +0100  Sebastian Rasmussen <sebrn@axis.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Ceil jpeg dimensions, instead of floor
	  A JPEG image inside an RTP stream has a preceeding RFC2435 header that
	  conveys width/height. The dimensions in this header are limited to be
	  multiples of 8. Since JPEG uses an MCU of 8x8 pixels any image must
	  already indirectly have image data dimensions that are rounded up in
	  order to contain enough data to render the image. Therefore this fix
	  safely rounds the image dimensions in the RFC2435 header up to the
	  closest multiple of 8.

2011-12-04 12:50:57 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: ensure we only check for sample/block mixup at start
	  Otherwise we might trigger at some point within the file, but the
	  check is only making sense for the second block.

2011-12-03 18:14:59 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: warn if accumulating headers after they were pushed
	  https://bugzilla.gnome.org/show_bug.cgi?id=665412

2011-10-25 12:54:43 -0700  David Schleef <ds@schleef.org>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: fix parsing
	  Mark more parts as belonging to streamheaders.

2011-12-03 17:30:10 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix discontinuity threshold check when timestamps go backwards
	  Since unsigned types are used, a negative value would show as very, very
	  positive.
	  Fixes A/V sync on some... less than well made files where timestamps go
	  backwards.

2011-12-02 12:01:22 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2src: add a comment about a "hidden" assumption on rank values
	  https://bugzilla.gnome.org/show_bug.cgi?id=665387

2011-12-01 14:13:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	  tests: fix up LIBS order som more`

2011-12-01 13:22:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: fix name of new property and the unit test
	  https://bugzilla.gnome.org/show_bug.cgi?id=654379

2011-09-25 14:57:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: add basic buffer list handling
	  We assume for now that all buffers in a buffer list
	  should end up in the same file (so we can group GOPs
	  in buffer lists, for example). Could optimise this
	  a bit to avoid the memcpy.

2011-09-23 18:43:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: write stream-headers when switching to the next file in max-size mode

2011-09-23 18:31:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: add new 'max-size' mode for switching to the next file

2011-09-23 17:49:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	  multifilesink: add "max-file-size" property for new next-file mode

2011-12-01 13:38:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't forget SSA subtitles in last commit

2011-12-01 13:34:52 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroskademux: Only check for markup and escape if necessary for plaintext subtitles
	  Otherwise we break USF and ASS/SSA subtitles.

2011-12-01 13:23:33 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/Makefile.am:
	  multifile: fix build in uninstalled setup
	  Add -base libs includes to CFLAGS, fix order of LIBS <cit>.

2011-12-01 13:08:01 +0100  Alessandro Decina <alessandro.d@gmail.com>

	* tests/check/elements/multifile.c:
	  tests: fix g_mkdtemp presence check in multifile tests
	  g_mkdtemp was added in glib 2.30 even though the doc claims it was added in
	  2.26.

2011-07-17 23:56:04 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/Makefile.am:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	* tests/check/Makefile.am:
	* tests/check/elements/multifile.c:
	  multifilesink: add flag to cut after a force key unit event

2011-12-01 12:47:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Copy all buffer flags when creating a subtitle buffer copy after postprocessing
	  This also copies the caps. Otherwise we could end up pusing
	  the first buffer without any caps, which causes downstream
	  to not get notified about the caps.
	  Fixes bug #664892.

2011-10-11 02:07:13 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/matroska/matroska-mux.c:
	  matroskamux: make default framerate optional per stream
	  there is at least two use cases where default frame rate
	  should or may be disabled:
	  - vp8 stream with altref frame enabled. If default frame rate
	  is enabled, some players will missinterprete it (critical!)
	  - for webm container, to reduce micro overhead
	  - for stream with variable frame rate.
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>

2011-11-30 22:13:11 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstripple.c:
	  rippletv: fix CLAMP end-values

2011-11-30 19:25:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	  docs: update docs

2011-11-30 19:00:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/Makefile.am:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/patternspec.c:
	* gst/multifile/patternspec.h:
	  splitfilesrc: specify filenames via normal wildcards instead of regular expressions
	  Less cracktastic in the end.

2011-10-10 18:28:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: check bytes actually read, just in case
	  Handle corner case where we try to read beyond the end of the
	  last file part, in which case we want to return a short read.
	  If we get fewer bytes than expected for any other file part,
	  we should just error out, since something fishy's going on
	  then.

2011-10-06 08:33:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multifile/gstsplitfilesrc.c:
	  splitfilesrc: set offsets on buffers
	  Looks like some parsers (in some versions at least) expect the
	  offsets to be set, and behave weird if that's not the case
	  (e.g. off-by-one in h264parse).

2011-07-28 20:19:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	* gst/multifile/Makefile.am:
	* gst/multifile/gstmultifile.c:
	* gst/multifile/gstsplitfilesrc.c:
	* gst/multifile/gstsplitfilesrc.h:
	  multifile: add splitfilesrc element
	  Add new splitfilesrc element that presents multiple files
	  (selectable via a location regex) as one single contiguous
	  file.

2011-11-29 17:34:10 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/pulse/pulseaudiosink.c:
	  Revert "pulseaudiosink: fix caps leak"
	  This reverts commit d6a9de9e2aedc8b66ab3219902b5a37e8d65ada2.
	  setcaps functions aren't supposed to take ownership of the caps passed

2011-11-28 12:58:44 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/aalib/gstaasink.c:
	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/cairo/gstcairooverlay.c:
	* ext/cairo/gstcairorender.c:
	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttimeoverlay.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/esd/esdmon.c:
	* ext/esd/esdsink.c:
	* ext/flac/gstflacdec.c:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflactag.c:
	* ext/gconf/gstswitchsink.c:
	* ext/gconf/gstswitchsrc.c:
	* ext/gdk_pixbuf/gstgdkpixbuf.c:
	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/hal/gsthalaudiosink.c:
	* ext/hal/gsthalaudiosrc.c:
	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosrc.c:
	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libcaca/gstcacasink.c:
	* ext/libmng/gstmngdec.c:
	* ext/libmng/gstmngenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/mikmod/gstmikmod.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	* ext/shout2/gstshout2.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpsrc.c:
	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexenc.c:
	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* ext/taglib/gsttaglibmux.c:
	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackparse.c:
	* gst/alpha/gstalpha.c:
	* gst/alpha/gstalphacolor.c:
	* gst/apetag/gstapedemux.c:
	* gst/audiofx/audiopanorama.c:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/auparse/gstauparse.c:
	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosrc.c:
	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosrc.c:
	* gst/avi/gstavidemux.c:
	* gst/avi/gstavimux.c:
	* gst/avi/gstavisubtitle.c:
	* gst/cutter/gstcutter.c:
	* gst/debugutils/breakmydata.c:
	* gst/debugutils/cpureport.c:
	* gst/debugutils/efence.c:
	* gst/debugutils/gstcapsdebug.c:
	* gst/debugutils/gstcapssetter.c:
	* gst/debugutils/gstnavigationtest.c:
	* gst/debugutils/gstnavseek.c:
	* gst/debugutils/gstpushfilesrc.c:
	* gst/debugutils/gsttaginject.c:
	* gst/debugutils/progressreport.c:
	* gst/debugutils/rndbuffersize.c:
	* gst/debugutils/testplugin.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvmux.c:
	* gst/flx/gstflxdec.c:
	* gst/goom/gstgoom.c:
	* gst/goom2k1/gstgoom.c:
	* gst/icydemux/gsticydemux.c:
	* gst/id3demux/gstid3demux.c:
	* gst/imagefreeze/gstimagefreeze.c:
	* gst/interleave/deinterleave.c:
	* gst/interleave/interleave.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/qtdemux.c:
	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-decode.c:
	* gst/law/mulaw-encode.c:
	* gst/level/gstlevel.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	* gst/median/gstmedian.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesrc.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/replaygain/gstrganalysis.c:
	* gst/replaygain/gstrglimiter.c:
	* gst/replaygain/gstrgvolume.c:
	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg726pay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpsirenpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/smpte/gstsmpte.c:
	* gst/smpte/gstsmptealpha.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstaspectratiocrop.c:
	* gst/videocrop/gstvideocrop.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideobalance.c:
	* gst/videofilter/gstvideoflip.c:
	* gst/videomixer/videomixer.c:
	* gst/videomixer/videomixer2.c:
	* gst/wavenc/gstwavenc.c:
	* gst/wavparse/gstwavparse.c:
	* gst/y4m/gsty4mencode.c:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	* sys/oss4/oss4-sink.c:
	* sys/oss4/oss4-source.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxvideo/osxvideosink.m:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/waveform/gstwaveformsink.c:
	* sys/ximage/gstximagesrc.c:
	* tests/check/elements/qtmux.c:
	  various: fix pad template leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=662664

2011-11-28 11:47:11 +0100  Chad <channa@caltech.edu>

	* gst/debugutils/gsttaginject.c:
	  taginject: set gap-aware
	  The element does not modify the data anyway.

