Commit 96f686fc authored by Sebastian Dröge's avatar Sebastian Dröge

Release 1.1.2

parent a9b4801c
=== release 1.1.2 ===
2013-07-11 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
releasing 1.1.2
2013-07-10 17:16:14 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: Only give sinks a new bus if they have no parent yet
Otherwise we will remove the bus that would proxy messages to playsink
and never set it again. If the sink is already in playsink, all failures
are fatal anyway as it's either a sink that worked before or one that
was set by the user.
https://bugzilla.gnome.org/show_bug.cgi?id=701997
2013-07-10 13:22:04 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: Store a/v/t sinks locally too, not just in playsink
2013-07-10 13:21:29 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaysink.c:
playsink: ref_sink() any sinks that are set on playsink
Otherwise the behaviour of the properties is inconsistent.
2013-07-10 13:20:34 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* tests/check/elements/playbin.c:
playbin: Fix assumptions in the unit test
Unused sinks are still set to READY now during autoplugging
to check their caps. Also playsink owns a ref to the sinks too.
2013-07-10 13:00:21 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Non-TIME segment streams are not waiting automatically
This was leftover code from porting to 1.0 and fixes the playbin
unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=701943
2013-07-09 23:04:49 +0200 Branko Subasic <branko@axis.com>
* win32/common/libgstrtp.def:
win32: add missing rtp buffer methods
2013-07-09 14:55:57 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
playbin: Change sink ownership handling to be a bit more sane
playbin will now only activate the sinks in a single place and
will never change the states of any sinks that are owned by
playsink.
Also handle text-sinks the same way as audio/video sinks inside
playbin.
2013-07-05 21:55:26 +0200 Piotr Drąg <piotrdrag@gmail.com>
* po/POTFILES.in:
po: update POTFILES.in
https://bugzilla.gnome.org/show_bug.cgi?id=703684
2013-07-04 17:09:00 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
* gst-libs/gst/video/colorbalance.c:
colorbalance: Fix the typo in base_init().
2013-07-04 12:54:59 -0400 Thibault Saunier <thibault.saunier@collabora.com>
* gst/adder/gstadder.c:
adder: Do not send flush_start event with the stream lock taken
FLUSH_START is not serialized, so the lock should not be taken when
sending it.
2013-07-05 00:47:08 +0100 Marcin Lewandowski <marcin@saepia.net>
* gst-libs/gst/tag/id3v2frames.c:
tag: ignore malformed ID3v2 TDAT frames
Just skip them, don't cause criticals.
https://bugzilla.gnome.org/show_bug.cgi?id=703283
2013-07-03 09:44:32 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst/audioresample/speex_resampler_int.c:
audioresample: make explicit that neon is disabled and why
https://bugzilla.gnome.org/show_bug.cgi?id=703477
2013-07-02 18:20:39 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
* gst/audioresample/speex_resampler_int.c:
audioresample: disable 16-bit integer NEON support
it seems to be broken (produces no audio), plus the performance gain
is small
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
2013-07-02 14:25:28 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: If we had a previous autoplugged sink, try to reuse it
https://bugzilla.gnome.org/show_bug.cgi?id=701997
2013-07-02 14:18:20 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaysink.c:
playsink: If we switch sinks, make sure that the old sink is set to NULL
2013-07-02 14:02:57 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: Don't change the state of sinks that we passed to playsink already
2013-07-02 14:01:52 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaysink.c:
playsink: Consider new audio/video sinks when reconfiguring
2013-07-02 12:27:03 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: Improve debug output regarding sink selection
2013-07-01 12:52:43 -0600 Brendan Long <self@brendanlong.com>
* gst/playback/gstplaybin2.c:
playbin: Post an error message if a stream combiner doesn't return a request pad.
2013-07-01 13:45:25 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: Only intersect to check if a sink can handle raw caps
Doing a subset check requires fixed caps, which we might not have here.
https://bugs.webkit.org/show_bug.cgi?id=116042
2013-07-01 10:39:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/descriptions.c:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/pbutils-private.h:
pbutils: allow describing unfixed caps if they share the same media type
Caps description and missing plugin code does not really need caps to
be fixed, and indeed they may not be if giving encodebin unfixed caps
that correspond to an unknown encoder or muxer.
So we relax the check, and allow unfixed caps if all the structures
refer to the same media type.
