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

=== release 0.10.31 ===

2012-02-21 Tim-Philipp Müller <[email protected]>

* configure.ac:
releasing 0.10.31, "Faster"

2012-02-20 12:22:12 -0500 Olivier Crête <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2012-02-09 22:12:14 +0100 Mark Nauwelaerts <[email protected]>

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

2012-02-09 10:11:48 +0100 Marc Leeman <[email protected]>

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

2012-02-07 14:10:44 -0800 Ralph Giles <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2012-02-03 22:05:59 +0530 Arun Raghavan <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2012-01-25 15:21:44 +0000 Jayakrishnan M <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2012-01-20 17:02:15 +0100 Mark Nauwelaerts <[email protected]>

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

2012-01-20 17:01:37 +0100 Mark Nauwelaerts <[email protected]>

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

2012-01-20 16:58:28 +0100 Mark Nauwelaerts <[email protected]>

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

2012-01-20 16:57:52 +0100 Mark Nauwelaerts <[email protected]>

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

2012-01-20 16:55:06 +0100 Mark Nauwelaerts <[email protected]>

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

2012-01-20 16:49:14 +0100 Mark Nauwelaerts <[email protected]>

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

2012-01-20 16:48:49 +0100 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

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

2012-01-20 16:46:21 +0100 Mark Nauwelaerts <[email protected]>

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

2012-01-20 16:44:21 +0100 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2012-01-16 16:10:08 +0000 Vincent Penquerc'h <[email protected]>

* 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 <[email protected]>

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

2012-01-16 12:13:50 +0000 Vincent Penquerc'h <[email protected]>

* 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 <[email protected]>

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

2012-01-13 17:25:59 +0000 Vincent Penquerc'h <[email protected]>

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

2012-01-13 15:57:20 +0000 Vincent Penquerc'h <[email protected]>

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

2012-01-13 10:32:59 +0000 Tim-Philipp Müller <[email protected]>

* gst/rtpmanager/gstrtpptdemux.c:
rtpptdemux: plug pad leak in error code path
Based on patch by: Stig Sandnes <[email protected]>
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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2012-01-12 14:19:22 +0000 Vincent Penquerc'h <[email protected]>

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

2012-01-12 10:30:11 +0000 Vincent Penquerc'h <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2012-01-05 19:08:03 +0000 Vincent Penquerc'h <[email protected]>

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

2012-01-04 19:40:14 +0000 Tim-Philipp Müller <[email protected]>

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

2012-01-04 17:59:55 +0000 Tim-Philipp Müller <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

2011-12-21 20:50:21 +0100 Nicola Murino <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-12-13 17:02:01 +0100 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

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

2011-12-12 15:15:46 +0100 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

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

2011-12-12 10:38:20 +0000 Tim-Philipp Müller <[email protected]>

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

2011-12-12 10:18:14 +0000 Tim-Philipp Müller <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-12-10 15:13:07 +0000 Tim-Philipp Müller <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-12-10 12:32:32 +0000 Tim-Philipp Müller <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-11-24 13:58:01 +0100 Sebastian Rasmussen <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-12-01 13:22:42 +0000 Tim-Philipp Müller <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-12-01 13:34:52 +0100 Sebastian Dröge <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

2011-11-30 22:13:11 +0100 Stefan Sauer <[email protected]>

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

2011-11-30 19:25:37 +0000 Tim-Philipp Müller <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-11-26 16:06:59 +0000 Tim-Philipp Müller <[email protected]>

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

2011-11-25 19:28:55 -0300 Thiago Santos <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-11-14 12:43:27 +0530 Arun Raghavan <[email protected]>

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

2011-11-14 12:41:41 +0530 Arun Raghavan <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-11-25 10:15:35 +0100 Stefan Sauer <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-11-24 09:48:33 +0100 Sebastian Dröge <[email protected]>

* 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 <[email protected]>

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

2011-11-24 09:43:14 +0100 Sebastian Dröge <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-11-16 18:57:38 +0100 Mark Nauwelaerts <[email protected]>

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

2011-11-16 18:57:21 +0100 Mark Nauwelaerts <[email protected]>

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

2011-11-14 15:34:57 +0000 Vincent Penquerc'h <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-11-11 14:55:48 +0100 Mark Nauwelaerts <[email protected]>

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

2011-11-11 18:05:35 +0530 Arun Raghavan <[email protected]>

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

2011-11-08 15:35:26 +0000 Vincent Penquerc'h <[email protected]>

* 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 <[email protected]>

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

2011-11-02 12:58:12 -0400 Olivier Crête <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-10-07 19:41:35 +0100 Vincent Penquerc'h <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-10-11 20:56:51 +0400 Stas Sergeev <[email protected]>