2011-11-26 21:39:33 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: also sync the parameters for the filter bands

2011-11-26 16:06:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-ids.c:
	  matroskademux: initialise seen_markup_tag field on subtitle stream context

2011-11-25 19:28:55 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/gstqtmuxmap.c:
	  ismlmux: Use iso-fragmented as variant type
	  Using 'iso' conflicts with mp4mux variant type, ismlmux now
	  uses iso-fragmented
	  Fixes #656823

2011-11-24 12:05:33 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: Implement GstStreamVolume interface
	  PulseAudio 1.0 supports per-source-output volumes, and this exposes the
	  functionality via the GstStreamVolume interface.
	  When compiled against pre-1.0 PulseAudio, the interface is not
	  implemented, and the "volume" or "mute" properties are not available.
	  This bit of ugliness will go away when we can depend on PulseAudio 1.0
	  or greater.
	  https://bugzilla.gnome.org/show_bug.cgi?id=595055

2011-09-10 21:21:38 -0700  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Trivial comment copy-paste-o fix

2011-11-14 12:43:27 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: Remove redundant code

2011-11-14 12:41:41 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: Clean up refcounting in event probe
	  Makes sure we don't leak a refcount if the object is disposed before a
	  NEWSEGMENT turns up.

2011-11-24 16:31:38 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix seeking
	  Which I accidentally broke when fixing flv videos breaking on
	  spurious timestamp discontinuities in broken files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=631430

2011-11-25 13:13:47 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstradioac.c:
	* gst/effectv/gstradioac.h:
	  effectv: repair color modes in radioactv by taking rgb,bgr into account

2011-11-25 11:44:49 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstradioac.c:
	  radioactv: add one more set of caps
	  It also work in this format. Avoids the need for conversion.

2011-11-25 11:44:18 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstradioac.c:
	* gst/effectv/gstshagadelic.c:
	  effecttv: fix reverse negotiation
	  The plugins were using _fixed_caps_ and thus not adjusting to new upstream
	  sizes. Spotted by Tim Müller.

2011-11-25 11:43:16 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstwarp.c:
	  warptv: remove not needed ifdef

2011-11-25 10:15:35 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstripple.c:
	  rippletv: clean up the rendering code a bit
	  This is corrrupts the memoy when resizing. Add a FIXME to make it resizeable
	  once that is solved.

2011-11-24 20:42:49 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst/effectv/gstquark.c:
	* gst/effectv/gststreak.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	  effecttv: fix reverse negotiation
	  The plugins were using _fixed_caps_ and thus not adjusting to new upstream
	  sizes. Spotted by Tim Müller.

2011-11-24 14:14:53 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: Fix leak of filename strings
	  Do not forget to free the filename strings when deleting
	  the list of files.

2011-11-24 14:11:33 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* tests/check/elements/multifile.c:
	  multifile: fix build of tests
	  Tests fail to build because g_mkdtemp is available from glib since
	  2.26.
	  This patch adds a condition around the redefinition of
	  g_mkdtemp on the tests to only build it if glib is older than
	  2.26.

2011-09-27 16:49:45 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: skip id32 tags
	  This allows decoding at least one sample where something has
	  stuffed some ID3 tag before the (supposedly initial) FMT\ .
	  https://bugzilla.gnome.org/show_bug.cgi?id=660249

2011-10-31 17:06:18 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/effectv/gstedge.c:
	  edgetv: trivial comment fix for clarity
	  https://bugzilla.gnome.org/show_bug.cgi?id=661841

2011-10-31 17:04:23 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/effectv/gstedge.c:
	  edgetv: don't leave bits of the output buffer uninitialized
	  Let's initialize them to zero. It looks alright, but then it
	  also looks alright with v3, or with the corresponding pixels
	  from the source. I don't know what the original intent would
	  be, and the original effectv source also has this bug/feature.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661841

2011-11-24 10:25:02 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparse: Use the sinkpad template caps as fallback, not the srcpad ones

2011-11-24 09:59:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:57:57 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:55:47 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:53:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	  amrparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:49:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	  amrparse: Mark some more functions as static

2011-11-24 09:48:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-24 09:44:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Mark some functions as static and remove unused function declarations

2011-11-24 09:43:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream

2011-11-23 00:57:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/matroskaparse.c:
	* tests/files/pinknoise-vorbis.mkv:
	  tests: add basic unit test for matroskaparse

2011-11-23 00:56:26 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: don't leak stream headers
	  https://bugzilla.gnome.org/show_bug.cgi?id=664548

2011-11-16 19:08:05 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: ensure to free allocated padded data

2011-11-16 18:57:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: reset tag setter interface when appropriate

2011-11-16 18:57:21 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/gstflacenc.c:
	  flacenc: reset tag setter interface when appropriate

2011-11-14 15:34:57 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstflacparse.h:
	  flacparse: detect when a file lies about fixed block size
	  If the sample/block number happens to be the same as the block
	  size, we assume variable block size, and thus counters in samples
	  in the headers. This can only get us a false positive for a block
	  size of 1, which is invalid. We can get false negatives more
	  often though (eg, if not starting at the start of the stream),
	  but then that's already GIGO.

2011-09-02 19:20:07 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	  gstrtpsession: Add special mode to use FIR as repair as Google does
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-09-01 17:47:38 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: Send FIR requests in response to key unit requests with all-headers=TRUE
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-09-01 16:25:21 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.h:
	  rtpsession: Put the PLI requests in each RTPSource
	  Also refactor a bit and put all the keyframe request code in one
	  place inside rtpsession.c
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-08-31 14:35:33 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Hack to FIR because Google doesn't set the sender ssrc correctly
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-08-30 19:06:13 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Process received Full Intra Requests
	  Process FIR requests according to RFC 5104
	  https://bugzilla.gnome.org/show_bug.cgi?id=658419

2011-11-07 18:43:26 +0000  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Set pixel-aspect-ratio to 1/1
	  We don't currently support setting the pixel-aspect-ratio from V4L2. So
	  simply set it to be 1/1 in the caps to prevent negotiation failures when
	  fixating to weird values (e.g. when the downstream caps has
	  pixel-aspect-ratio = [ MIN, MAX ] )
	  https://bugzilla.gnome.org/show_bug.cgi?id=663580

2011-11-11 10:06:25 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/pulse/pulseaudiosink.c:
	  pulseaudiosink: fix caps leak

2011-11-11 14:55:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: do not leak clientname when setting up property

2011-11-11 18:05:35 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulse: Chain up dispose() in pulseaudiosink

2011-11-08 15:35:26 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: fix wrong stride when inverting uncompressed video
	  Such frames have a stride multiple of 4, see
	  http://lscube.org/pipermail/ffmpeg-issues/2010-April/010247.html.
	  This showed up on a sample using a odd width of 24 bit video.
	  https://bugzilla.gnome.org/show_bug.cgi?id=652288

2011-11-09 10:32:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: minimal sanity check on creation datetime

2011-11-02 12:58:12 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Return the sink pad template as sink caps, not the src's
	  https://bugzilla.gnome.org/show_bug.cgi?id=577784

2009-03-15 19:26:48 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Also implement size/framerate restrictions in getcaps
	  https://bugzilla.gnome.org/show_bug.cgi?id=577784

2009-03-04 20:50:19 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Implement getcaps following RFC 4629, picks the right annexes
	  https://bugzilla.gnome.org/show_bug.cgi?id=577784

2011-11-08 14:31:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: also set segment stop at startup rather than only post seek
	  ... so as to ensure consistent playback with or without seek, especially
	  in presence of some bogus edit list entries.