2013-07-01 11:16:34 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: Send all pending events with type < CAPS before sending caps
2013-06-27 16:33:15 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
* gst-libs/gst/video/gstvideoencoder.c:
videoencoder: Send all pending events with type < CAPS before sending caps.
https://bugzilla.gnome.org/show_bug.cgi?id=703196
2013-06-28 14:48:19 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/typefind/gsttypefindfunctions.c:
typefind: avoid too low mpeg/ts probability on small amount of data
With the current test, we get into problems when we try to typefind
a MPEG stream from a small amount of data, which can happen when
we get data pushed from a HTTP source. We thus make a second test
to give higher probability if all the potential headers were either
pack or pes headers (ie, no potential header was unrecognized).
This fixes an issue with a MPEG1/MP2 stream being properly discovered
as video/mpeg from a file, but as audio/mpeg from souphttpsrc.
https://bugzilla.gnome.org/show_bug.cgi?id=703256
2013-06-30 18:17:15 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst-libs/gst/video/gstvideodecoder.c:
* gst-libs/gst/video/gstvideoencoder.c:
video(enc|dec)oder: Don't return not-negotiated if flushing
If the pad is flushing after a failed negotiation, return
GST_FLOW_FLUSHING instead from finish_frame().
https://bugzilla.gnome.org/show_bug.cgi?id=701763
2013-06-30 18:16:35 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: Don't return not-negotiated if flushing
If the pad is flushing after a failed negotiation, return
GST_FLOW_FLUSHING instead from finish_frame().
https://bugzilla.gnome.org/show_bug.cgi?id=701763
2013-06-14 07:23:40 +0200 Edward Hervey <edward@collabora.com>
* gst-libs/gst/pbutils/descriptions.c:
* tests/check/libs/pbutils.c:
pbutils: descriptions: Allow smart codec tag handling
We already have internally the information on what type of stream (audio,
video, container, subtitle, ...) a certain caps is.
Instead of forcing callers to specify which CODEC_TAG category a certain
caps is, use that information to make a smart choice.
Does not break previous behaviour of gst_pb_utils_add_codec_description_to_tag_list
(if tag is specified it will be used, if caps is invalid it will be rejected,
...).
https://bugzilla.gnome.org/show_bug.cgi?id=702215
2013-06-19 09:25:48 +0200 Edward Hervey <edward@collabora.com>
* gst-libs/gst/tag/gstxmptag.c:
xmptag: Add a debug category
Instead of using the default category
2013-06-27 12:23:27 +0200 Patricia Muscalu <patricia@axis.com>
* gst/videotestsrc/gstvideotestsrc.c:
videotestsrc: do not leak lines
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703177
2013-06-26 14:36:17 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst-libs/gst/rtp/gstrtpbasepayload.c:
rtpbasepayload: Do not leak the event when segment is delayed
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703119
2013-06-26 15:03:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: make read uncancelable when reading a message
When we start to read a message, we need to continue reading until the end of
the message or else we lose track and cause parse errors. Use a variable
may_cancel to avoid cancelation after we read the first byte until we have
the complete message.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703088
2013-06-21 20:41:15 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: Don't return not-negotiated if flushing
If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING.
https://bugzilla.gnome.org/show_bug.cgi?id=701763
2013-06-23 12:07:41 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* ext/ogg/gstoggstream.c:
ogg: The Daala headers are little endian, not big endian
2013-06-23 10:30:02 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggstream.c:
ogg: Add Daala support
2013-06-21 19:04:43 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add VP9 description
2013-06-17 08:58:13 +0200 Edward Hervey <edward@collabora.com>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: Fix drop frame handling at startup
In the unlikely case that the decoder drops a frame before the first
input frame is outputted, use the input segment (since it wasn't
carried over to the output segment yet)
https://bugzilla.gnome.org/show_bug.cgi?id=702502
2013-06-21 11:50:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: dispatch when initial buffer has data
When we have data in the inital buffer, dispath the read function to read it
even if the socket has no data to read.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702652
2013-06-20 17:28:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtsp: manage writer child source better
Only add the write child source when we have something to write or else
we will dispatch forever without doing anything.
2013-06-19 13:21:45 +0200 Jonas Holmberg <jonashg@axis.com>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: unref before memset
Unref allocator and input_caps in encoder context before memsetting the
context.