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

2011-10-06 13:04:54 +0200 Mark Nauwelaerts <[email protected]>

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

2011-10-28 10:40:36 +0200 Sebastian Dröge <[email protected]>

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

2011-10-28 09:36:17 +0200 Sebastian Dröge <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-10-21 22:32:38 +0200 René Stadler <[email protected]>

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

2011-10-16 20:30:25 +0200 René Stadler <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-10-18 12:05:01 +0200 Wim Taymans <[email protected]>

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

2011-10-18 11:58:57 +0200 Wim Taymans <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-10-13 16:59:50 +0530 Arun Raghavan <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-10-10 13:20:04 +0200 Stefan Sauer <[email protected]>

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

2011-10-09 21:31:27 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-09-29 17:21:22 +0200 Mark Nauwelaerts <[email protected]>

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

2011-09-29 16:33:01 +0200 Mark Nauwelaerts <[email protected]>

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

2011-09-29 17:32:23 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-09-02 13:41:41 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-08-18 14:12:21 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

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

2011-08-08 12:15:20 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

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

2011-08-08 12:09:41 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

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

2011-08-01 08:32:24 +0200 Mark Nauwelaerts <[email protected]>

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

2011-09-16 16:53:22 +0300 Lasse Laukkanen <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-03-29 12:09:18 +0530 Arun Raghavan <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-09-07 14:18:58 +0200 Stefan Sauer <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-09-06 15:40:32 +0200 Stefan Sauer <[email protected]>

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

2011-09-06 15:05:37 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>
Fixes #653080.

2011-09-06 13:18:40 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtpmp4gdepay.c:
rtpmp4gdepay: improve bogus interleaved index compensating
Patch by <[email protected]>
Fixes #654585.

2011-09-06 10:33:21 +0200 Sebastian Dröge <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-09-05 14:46:29 +0200 Brian Li <[email protected]>

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

2011-09-05 13:31:20 +0200 Mark Nauwelaerts <[email protected]>

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

2011-09-05 13:18:39 +0200 Marc Leeman <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-08-31 16:04:24 +0200 Peter Korsgaard <[email protected]>

* 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 <[email protected]>

2011-08-30 13:33:49 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-07-21 17:23:28 -0400 Monty Montgomery <[email protected]>

* 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 <[email protected]>

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

2011-07-13 11:20:34 -0700 David Schleef <[email protected]>

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

2011-07-10 21:40:20 -0700 David Schleef <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

2011-08-08 22:14:28 +0100 Vincent Penquerc'h <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-07-31 04:19:00 +0300 Mart Raudsepp <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-03-09 11:04:36 +0530 Arun Raghavan <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>
https://bugzilla.gnome.org/show_bug.cgi?id=650916

2011-05-24 11:29:57 +0300 Olivier Crête <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-07-18 16:46:27 -0400 Olivier Crête <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-07-19 12:05:51 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

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

2011-07-16 19:38:51 +0200 Alexey Fisher <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstmultifilesrc.h:
multifile: add loop property to multifilesrc
Fixes: #652727
Signed-off-by: Jonny Lamb <[email protected]>
Signed-off-by: David Schleef <[email protected]>

2009-11-20 10:07:43 +0100 Philip Jägenstedt <[email protected]>

* 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 <[email protected]>

2011-07-07 18:27:36 +0200 Alexey Fisher <[email protected]>

* 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 <[email protected]>

2011-07-06 17:18:05 -0700 David Schleef <[email protected]>

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

2011-06-29 20:59:26 +0300 René Stadler <[email protected]>

* 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 <[email protected]>

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

2011-06-29 19:50:42 +0300 René Stadler <[email protected]>

* 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 <[email protected]>

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

2011-07-04 14:30:09 +0200 Mark Nauwelaerts <[email protected]>

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

2011-07-04 11:25:28 +0200 Mark Nauwelaerts <[email protected]>

* 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 <[email protected]>

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

2011-07-03 19:51:32 -0700 David Schleef <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

* 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 <[email protected]>

2011-06-29 14:39:52 -0700 David Schleef <[email protected]>

* 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 <[email protected]>

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

2011-06-28 12:26:37 +0200 Jonas Larsson <[email protected]>

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

2011-06-25 11:39:23 -0700 David Schleef <[email protected]>

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

2011-06-20 18:49:57 +0200 Andoni Morales Alastruey <[email protected]>

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

2011-06-24 11:54:29 +0200 Miguel Angel Cabrera Moya <[email protected]>

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

2011-06-23 11:28:27 -0700 David Schleef <[email protected]>

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

2011-02-02 16:18:54 +0530 Arun Raghavan <[email protected]>

* 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 <[email protected]>

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

2011-06-20 16:47:36 -0400 Olivier Crête <[email protected]>

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

2011-06-18 13:43:02 +0100 Tim-Philipp Müller <[email protected]>

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 <[email protected]>

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

2011-06-17 03:07:09 +0300 Stefan Kost <[email protected]>

* 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.