2011-11-02 17:02:54 +0000  Raul Gutierrez Segales <rgs@collabora.co.uk>

	* gst/flv/Makefile.am:
	  gst/flv/: add amfdefs.h to noinst_HEADERS
	  https://bugzilla.gnome.org/show_bug.cgi?id=663334

2011-10-03 17:50:43 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: detect large pts gaps and resync
	  Should work on multiple gaps, but tested on only one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=631430

2011-08-22 10:40:45 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix off by one between granpos and last_stop

2011-10-07 19:41:35 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix last frame timestamp in fixed block size mode
	  The last block may have a different block size, so we should not
	  use it to scale or we'll end up with a wrong timestamp.
	  See comment and quote from the FLAC format documentation in the code.
	  Fixes looped playback of FLAC files (via about-to-finish).
	  https://bugzilla.gnome.org/show_bug.cgi?id=661215

2011-10-27 15:52:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/cairo/gsttextoverlay.c:
	* ext/cairo/gsttextoverlay.h:
	  cairotextoverlay: add a 'silent' property to skip rendering
	  https://bugzilla.gnome.org/show_bug.cgi?id=662856

2011-11-07 12:00:12 +0100  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroskamux: fix regression causing malformed files
	  This was caused by me in 1b213d. It seems I was too focused on 0.11 when I did
	  this and tested the wrong branch.
	  The problem was reported by Alexey Fisher.

2011-11-03 23:28:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtp: use GLib's G_BIG_ENDIAN define instead of BIG_ENDIAN
	  Fixes compiler warning on mingw32

2011-10-31 16:18:32 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: avoid shortcut evaluation when adding paired mp4 tag
	  Fixes (part of) #638711.

2011-10-31 15:43:25 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: do not use unoffical V_MJPEG codec id
	  ... but as not spec'ed especially, consider it a VfW compatibility case.
	  Fixes #659837.

2011-10-30 19:30:14 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/flac/gstflacenc.h:
	  flacenc: remove dead code from header
	  We require a new-enough libflac that this condition will never apply.

2011-10-28 09:57:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg
	  jpegdec (using libjpeg 6.2/8) can't decode some lossless types of JPEG.
	  https://bugzilla.gnome.org/show_bug.cgi?id=556648

2011-10-28 12:30:33 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: elaborate some debug statements

2011-10-11 20:56:51 +0400  Stas Sergeev <stsp@users.sourceforge.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: be careful with negative cts
	  Fixes #661477.

2011-10-06 13:04:54 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: tune non-update seek handling cases
	  Fixes #661049.

2011-10-28 10:40:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Use the clip function instead of the prepare_buffer function

2011-10-28 09:36:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/gstcollectpads2.c:
	* gst/videomixer/gstcollectpads2.h:
	* gst/videomixer/videomixer2.h:
	* gst/videomixer/videomixer2pad.h:
	  videomixer2: Use collectpads2 from core

2011-10-28 00:41:45 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Don't pointlessly hold object lock over caps operations
	  Avoids a deadlock when getcaps is recursive due to the getcaps being
	  reflected upstream/downstream. The lock isn't actually protecting
	  anything here.

2011-10-27 00:37:03 +1100  Jan Schmidt <thaytan@noraisin.net>

	* gst/flv/amfdefs.h:
	* gst/flv/gstflvmux.c:
	  flvmux: add some comments and defines to clarify code.

2011-10-10 15:36:14 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroska: refactor ebml-write to be more 0.11 friendly
	  Switching to a more 0.11-friendly pattern, where getting the buffer's data
	  pointer and setting the size many times is less natural. This is of course in
	  preparation to the upcoming port of the plugin.

2011-10-11 21:45:46 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/ebml-write.c:
	  matroska: remove stale floatcast include
	  GDOUBLE_TO_BE was moved to core a long time ago.

2011-10-11 22:10:27 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix possible crash with malformed dirac codec_data
	  Since size is unsigned, we need to safeguard against wrapping below zero.

2011-10-21 22:33:34 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/equalizer/gstiirequalizer.c:
	  equalizer: remove avoidable call to gst_object_set_name

2011-10-21 22:32:38 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: remove avoidable call to gst_object_set_name

2011-10-16 20:30:25 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/gstpngenc.c:
	  pngenc: increase arbitrary resolution limits
	  Apparently libpng can technically do up to 2^31-1 rows and columns. However it
	  imposes an (arbitrary) default limit of 1 million (that could theoretically be
	  lifted by using some additional API).
	  Moved array allocation to the heap now.

2011-10-16 20:25:41 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/gstpngenc.c:
	  pngenc: don't unconditionally allocate 4096 pointers on the stack
	  Instead allocate as many as needed (on the stack still).

2011-10-16 20:05:28 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/gstpngenc.c:
	  pngenc: ensure setcaps was called before chain function
	  This is needed to properly error out for e.g. "fakesrc ! pngenc ! fakesink".

2011-10-16 19:44:27 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/gstpngenc.c:
	  pngenc: validate input buffer size
	  Just for safety; of course such mismatch represents a bug in another element.

2011-10-16 19:41:28 +0200  René Stadler <mail@renestadler.de>

	* ext/libpng/Makefile.am:
	* ext/libpng/gstpngenc.c:
	* ext/libpng/gstpngenc.h:
	  pngenc: make setcaps more robust, use gstvideo functions
	  A setcaps function needs to actually verify the caps carefully. In this case,
	  it was possible to e.g. link a video decoder with YUV+RGB template caps to
	  pngenc.  That would cause a crash when the decoder pushes a YUV buffer. Same
	  thing when pushing a valid buffer that exceeds the resolution limits.
	  Also, missing framerate caps field would cause a glib critical warning due to
	  invalid GValue. This fails hard now.

2011-10-21 10:01:43 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  ebml: small correction to previous commit
	  Signal a short read with UNEXPECTED, exactly like the peek_bytes function.

2011-10-19 13:09:51 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/matroska/matroska-read-common.c:
	  ebml: Fix push-based behaviour
	  The 'peek' method was completely wrong (!?)

2011-10-18 18:31:17 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulseaudiosink.c:
	  pulse: Get caps correctly on pad block
	  Instead of always going upstream, we should first see if already got
	  caps from a setcaps() call.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661262

2011-10-18 12:25:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/wavpack/gstwavpackenc.c:
	  wavpackenc: don't unref buffer with gst_object_unref()

2011-10-18 12:05:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: only use is_pcm for 1.0 of pulseaudio

2011-10-18 11:58:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: only disable trickmodes for !pcm
	  Only disable trickmodes when we are not dealing with raw PCM samples.

2011-10-14 10:56:16 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Fix a leak
	  Buffers weren't being unref'ed in one case inside, causing memory usage
	  to blow up.

2011-10-14 09:10:01 +0200  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  set colour masks for video/x-raw-rgb in rtpvrawdepay

2011-10-13 16:59:50 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/videomixer/videomixer2.c:
	  videomixer2: Fix incorrect gst_buffer_replace() call
	  This got exposed when gst_buffer_replace() was changed from a macro to a
	  function.

2011-10-12 11:26:50 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  rtpvrawpay: Only use 24 LSB for depth=24 RGB caps
	  ... and indent the masks for clarity

2011-10-11 14:58:43 +0200  René Stadler <rene.stadler@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix segment handling, so we actually use running time
	  gst_matroska_mux_best_pad adjusts the buffer timestamp to running time using
	  the segment stored in the pad's collect data. However, the event handler didn't
	  pass the newsegment event on to collectpads' handler, so this segment was never
	  updated at all.
	  Re-fixes bug #432612.

2011-10-10 19:01:23 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/rtp/gstrtpg722pay.c:
	  gstrtpg722pay: Compensate for clockrate vs. samplerate difference
	  The RTP clock-rate used for G722 is 8000, even though the samplerate is
	  16000. Compensate for this by pretending G722 has 8 bits per sample
	  instead of the 4 bits as if it were a codec that ran at half the speed,
	  but with twice the number of bits. Fixes #661376

2011-09-27 19:25:53 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Implement upstream negotiation
	  Add upstream negotiation for jpegdec. Fixes #660275

2011-10-10 19:02:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: don't leak audio codec_data buffer

2011-10-10 13:20:04 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/examples/cairo/Makefile.am:
	  tests: add missing PLUGIN_ASE_LIBS to LDADD

2011-10-09 21:31:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	* ext/speex/gstspeexenc.h:
	  speexenc: only push header buffers following initial events

2011-10-09 11:18:18 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/atomsrecovery.c:
	  qtmux: Fix memory leak on atoms recovery function
	  Remember to free the ftyp data after writing it to a file.
	  Fixes #660969

2011-09-21 18:45:42 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: improve segment handling with non-zero starting timestamp
	  ... as well as related items, such as seeking and position reporting.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659808

2011-09-29 18:41:53 +0400  Stas Sergeev <stsp@users.sourceforge.net>

	* sys/v4l2/gstv4l2object.c:
	* sys/ximage/gstximagesrc.c:
	  v4l2, ximagesrc: fix some printf format compiler warnings
	  https://bugzilla.gnome.org/show_bug.cgi?id=660150

2011-09-30 12:42:22 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: Refactor bitrate check test
	  Refactor bitrate check test to accomodate multiple tests
	  for bitrate

2011-09-30 13:02:31 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/atoms.c:
	  qtmux: update esds atom under wave atom for aac bitrates
	  AAC in mov format puts an ESDS atom inside of a WAVE atom in
	  STSD atom, we need to update the bitrate on this ESDS. This patch
	  fixes it.