2013-06-19 09:22:50 +0200 Edward Hervey <edward@collabora.com>
* gst-libs/gst/tag/gstxmptag.c:
xmptag: More efficient GSList usage
Instead of constantly appending (which gets more and more expensive), just
prepend to the list (O(1)) and reverse the list before usage.
https://bugzilla.gnome.org/show_bug.cgi?id=702545
2013-06-16 22:39:30 +0200 Branko Subasic <branko@axis.com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c:
rtpbuffer: add gst_rtp_buffer_get_payload_bytes
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.
The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.
https://bugzilla.gnome.org/show_bug.cgi?id=698562
2013-06-17 16:34:26 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst-libs/gst/audio/gstaudiobasesrc.c:
audiobasesrc: add 2 missing gst_buffer_unmap () calls
There are 2 missing calls to gst_buffer_unmap () in the error handling in
create ().
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702467
2013-06-17 16:02:41 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
* gst/playback/gstplaysink.c:
playsink: Fix the block diagram of deinterlace bin.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702465
2013-06-13 11:08:20 -0600 Brendan Long <b.long@cablelabs.com>
* gst/playback/gstplaybin2.c:
playbin: Emit {audio,text,video}-changed signals when pads are removed
https://bugzilla.gnome.org/show_bug.cgi?id=702195
2013-06-11 15:22:50 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/videoconvert/videoconvert.c:
videoconvert: Fix leaking of the chroma resample helper objects
2013-06-10 14:43:35 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
* tests/check/Makefile.am:
* tests/check/elements/playbin-complex.c:
tests: add more unit test for playbin
Add unit test for autoplugging of video_decoder/video_sink combination
based on capsfeatures.
2013-06-10 15:31:38 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Make sure to set a sensible default port for the GSocketConnection
Otherwise it will connect to port 0 if no port is given in the URI.
https://bugzilla.gnome.org/show_bug.cgi?id=701798
2013-06-09 19:20:20 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/adder/gstadder.c:
adder: Reject segments that have a different rate than the output segment
adder does no rate conversion.
2013-06-08 23:51:13 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: When activating a fixed sink, proxy error messages too
If activating a fixed sink fails, everything will fail later anyway
and we can just error out early.
2013-06-08 23:34:53 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: Improve autoplugging of decoder/sink combinations by trying to activate the sink
And if that fails don't bother autoplugging that sink. Also gives
us more accurate sink caps.
2013-06-08 23:08:05 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: Proxy the playbin context to the sinks
2013-06-08 23:04:43 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/playback/gstplaybin2.c:
playbin: Proxy sink messages if we activate a sink in playbin already
This makes sure the application gets any context related messages and
can do whatever is required to a) get the sink a context or b) share
the context with other elements in the pipeline.
The proxying is necessary because the sink is not a child element of
playbin, but instead will at a later point be a child of some bin
inside playsink.
https://bugzilla.gnome.org/show_bug.cgi?id=700967
2013-06-06 15:57:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
decodebin: Let serialize queries before caps events through
Otherwise we're going to deadlock forever because no autoplugging
happens without having caps, but caps can never be send because
we're blocking.
Serialized queries before caps should never be sent unless really
necessary.
2013-06-05 18:36:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
Back to development
=== release 1.1.1 ===
2013-06-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2013-06-05 17:58:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ChangeLog:
* NEWS:
* RELEASE:
* common:
* configure.ac:
releasing 1.1.1
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-ivorbisdec.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-libs/gst/audio/gstaudiopack-dist.c:
* gst-libs/gst/video/video-orc-dist.c:
* gst-libs/gst/video/video-orc-dist.h:
* gst-plugins-base.doap:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/videoconvert/gstvideoconvertorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/volume/gstvolumeorc-dist.c:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/eo.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* win32/common/_stdint.h:
* win32/common/audio-enumtypes.c:
* win32/common/config.h:
* win32/common/video-enumtypes.c:
* win32/common/video-enumtypes.h:
Release 1.1.1
2013-06-05 16:20:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
This is GStreamer Base Plugins 1.1.1
This is GStreamer Base Plugins 1.1.2
Release notes for GStreamer Base Plugins 1.1.1
Release notes for GStreamer Base Plugins 1.1.2
The GStreamer team is proud to announce a new bug-fix release
......@@ -63,93 +63,26 @@ Features of this release
Bugs fixed in this release
* 700342 : decodebin: Crashes and deadlocks when setting to READY while still autoplugging
* 690197 : alsasrc: gets stuck in infinite loop if usb audio device is disconnected while being used
* 697112 : GLTextureUploadMeta: No support for multi-texture formats
* 634407 : decodebin should expose pads in a deterministic order
* 636753 : pbutils function to map (container) caps to filename extension
* 654830 : discoverer, uridecodebin, encodebin and multiple audio streams
* 663350 : theoraenc: do not reset the encoder when we need a keyframe
* 665751 : video: define for formats supported by gst_video_overlay_composition_blend()
* 676884 : audiotestsrc: segment one sample too short due to rounding errors
* 678892 : uridecodebin: differentiate between no URI handler found and URI not accepted by handler
* 679456 : videodecoder: fix compiler optimization hint macro usage
* 681719 : audiovisualizer does not handle VideoMeta
* 685637 : [audioresample] Performance improvements & ARM NEON support
* 687146 : rtpbasedepay: remove unused variable
* 687284 : audioconvert: prefer output formats with the same depth or at least a higher depth
* 687466 : audiobasesink: use the same type as the internal type to return it
* 687472 : video-blend: fix memory leak
* 687817 : textoverlay: support shaded background drawing for all formats
* 689326 : multifdsink: document that adding fd in NULL is not allowed
* 689845 : Encodebin API to handle multiple streams lacking
* 690240 : encodebin: remove test of encoder name vs preset name
* 690591 : No decoder available for type 'audio/x-avi-unknown, codec_id=(int)65534'.
* 690994 : videodecoder: Allow parse function to not use all data on the adapter
* 691072 : decodebin: Doesn't expose pads if no data is received before EOS
* 692358 : appsrc deadlock setting the pipeline to NULL state
* 692613 : tests: reduce number of wake-ups in test applications
* 692930 : avidemux: add raw 8-bit monochrome
* 693302 : decodebin: g_mutex_new is deprecated
* 693401 : gstdecodebin2 doesn't set send event on pad before exposing pad
* 693484 : uridecodebin: query URI to source element and fallback to decoder's URI
* 693750 : Riffmedia doesn't set systemstream=false for some video/mpeg caps
* 693862 : Crash in videoscale (with Orc enabled) on Raspberry Pi
* 694346 : pbutils, typefinding: improve handling of MVC/SVC H.264 streams
* 694389 : non flushing seeks after a segment done, don't sync the ringbuffer
* 694443 : libgstaudio: add support for AAC pass-through
* 694553 : adder: rhythmbox crossfading stopped working after commit a86ca53
* 695203 : xvimagesink: crash in gst_xvimagesink_xvimage_put() with HLS bip-bop stream after a while
* 695276 : libsabi test needs an update for i386
* 695540 : riff: support raw avi with negative height
* 695658 : build: Link libgstrtsp-1.0.so to libm for pow()
* 695660 : appsink: update the emit-signal description
* 695832 : audio: a print causes a floating point exception
* 696100 : videoconvert/videoscale: broken conversion for interlaced Y41B
* 696411 : audiotestsrc: incorrect data size in last buffer
* 696550 : riff: add " note " tag
* 696598 : decodebin pads no longer match order in file
* 696818 : rtsprange: use gst_util_gdouble_to_guint64 in get_seconds
* 696915 : decodebin: get_sticky event STREAM_START fails on newly-exposed pad
* 696916 : videofilter doesn't add caps in pool config
* 697628 : ximagesink: Compile error without HAVE_XSHM
* 697631 : videoscale and videoconvert unit tests need to be updated for latest changes
* 697665 : Add format=WMV3 for WMV 3 video
* 697672 : VP8 passed through rtpbin decodes a single frame and then fails to decode until a key frame passed through
* 697723 : audioringbuffer: Reset segdone when releasing audioringbuffer
* 697808 : sdp: add boxed type for GstSDPMessage
* 698277 : Use gst_plugin_feature_rank_compare() API instead of duplicating the code in many places
* 698410 : Adder: Can not send flush_start and flush_stop in a row
* 698558 : sdp: make it possible to modify session/media attributes
* 698712 : playbin: autoplug video decoder and sink based on caps features
* 698851 : playbin: ability to mix or play multiple audio and text streams simultaneously
* 698888 : SDP session bandwidth not duplicated, causing segfault when freeing...