2011-09-30 12:41:52 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/atoms.c:
	* gst/isomp4/fourcc.h:
	  qtmux: Also update btrt atom
	  When rewriting bitrates, also update the btrt atom under stsd

2011-09-30 10:55:53 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: add tests for bitrate average calculation
	  Adds tests to make sure qtmux/mp4mux sets average bitrate
	  correctly

2011-09-28 11:41:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Calculate average bitrate for streams
	  Calculate and use average bitrate for streams when no
	  bitrate tag was received

2011-09-28 10:41:14 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Avoid a buffer metadata copy if possible
	  If first_ts is 0 there is no need to subtract, so we might
	  skip some copying to make the buffer metadata writable.

2011-09-29 23:21:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: initialise variable before adding to it

2011-09-29 17:21:22 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexdec.c:
	* ext/speex/gstspeexdec.h:
	  speexdec: port to audiodecoder

2011-09-29 16:33:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/gstspeexenc.h:
	  speexenc: clean up some unused remnants

2011-09-29 17:32:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/speex/Makefile.am:
	* ext/speex/gstspeexenc.c:
	* ext/speex/gstspeexenc.h:
	  speexenc: port to audioencoder

2011-09-28 16:09:58 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/flac/Makefile.am:
	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  flacenc: port to audioencoder

2011-09-27 15:59:24 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-parse.c:
	  matroskademux: ensure minimal alignment for audio/x-raw-* buffers
	  Since matroskademux will attempt to push unaligned buffers,
	  downstream might have trouble with those, especially if downstream
	  uses ORC, such as audioconvert.
	  Ensure we push buffers aligned to the basic type at least for
	  those raw buffers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659798

2011-09-28 00:10:09 +0300  Raimo Järvi <raimo.jarvi@gmail.com>

	* gst/goom2k1/goom_core.c:
	  goom2k1: Fix compiler warnings on 64 bit mingw-w64
	  Fixes bug #660294.

2011-09-25 15:13:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/Makefile.am:
	* ext/soup/gstsoup.c:
	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpclientsink.h:
	* ext/soup/gstsouphttpsink.c:
	* ext/soup/gstsouphttpsink.h:
	  soup: rename souphttpsink to souphttpclientsink
	  To avoid confusion, and because we might want a server
	  sink at some point too.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659947

2011-09-23 16:39:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/soup/gstsouphttpsink.c:
	* ext/soup/gstsouphttpsink.h:
	  souphttpsink: don't create unused second sink pad object
	  The base class will create the sink pad.

2011-09-23 15:36:36 +0200  Julien Isorce <julien.isorce@gmail.com>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: correctly check for ac3/e-ac3 switch
	  https://bugzilla.gnome.org/show_bug.cgi?id=659943

2011-09-20 13:38:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: improve downstream flow return feedback to upstream
	  ... although basertpdepay does not really make it easy/possible to do so
	  all the way.

2011-09-20 12:11:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	* sys/ximage/gstximagesrc.h:
	  ximagesrc: add xid and xname properties to allow capturing a particular window
	  A particular window may be selected using the new xid (X-Window
	  XID, eg a pointer) and xname (window title) properties. If both
	  are specified, the XID is used in preference, falling back to
	  xname if not found.
	  Default (if none of xid and xname are specified, or if no such
	  window is found) is to capture the root window.
	  https://bugzilla.gnome.org/show_bug.cgi?id=546932

2011-08-02 17:39:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  tests: add unit test to make sure encodebin picks mp4mux for variant=iso
	  https://bugzilla.gnome.org/show_bug.cgi?id=651496

2011-09-19 12:15:11 +0200  Ha Nguyen <hanguytv@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Fix a leaked clock for each buffering message
	  Fixes bug #659237.

2011-09-19 12:11:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: parse embedded ID32 tags

2011-09-02 13:41:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpsession: avoid source premature timing out
	  Use slightly adjusted sender interval to determine sender timeout rather than
	  our own sender side interval (which may have been forced small).

2011-08-25 12:40:52 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: avoid timing out source too quickly
	  ... following a PAUSE/PLAY cycle, particularly applicable when operating
	  with a short RTCP interval (possibly forced so server-side).

2011-08-24 14:37:52 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer/rtpbin: relax dropping rtcp packets
	  ... to at least having it trigger a/v synchronization, possibly without
	  using provided values which are still not considered sane
	  (as previously dropped).

2011-08-24 14:34:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: some more reset when clearing pt map
	  ... which in particular caters for some more reset following a possible
	  rtsp PLAY.

2011-08-21 21:58:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: do not set elements to PLAYING when doing seek in PAUSED

2011-09-01 14:47:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: only reset skew on gap if input ts available

2011-08-18 14:12:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: check some more for possible rtp timestamp discontinuity
	  ... when operating in non slave mode, and reset if detected.
	  This should avoid some (large) bogus outgoing timestamp due to jumps
	  in rtp time, as result of PAUSE/PLAY or seek or ...

2011-08-08 12:48:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: switch to rtp time based syncing when guessed appropriate

2011-08-08 12:15:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: alternative inter-stream syncing methods
	  ... at least if not syncing to NPT time:
	  * either sync using RTCP SR data (as currently)
	  * only perform the above once using initial RTCP SR packets
	  * discard RTCP and sync by equating provided stream's clock-base rtptime,
	  as provided by jitterbuffer (typically obtained from RTP-Info in RTSP).

2011-08-08 12:11:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: also provide clock-base to sync signal

2011-08-08 12:09:41 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: allow configurable rtcp stream syncing interval
	  ... rather than necessarily syncing at each RTCP SR.

2011-08-01 08:35:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: trigger reconsideration if rtcp interval set

2011-08-01 08:32:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: configure rtcp interval if provided
	  ... in PLAY response.

2011-09-16 16:53:22 +0300  Lasse Laukkanen <lasse.laukkanen@digia.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: Fix allowing zero duration tracks
	  https://bugzilla.gnome.org/show_bug.cgi?id=637486

2011-09-05 10:11:18 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/udp/gstudpnetutils.c:
	  udpsrc: error out when no protocol is specified in the uri
	  It is certainly better than to crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658178

2011-09-19 09:37:58 +0200  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/speex/gstspeexenc.c:
	  speexenc: do not use invalid buffer timestamps

2011-03-29 12:09:18 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/Makefile.am:
	* ext/pulse/plugin.c:
	* ext/pulse/pulseaudiosink.c:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.h:
	  pulse: New pulseaudiosink element to handle format changes
	  This introduces a new bin which wraps around pulsesink and depending on
	  the formats supported by the sink, plugs in/out a decodebin2 as
	  required. This allows users to switch sinks on the stream and adapts
	  accordingly (for example, you could watch a movie in passthrough mode on
	  your receiver which supports AC3 decode, then plug out and switch to a
	  non-digital profile to continue uninterrupted on analog output).
	  The bin is required because doing the same with playbin2/playsink will
	  require API changes that cannot be made in 0.10. With 0.11/1.0, we
	  should be able to ask for upstream caps renegotiation to deal with all
	  this.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657179

2011-09-16 15:03:23 +0200  Branko Subasic <branko@axis.com>

	* gst/matroska/ebml-read.c:
	* gst/matroska/ebml-read.h:
	* gst/matroska/matroska-read-common.c:
	  matroskademux: Avoid sending EOS when in paused state
	  Changed the ebml reader's gst_ebml_peek_id_length() function so
	  that it returns the actual reason for why the peek failed, instead
	  of (almost) always returning GST_FLOW_UNEXPECTED. This prevents
	  the pulling task from sending EOS when doing a flushing seek.

2011-09-15 15:53:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix stuttering A/V
	  Someone got had by implicit promotion to unsigned in ops with
	  a signed and an unsigned value.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659153

2011-09-14 16:37:12 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/debugutils/gstnavseek.c:
	  navseek: toggle pause/play on space bar
	  A useful thing to have.
	  https://bugzilla.gnome.org/show_bug.cgi?id=659065

2011-09-14 14:46:00 +0200  David Svensson Fors <davidsf@axis.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: configurable timestamp gap handling
	  matroskademux performs segment tricks to skip gaps in streams,
	  notably at start for non 0 based files.  There may however be
	  cases when full presentation (including intermediate gaps) is
	  desired, so a property allows to configure as of which gap
	  to act (or not at all).
	  API: GstMatroskaDemux::max-gap-time
	  Fixes #659009.

2011-09-12 09:21:47 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* tests/check/elements/flvmux.c:
	  tests: flvmux: Fix flvmux's tests after fix for request pads handling
	  Now that flvmux doesn't release its request pads on PAUSED->READY the
	  test doesn't need to re-request them for every reuse test start.

2011-09-09 09:12:56 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix ctts generation for streams that don't start at 0 timestamps
	  Subtract the first timestamp of a stream from all input buffers to
	  get 0-based timestamps for creating a sane ctts table. Without this
	  patch the ctts could have larger values than needed, causing the
	  playback to have a delay at startup.
	  As the first timestamp is only found after a few buffers are queued
	  (due to possible reordered buffers), once we find the first timestamp
	  we subtract it from all buffers on the queue, from that point on,
	  all buffers have their timestamps subtract when they are collected.
	  https://bugzilla.gnome.org/show_bug.cgi?id=658659

2011-09-12 07:55:19 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/flv/gstflvmux.c:
	  flvmux: don't release request pads going PAUSED->READY
	  Don't release request pads but just reset them. This makes pipelines using
	  flvmux reusable.

2011-09-09 12:35:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: use bsid 9 and 10 to control sample rate
	  See http://matroska.org/technical/specs/codecid/index.html
	  The spec is silent about this though...
	  https://bugzilla.gnome.org/show_bug.cgi?id=658546

2011-09-07 14:13:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: ensure some initial state variable setup
	  ... which might otherwise be skipped if the PLAY command is issued before
	  the OPEN command had a chance to actually be acted upon.
	  Fixes #657376.

2011-09-08 15:02:05 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: tweak gap handling
	  ... so as to avoid buffers before and after gap to have identical running time.

2011-09-08 13:28:24 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: use GST_RESOURCE_ERROR_BUSY if v4l2_ioctl fails with EBUSY
	  https://bugzilla.gnome.org/show_bug.cgi?id=658543

2011-09-07 08:54:17 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: remove one G_UNLIKELY for user property
	  Using G_UNLIKELY on user properties isn't nice, specially when
	  that is the default option.

2011-03-15 11:03:53 +0100  Andoni Morales Alastruey <amorales@flumotion.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: handle GstForceKeyUnit event
	  ... by starting a new cluster after forwarding event.
	  Fixes #644154.

2011-09-07 14:27:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/cmmldec.c:
	* tests/check/elements/cmmlenc.c:
	  cmml: Use complete cmml caps in the unit test

2011-09-07 14:26:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/elements/qtmux.c:
	  qtmux: Use complete MPEG caps in the unit test

2011-09-07 14:18:58 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	  docs: cleanup makefiles
	  Remove commented out parts that we don't need. Remove "the wingo addition" - no
	  so useful after all. Narrow down file-globs for plugin docs.

2011-08-29 14:12:22 +0200  Konstantin Miller <konstantin.miller@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Don't handle HTTP response 407 as error if proxy authentication data is available
	  Fixes bug #657422.

2011-09-07 12:11:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	  ac3parse: Add Converter to the classification because it can convert between different alignments
	  This allows decodebin2 to let it negotiate properly.

2011-09-07 12:10:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	  audioparsers: Improve src template caps
	  Remove the parsed/framed fields and add all fields to the template
	  caps that always exist.

2011-09-06 15:59:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	  aacparse: parse codec_data to determine number of samples per frame
	  Fixes #656734.

2011-09-06 21:24:46 +0200  Stefan Sauer <ensonic@users.sf.net>

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

2011-09-06 15:40:32 +0200  Stefan Sauer <ensonic@users.sf.net>

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

2011-09-06 15:05:37 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: make default duration check less sensitive
	  Frame duration might vary for 1 usecond, in this case matroskamux
	  decides to create BLOCKGROUP instead of SIMPLEBLOCK.
	  Convert duration to timecodescale which is (typically) less precise, and
	  then also allow the difference of 1/-1 to arrange for less sensitive check.
	  Based on patch by Alexey Fisher <bug-track@fisher-privat.net>
	  Fixes #653080.

2011-09-06 13:18:40 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: improve bogus interleaved index compensating
	  Patch by <gudake@gmail.com>
	  Fixes #654585.

2011-09-06 10:33:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Allow positive, non-1.0 segment rates
	  Only negative rates are not supported. Fixes bug #658305.

2011-09-05 15:50:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* tests/check/elements/parser.c:
	  tests: parsers: provide more real data when testing draining of garbage

2011-09-05 15:50:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/audioparsers/gstamrparse.c:
	  amrparse: fix and streamline valid frame checking
	  ... to handle various combinations of sync or not, and sufficient data
	  or not as might be expected.
	  Fixes #650714.

2011-09-05 14:49:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fragmented support; avoid adjustment for keyframe seek
	  ... since all index data may not yet be available at that time.

2011-09-05 14:48:02 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fragmented support; mark all audio track samples as keyframe

2011-09-05 14:46:29 +0200  Brian Li <brian7003@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fragmented support; properly init return variable value
	  Fixes #655918.

2011-09-05 13:31:20 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add gtk-doc for new short-header property

2011-09-05 13:18:39 +0200  Marc Leeman <marc.leeman@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: allow sending short RTSP requests to a server
	  Some encoders (Arecont) do not like the long OPTIONS sent at startup as sent by
	  GStreamer, but do accept the short header as sent by Live555.
	  This patch makes the extending the request optional by adding a property
	  (short-header).
	  Fixes #655805.
	  API: GstRTSPSrc:short-header

2009-03-04 14:51:09 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Set H263-2000 if thats what the other side wants
	  The static caps states this element supports H263-2000, but setcaps never
	  sets it, so it was lie.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=577784

2011-08-30 19:02:51 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Initialise the last_keyframe_request variable

2011-08-31 16:04:24 +0200  Peter Korsgaard <jacmet@sunsite.dk>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: make add/remove/clear/get-stats action signals
	  http://bugzilla.gnome.org/show_bug.cgi?id=657830
	  Signed-off-by: Peter Korsgaard <jacmet@sunsite.dk>

2011-08-30 13:33:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: push mode; perform some extra checks prior to upstream seeking

2011-08-30 13:28:21 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: push mode; fix buffered streaming
	  That is, in case where no seek is peformed to moov, but preceding
	  limited mdat is buffered.

2011-08-29 15:13:56 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid overflow wraparound in timestamp when adding durations
	  Do some type juggling to avoid overflow, while still allowing for 'negative'
	  durations (which would need a wraparound effect).

2011-08-25 23:37:47 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: make this work more than once in a row
	  We used to skip frame rate setup if the camera was already setup
	  with the requested frame rate. This breaks some cameras though,
	  causing them to not output data (several models of Thinkpad cameras
	  have this problem at least).
	  So, don't skip.
	  https://bugzilla.gnome.org/show_bug.cgi?id=638300

2011-08-23 12:12:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: only require two frames in a row when we do not have sync
	  This avoids a single bit error dropping two frames unnecessarily.
	  The two consecutive frames check is still required when we don't
	  have sync.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657080

2011-08-23 21:41:15 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Trivial indentation fix

2011-07-21 17:23:28 -0400  Monty Montgomery <cmontgom@redhat.com>

	* ext/flac/gstflacdec.c:
	  flacdec: Correct sample number rounding resulting in timestamp jitter
	  flacdec converts the src timestamp to a sample number, uses that internally, then reconverts the sample number to a timestamp for the output buffer.  Unfortunately, sample numbers can't be represented in an integer number of nanoseconds, and the conversion process was truncating rather than rounding, resulting in sample numbers and output timestamps that were often off by a full sample.
	  This corrects the time->sample convesion

2011-08-20 14:48:20 -0700  David Schleef <ds@schleef.org>

	* gst/debugutils/breakmydata.c:
	  breakmydata: element is not passthrough

2011-07-13 11:20:34 -0700  David Schleef <ds@schleef.org>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: quiet debugging

2011-07-10 21:40:20 -0700  David Schleef <ds@schleef.org>

	* gst/deinterlace/gstdeinterlace.c:
	* gst/deinterlace/gstdeinterlace.h:
	* gst/deinterlace/gstdeinterlacemethod.c:
	* gst/deinterlace/gstdeinterlacemethod.h:
	* gst/deinterlace/tvtime/greedy.c:
	* gst/deinterlace/tvtime/greedyh.c:
	* gst/deinterlace/tvtime/linearblend.c:
	* gst/deinterlace/tvtime/scalerbob.c:
	* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
	* gst/deinterlace/tvtime/vfir.c:
	* gst/deinterlace/tvtime/weave.c:
	* gst/deinterlace/tvtime/weavebff.c:
	* gst/deinterlace/tvtime/weavetff.c:
	  deinterlace: change field handling through methods
	  This likely breaks stuff.  The good: all of the methods now create
	  field images aligned with input frames, without timestamp mangling.
	  The bad: this touches a lot of code, much of which is hairy and in
	  need of cleanup.  However, at this point we can reasonably create a
	  PSNR-based test.

2011-08-21 14:41:14 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: reset ->streamheaders to NULL on _stop
	  Fixes invalid memory access reusing multifilesink

2011-08-18 13:37:39 +0200  David Henningsson <david.henningsson@canonical.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Allow writes in bigger chunks
	  There's no use in splitting the incoming data down to the segsize
	  limit - by writing as much as possible in one chunk, we increase
	  performance and avoid PulseAudio unnecessary rewinds.
	  Signed-off-by: David Henningsson <david.henningsson@canonical.com>

2011-08-08 22:14:28 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/matroska/matroska-demux.c:
	  matroskademux: ensure no-more-pads is always emitted
	  In particular, do so even if failing to read while prerolling,
	  such as when reading from a partial file (eg, while it is being
	  downloaded).
	  This fixes a wedge in playbin2.
	  https://bugzilla.gnome.org/show_bug.cgi?id=651965

2011-08-16 17:27:13 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: avoid timestamp/offset tracking going out of sync
	  The libFLAC API is callback based, and we must only call it to
	  output data when we know we have enough input data. For this
	  reason, a single processing step is done when receiving a buffer.
	  However, if there were metadata buffers still pending, a step
	  intended for the first audio frame might end up writing that
	  leftover metadata. Since a single step is done per buffer, this
	  will cause every buffer to be written one step late.
	  This would add some latency (a bufferfull's worth), possibly
	  lose a buffer when seeking or the like, and also cause timestamp
	  and offset to be applied to the wrong buffer, as updates to
	  the "current" segment last_stop (from incoming buffer timestamp)
	  will be applied to an output buffer originating from the previous
	  incoming buffer.
	  This fixes the issue by ensuring that, upon receiving the first
	  audio frame, processing is done till all metadata is processed,
	  so the next "single step" done will be for the audio frame. After
	  this, we should keep to 1 input buffer -> 1 output buffer and so
	  avoid getting out of sync.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650960

2011-08-16 15:32:07 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: bail on reserved value
	  Now that we look at the right bits, we can test against the reserved
	  value as we do for other fields.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650960

2011-08-16 15:27:43 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: fix bit twiddling
	  Right shifting a 8 bit value by 8 bits is twice too much
	  to get the high 4 bits.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650960

2011-08-16 15:22:46 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/flac/gstflacdec.c:
	  flacdec: warn if we see a variable block size where unsupported
	  https://bugzilla.gnome.org/show_bug.cgi?id=650960

2011-08-16 18:25:29 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/spectrum/gstspectrum.c:
	  spectrum: avoid crashing by resetting the correct number of channels
	  https://bugzilla.gnome.org/show_bug.cgi?id=656606

2011-08-16 13:16:22 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix off by one in frame size check
	  Yes, I was tracking another bug and the small test file I generated
	  to test with improbably just happened to trigger this, with a second
	  and last frame of 1615 bytes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=656649

2011-08-14 20:46:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/id3demux/id3v2.3.0.html:
	* gst/id3demux/id3v2.4.0-frames.txt:
	* gst/id3demux/id3v2.4.0-structure.txt:
	  id3demux: remove specs from git as well now that parsing code is in -base

2011-07-14 15:42:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* configure.ac:
	* gst/id3demux/Makefile.am:
	* gst/id3demux/gstid3demux.c:
	* gst/id3demux/id3tags.c:
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c:
	  id3demux: use -base provided id3 tag parsing
	  https://bugzilla.gnome.org/show_bug.cgi?id=654388

2011-08-13 16:51:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/jack/gstjackaudiosrc.c:
	  jackaudiosrc: fix error message code
	  And also post 'not found' error if jackd is not even installed.

2011-08-12 16:32:58 +0200  Stefan Kost <ensonic@users.sf.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: initialize bitrate variable and reset for each loop
	  Don't check eventually unset variable and don't accidentially use values from last
	  cycle.

2011-08-09 11:28:17 +0200  Edward Hervey <edward.hervey@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Properly error out if SDP contains no streams
	  Also fixes unitialized variable error on macosx.

2011-08-09 09:05:31 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: clear flags on buffer reuse
	  This will ensure a logically new buffer does not keep flags from
	  a previous use of that buffer (eg, DISCONT would be set on the first
	  buffer, and mistakenly kept when reused).
	  https://bugzilla.gnome.org/show_bug.cgi?id=653709

2011-08-08 10:54:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: take care not to change the current format where appropriate
	  Some drivers are buggy are will change the current format when
	  processing VIDIOC_TRY_FMT. Save and restore the current format
	  to ensure the format is kept unchanged.
	  https://bugzilla.gnome.org/show_bug.cgi?id=649067

2011-08-07 12:23:26 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/v4l2src_calls.c:
	  v4l2src: Use fraction compare util function.
	  Use the fraction compare utility to compare function, not the
	  handcrafted one. The handcrafted one is buggy as it doesn't take into
	  account rounding error. For example comparing a framerate of 20/1 on a
	  camera configured as 30/1 fps would yield true: 1 == (1 * 20)/30 and not
	  re-configure the camera. Fixes #656104

2011-08-03 22:50:05 +1000  Jan Schmidt <thaytan@noraisin.net>

	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	* gst/matroska/matroska.c:
	  matroska: Register new debug category
	  Register the matroskareadcommon debug category when the
	  plugin is loaded to avoid assertion output when debug is turned on.

2011-07-29 13:03:55 +0200  Philippe Normand <pnormand@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: soften assertion check on stream size
	  https://bugzilla.gnome.org/show_bug.cgi?id=655570

2011-08-03 10:09:42 +0200  Robert Krakora <rob.krakora@messagenetsystems.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtpjpegpay: Add support for H.264 payload in MJPEG container
	  See http://www.quickcamteam.net/uvc-h264/USB_Video_Payload_H.264_0.87.pdf
	  Fixes bug #655530.

2011-08-02 22:05:08 -0400  Tristan Matthews <tristan@sat.qc.ca>

	* ext/jack/gstjackaudiosink.c:
	* ext/jack/gstjackaudiosink.h:
	  jackaudiosink: Don't call g_alloca() in process_cb
	  g_alloca() is not RT-safe, so instead we should allocate the
	  memory needed in advance. Fixes #655866

2011-08-02 23:42:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/multipart/multipartdemux.c:
	* sys/v4l2/gstv4l2object.c:
	  docs: fix two more Since: tags

2011-07-31 04:19:00 +0300  Mart Raudsepp <leio@gentoo.org>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Fix Since tags for fieldanalysis related new properties
	  commit c1b100cf9c is after 0.10.29 and 0.10.30 was a branched release.
	  So fix Since tags from 0.10.29 to 0.10.31 for the new properties.

2011-07-29 13:05:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: fix variable-set-but-not-used compiler warning with older pulse versions

2011-07-29 12:07:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: properly init rtcp_min_interval

2011-03-09 11:04:36 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.c:
	  pulsesink: Add support for compressed formats
	  This adds support for various compressed formats (AC3, E-AC3, DTS and
	  MP3) payloaded in IEC 61937 format (used for transmission over S/PDIF,
	  HDMI and Bluetooth).
	  The acceptcaps() function allows bins to probe for what formats the sink
	  being connected to support. This only works after the element is set to
	  at least READY.
	  If the underlying sink changes and the format we are streaming is not
	  available, we emit a message that will allow upstream elements/bins to
	  block and renegotiate a new format.

2011-03-01 15:34:46 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulsesink: Use the extended stream API if available
	  This uses the new extended API for creating streams. This will allow us
	  to support compressed formats natively in pulsesink as well.

2011-07-29 00:07:52 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: Add a source-output-index property
	  This exposes the source output index of the record stream that we open
	  so that clients can use this with the introspection if they want (to
	  move the stream, for example).

2011-07-28 14:44:57 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: keep a ref on the src pad while using it
	  Prevent a possible race if clear_ssrc() is called between getting the pad and
	  doing the push.
	  Based on patch by <olivier.crete@collabora.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=650916

2011-05-24 11:29:57 +0300  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.h:
	  rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit
	  We need to keep the lock held because we don't want a push before the "new-ssrc-pad"
	  handler has completed. But we may want to push an event from inside that handler, hence
	  the recursive mutex.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650916

2011-05-24 11:17:25 +0300  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpssrcdemux: Use PADs lock
	  https://bugzilla.gnome.org/show_bug.cgi?id=650916

2011-07-27 18:15:20 +0100  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: Cope with FU-A E bit not being set
	  Some h264 payloaders are unfortunately buggy and don't correctly set the
	  E bit in FU-A NAL when they have ended. Work around this by assuming
	  such a fragmentation unit has ended when there was no packet loss and a
	  new NAL is started

2011-04-12 17:01:47 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	  ac3parse: Support switching alignment on-the-fly
	  This allows switching of alignment for E-AC3 streams at run-time. This
	  is requested by downstream elements via a custom event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650313

2011-04-09 12:26:56 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	* tests/check/elements/ac3parse.c:
	  ac3parse: Add support for IEC 61937 alignment
	  When pushing out buffers over S/PDIF or HDMI, IEC 61937 payloading
	  requires each buffer to contain 6 blocks from each substream. This adds
	  code to collect all the frames needed to meet this requirement before
	  pushing out a buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=650313

2011-06-08 15:57:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Always send application requested feedback in immediate mode
	  Send as many application requested feedback messages in immediate mode, even if they
	  have already been sent.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654583

2011-06-08 14:48:01 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Don't let the computed RTP bandwidth fall too low
	  If it falls too low, the computed RTCP bandwidth will be near zero and
	  the RTCP thread will be stopped.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654583

2011-04-25 16:13:38 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Wait longer to timeout SSRC collision
	  Using the current RTCP interval to timeout SSRC collision can lead to
	  collisions being timed out immediately if a BYE packet is sent because
	  it is sent immediately, so the interval is 0. This is not what we
	  want. So just set a static 10 times the default RTCP interval, it
	  should be enough
	  https://bugzilla.gnome.org/show_bug.cgi?id=648642

2011-07-19 13:38:01 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: set SOURCE flag at init time
	  Fixes #654816.

2011-07-18 16:46:27 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Complete merged AU on marker bit
	  The marker bit on a RTP packet means the AU has been completed, so push it out
	  immediately to reduce the latency.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654850

2011-07-18 20:27:38 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Only set the marker bit on the last NALU of a multi-NALU access unit
	  An access unit could contain multiple NAL units, in that case, only the last
	  RTP packet of the last NALU should have its marker bit set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654850

2011-07-20 08:52:58 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multipart/multipartmux.c:
	  multipart: fix compiler warning

2011-07-19 12:05:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/auparse/gstauparse.c:
	  auparse: avoid hanging on invalid short input
	  ... as in such case there is no srcpad yet on which to forward EOS.

2011-07-18 15:13:33 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Fix default value leaking
	  Remember to free the default value of client name, avoiding a
	  leak

2011-07-18 14:24:48 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: reset upon FLUSH_STOP
	  ... which is particularly needed when merging NAL units, where not resetting
	  would lead to output of an older (pre-flush) AU (with unintended timestamp).

2011-07-18 14:30:51 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: do not use g_slist_free_full
	  ... as that is only in GLib 2.28, which is not yet required at this time.

2011-07-18 09:38:26 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/gstmultifilesink.c:
	* gst/multifile/gstmultifilesink.h:
	* tests/check/elements/multifile.c:
	  multifilesink: add max-files property
	  Add max-files property to limit the number of files saved on disk.
	  API: multifilesink::max-files

2011-07-17 23:36:55 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: refactor file opening and closing code

2011-07-16 19:38:51 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix pixel-aspect-ratio if header has only one display variable
	  Current matroska demux calculates the pixel aspect ratio only if both
	  DisplayHeight and DisplayWidth are set, but it is legal to use only
	  one variable if the other is equal to PixelWidth or PixelHeight, at
	  least the mkclean utility is doing that. So this makse mkcleaned
	  files play correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654744

2011-07-16 23:47:50 +0100  Antoine Jacoutot <ajacoutot@openbsd.org>

	* gst/goom/plugin_info.c:
	  goom: fix build on PPC on openbsd
	  A missing sys/param.h include results in:
	  /usr/include/sys/proc.h:64: error: 'MAXLOGNAME' undeclared here (not in a
	  function)
	  /usr/include/sys/proc.h:285: error: 'MAXCOMLEN' undeclared here (not in a
	  function)
	  when compiling goom on openbsd/ppc. We can just remove the two sys/ includes
	  here, they are not needed for anything.
	  https://bugzilla.gnome.org/show_bug.cgi?id=654749

2011-07-14 20:10:02 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	  rtppcmApay/depay: Static clock rates on static payloads, dynamic on dynamic
	  Partially reverts 397dc60b

2011-03-04 15:41:22 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Implement getcaps
	  Convert profile-level-id from RTP caps into video/x-h264 style caps (with profile and level)

2011-07-12 15:04:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix seeking regression
	  ... introduced when shuffling around code for the async implementation
	  by setting state of source (and udp sources) in _play before downstream
	  flushing is undone.

2011-07-11 15:23:41 +0300  René Stadler <rene.stadler@nokia.com>

	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstac3parse.h:
	  ac3parse: fix buffer duration on blocks-per-frame change
	  The gst_base_parse_set_frame_rate call was predicated on a change to
	  sample rate, duration or profile. However, the block count per frame can
	  also change between packets, which would result in incorrect buffer
	  durations.

2011-07-09 19:23:41 -0700  David Schleef <ds@schleef.org>

	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstmultifilesrc.h:
	  multifilesrc: Improve looping
	  Add start-index and stop-index properties.

2011-06-16 13:57:03 +0100  Jonny Lamb <jonnylamb@jonnylamb.com>

	* gst/multifile/gstmultifilesrc.c:
	* gst/multifile/gstmultifilesrc.h:
	  multifile: add loop property to multifilesrc
	  Fixes: #652727
	  Signed-off-by: Jonny Lamb <jonnylamb@jonnylamb.com>
	  Signed-off-by: David Schleef <ds@schleef.org>

2009-11-20 10:07:43 +0100  Philip Jägenstedt <philipj@opera.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: 16-bit audio is signed, 8-bit is unsigned.
	  Pretending to handle 8-bit signed causes distorted audio when
	  actually given such audio, which you will get if passing 8-bit
	  unsigned through audioconvert ! audioresample, as audioresample
	  only handles 8-bit signed.  Fixes #605834.
	  Signed-off-by: David Schleef <ds@schleef.org>

2011-07-07 18:27:36 +0200  Alexey Fisher <bug-track@fisher-privat.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: handle blocks with duration=0
	  Some video frames, for example alt-ref frame in VP8, will be
	  never displayed. This is why it has duration=0.
	  This patch allow to use this duration.
	  Bug: 654175
	  Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>

2011-07-06 17:18:05 -0700  David Schleef <ds@schleef.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Add direct dirac mapping

2011-06-29 20:59:26 +0300  René Stadler <rene.stadler@nokia.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: prevent race condition causing ref leak
	  Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the
	  deferred call to be run before returning. This causes a race when
	  READY->NULL is executed shortly after, which stops the mainloop. This
	  leaks the element reference which is passed as userdata for the callback
	  (introduced in commit 7cf996, bug #614765).
	  The correct fix is to wait in READY->NULL for all outstanding calls to
	  be fired (since libpulse doesn't provide a DestroyNotify for the
	  userdata). We get rid of the reference passing from 7cf996 altogether,
	  since finalization from the callback would anyways lead to a deadlock.
	  Re-fixes bug #614765.

2011-07-04 08:58:14 +0300  René Stadler <rene.stadler@nokia.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: small cleanup of copy-paste code

2011-06-29 19:50:42 +0300  René Stadler <rene.stadler@nokia.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: remove unused member variable and misleading log message
	  Wim changed it in commit 8bfd80 so that pa_defer_ran is not read
	  anywhere.
	  The log message used to annotate a mainloop_wait call which is gone.

2011-07-04 12:58:38 -0700  David Schleef <ds@schleef.org>

	* gst/goom/gstgoom.c:
	  goom: Don't answer lantency queries before negotiation

2011-07-04 14:30:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: avoid crashing on invalid input without components

2011-07-04 11:25:28 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: pass along segment info to collectpads
	  ... so it can track this and be subsequently used to determine running time etc.

2011-07-04 11:24:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/flv/gstflvdemux.c:
	  flvdemux: indicate raw format in aac caps

2011-07-03 19:51:32 -0700  David Schleef <ds@schleef.org>

	* ext/pulse/plugin.c:
	  pulse: Increase ranks to PRIMARY + 10
	  So that pulsesrc/pulsesink get chosen over other possible PRIMARY
	  src/sinks by autoaudiosink.  Presumably, if pulse is available, it
	  is always preferred over another src/sink.
	  Fixes: #647540.

2011-06-30 18:47:48 -0700  David Schleef <ds@schleef.org>

	* gst/multipart/multipartmux.c:
	  multipartmux: Add \r\n to tail of pushed buffers
	  Clients such as Firefox require the \r\n after the payload.

2011-06-16 14:52:51 +0200  Branko Subasic <branko@axis.com>

	* gst/matroska/ebml-read.c:
	* gst/matroska/matroska-demux.c:
	  matroskademux: avoid looping when searching for clusters
	  Fixes some bugs that results in the demuxer looping when seaching
	  for clusters in non-finalized files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=652195

2011-06-10 18:54:48 +0530  Debarshi Ray <rishi@gnu.org>

	* gst/matroska/matroska-parse.c:
	  matroskaparse: fix reference counting of parse->streamheader
	  https://bugzilla.gnome.org/show_bug.cgi?id=652286
	  Signed-off-by: David Schleef <ds@schleef.org>

2011-06-29 14:39:52 -0700  David Schleef <ds@schleef.org>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Don't round up size of encoded buffers
	  For some reason, in code dating to 2001, encoded jpeg buffers were
	  rounded up to multiples of 4 bytes.  With the added bonus that the
	  extra bytes are unwritten, causing valgrind issues.  Oops.  I can't
	  think of any reason why JPEG buffers need to be multiples of 4 bytes,
	  so I removed the padding.  There might be some code somewhere that
	  depends on this behavior, so if this needs to be reverted, please fix
	  the valgrind issues.

2011-06-29 12:05:04 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: free date tag

2011-06-28 12:26:37 +0200  Jonas Larsson <jonas.larsson@hiq.se>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: not so greedy minimum frame size
	  Fixes #653559.

2011-06-25 11:39:23 -0700  David Schleef <ds@schleef.org>

	* configure.ac:
	  configure: remove non-pkg-config check for shout
	  Fixes: 653327

2011-06-20 18:49:57 +0200  Andoni Morales Alastruey <amorales@flumotion.com>

	* ext/raw1394/gst1394clock.c:
	  dv1394src: make the internal clock thread safe
	  Fixes: #653091.

2011-06-24 11:54:29 +0200  Miguel Angel Cabrera Moya <madmac2501@gmail.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: return correct type when assertion fails

2011-06-23 11:28:27 -0700  David Schleef <ds@schleef.org>

	* common:
	  Automatic update of common submodule
	  From 69b981f to 605cd9a

2011-02-02 16:18:54 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulse: Drop support for PA versions before 0.9.16
	  This drops support fof PulseAudio versions prior to 0.9.16, which was
	  released about 1.5 years ago. Testing with very old versions is not
	  feasible and we don't want to maintain 2 independent code-paths.

2011-06-21 15:15:06 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

	* gst/rtp/gstrtpmp4adepay.c:
	  rtpmp4adepay: fix output buffer timestamps in case of multiple frames

2011-06-20 16:47:36 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: The signal has 5 arguments, not 4

2011-06-18 13:43:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	  Bump git version after unplanned 0.10.30 release
	  Merge branch '0.10.30'
	  Conflicts:
	  configure.ac
	  docs/plugins/inspect/plugin-1394.xml
	  docs/plugins/inspect/plugin-aasink.xml
	  docs/plugins/inspect/plugin-alaw.xml
	  docs/plugins/inspect/plugin-alpha.xml
	  docs/plugins/inspect/plugin-alphacolor.xml
	  docs/plugins/inspect/plugin-annodex.xml
	  docs/plugins/inspect/plugin-apetag.xml
	  docs/plugins/inspect/plugin-audiofx.xml
	  docs/plugins/inspect/plugin-audioparsers.xml
	  docs/plugins/inspect/plugin-auparse.xml
	  docs/plugins/inspect/plugin-autodetect.xml
	  docs/plugins/inspect/plugin-avi.xml
	  docs/plugins/inspect/plugin-cacasink.xml
	  docs/plugins/inspect/plugin-cairo.xml
	  docs/plugins/inspect/plugin-cutter.xml
	  docs/plugins/inspect/plugin-debug.xml
	  docs/plugins/inspect/plugin-deinterlace.xml
	  docs/plugins/inspect/plugin-dv.xml
	  docs/plugins/inspect/plugin-efence.xml
	  docs/plugins/inspect/plugin-effectv.xml
	  docs/plugins/inspect/plugin-equalizer.xml
	  docs/plugins/inspect/plugin-esdsink.xml
	  docs/plugins/inspect/plugin-flac.xml
	  docs/plugins/inspect/plugin-flv.xml
	  docs/plugins/inspect/plugin-flxdec.xml
	  docs/plugins/inspect/plugin-gconfelements.xml
	  docs/plugins/inspect/plugin-gdkpixbuf.xml
	  docs/plugins/inspect/plugin-goom.xml
	  docs/plugins/inspect/plugin-goom2k1.xml
	  docs/plugins/inspect/plugin-gstrtpmanager.xml
	  docs/plugins/inspect/plugin-halelements.xml
	  docs/plugins/inspect/plugin-icydemux.xml
	  docs/plugins/inspect/plugin-id3demux.xml
	  docs/plugins/inspect/plugin-imagefreeze.xml
	  docs/plugins/inspect/plugin-interleave.xml
	  docs/plugins/inspect/plugin-isomp4.xml
	  docs/plugins/inspect/plugin-jack.xml
	  docs/plugins/inspect/plugin-jpeg.xml
	  docs/plugins/inspect/plugin-level.xml
	  docs/plugins/inspect/plugin-matroska.xml
	  docs/plugins/inspect/plugin-mulaw.xml
	  docs/plugins/inspect/plugin-multifile.xml
	  docs/plugins/inspect/plugin-multipart.xml
	  docs/plugins/inspect/plugin-navigationtest.xml
	  docs/plugins/inspect/plugin-oss4.xml
	  docs/plugins/inspect/plugin-ossaudio.xml
	  docs/plugins/inspect/plugin-png.xml
	  docs/plugins/inspect/plugin-pulseaudio.xml
	  docs/plugins/inspect/plugin-replaygain.xml
	  docs/plugins/inspect/plugin-rtp.xml
	  docs/plugins/inspect/plugin-rtsp.xml
	  docs/plugins/inspect/plugin-shapewipe.xml
	  docs/plugins/inspect/plugin-shout2send.xml
	  docs/plugins/inspect/plugin-smpte.xml
	  docs/plugins/inspect/plugin-soup.xml
	  docs/plugins/inspect/plugin-spectrum.xml
	  docs/plugins/inspect/plugin-speex.xml
	  docs/plugins/inspect/plugin-taglib.xml
	  docs/plugins/inspect/plugin-udp.xml
	  docs/plugins/inspect/plugin-video4linux2.xml
	  docs/plugins/inspect/plugin-videobox.xml
	  docs/plugins/inspect/plugin-videocrop.xml
	  docs/plugins/inspect/plugin-videofilter.xml
	  docs/plugins/inspect/plugin-videomixer.xml
	  docs/plugins/inspect/plugin-wavenc.xml
	  docs/plugins/inspect/plugin-wavpack.xml
	  docs/plugins/inspect/plugin-wavparse.xml
	  docs/plugins/inspect/plugin-ximagesrc.xml
	  docs/plugins/inspect/plugin-y4menc.xml
	  win32/common/config.h

2011-06-17 10:37:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosink.h:
	  sunaudio: fix typo in comment

2011-06-17 03:07:09 +0300  Stefan Kost <ensonic@users.sf.net>

	* gst/audiofx/audioecho.c:
	  audioecho: fix param flags
	  If the parameter cannot be changed in paused&playing, it is not controlable. Set
	  the appropriate mutability flag instead.