* 699124 : vorbisdec: crash on shutdown in webkit unit test
* 699187 : videorate: ends up outputting buffers with incorrect duration
* 699470 : dmabuf: handle mmap failure
* 699563 : dmabuf: fix formating
* 699565 : dmabuf: fix memory initialization
* 699566 : dmabuf: don't touch the GstMemory size
* 699744 : alsasrc: timestamps provided by audiosrc subclass not used when running under slave clock
* 699792 : oggmux: Never emitting EOS in GES
* 699894 : videoencoder: Caps event sent before stream-start
* 699960 : videodecoder: Reordering sticky events
* 699971 : oggmux: Sends a segment event before sending a caps event.
* 700006 : audio/video: base classes have suboptimal error handling when allocating a buffer not via a bufferpool
* 700222 : rtpbasepayload: Need to delay segments event after caps event
* 700259 : audio: fix buffer overflow for channels > 64
* 700272 : playback: Use subset checks instead of intersections
* 700324 : playbin hangs trying to play 4K video, and hangs again on interrupt
* 700377 : video: add NV16 pixel format support
* 700400 : video: can't build without orc support - implicit declaration of function 'video_orc_pack_NV16'
* 700411 : dmabuf: Make sure that memory is unmapped before releasing it
* 700413 : ximagesink: add alpha mask support
* 700427 : dmabuf: set the initial memory size to the full size
* 701202 : playsink: Badly initialized contrast/brightness
* 701234 : SIGSEGV in videoconvert_convert_free when using fastpath
* 701316 : rtspconnection: using g_pollable_stream_read and write breaks builds on Ubuntu and Debian stable
* 589242 : videoconvert: need special handling for interlaced I420
* 648359 : baseaudiosrc: ringbuffer: segbase/segdone not updated when ring buffer cleared leads to incorrect timestamps
* 696300 : H264 video is playing too fast because of invalid PTS:
* 698562 : rtpbuffer: broken language bindings for gst_rtp_buffer_get_payload
* 700967 : playbin: sink messages are not received in the bus
* 701798 : rtspsrc: Regression with connections to certain live stream
* 701924 : tests: add playbin test for autoplugging of decoder-sink combination based on capsfeatures
* 701943 : playbin: unit tests fail
* 701997 : [regression] playbin: loses audio clock and hangs when switching songs after about-to-finish
* 702195 : playbin: Emit {audio,text,video}-changed signals when pads are removed
* 702215 : pbutils: descriptions: Allow smart codec tag handling
* 702465 : playsink: Fix the block diagram of deinterlace bin.
* 702467 : audiobasesrc: missing calls to gst_buffer_unmap
* 702545 : tags: xmp: adding tags is very slow
* 702652 : rtspconnection: tunneled connections do not work
* 702840 : queue2 does not work with use-buffering=true
* 703088 : rtspsrc parse error race condition
* 703119 : rtpbasepayload leaks the segment event
* 703196 : videoencoder: Send all pending events with type < CAPS before sending caps.
* 703256 : typefind: Fix low probability typefinding for video/mpeg on small amount of data
* 703477 : audioresample: disable NEON code path for 16-bit integer
* 703684 : POTFILES.in is out of date
==== Download ====
......@@ -186,63 +119,20 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* Akihiro Tsukada
* Alessandro Decina
* Alexander Schrab
* Andoni Morales Alastruey
* Anton Belka
* Arnaud Vrac
* B.Prathibha
* Benjamin Gaignard
* Branko Subasic
* Brendan Long
* Carlos Rafael Giani
* Christian Fredrik Kalager Schaller
* Daniel Drake
* David Schleef
* David Svensson Fors
* Dirk Van Haerenborgh
* Edward Hervey
* Emanuele Aina
* Evan Nemerson
* Greg Rutz
* Jan Schmidt
* Jan Schole
* Jihyun Cho
* Jonas Holmberg
* Jonathan Liu
* Jose Antonio Santos Cadenas
* Josep Torra
* Julien Moutte
* Marc Leeman
* Martin Pitt
* Matej Knopp
* Marcin Lewandowski
* Mathieu Duponchelle
* Matthew Waters
* Michael Olbrich
* Miguel Angel Cabrera Moya
* Nicola Murino
* Nicolas Dufresne
* Ognyan Tonchev
* Olivier Crête
* Patricia Muscalu
* Paul HENRYS
* Pete Beardmore
* Philippe Normand
* Rasmus Rohde
* Rico Tzschichholz
* Piotr Drąg
* Sebastian Dröge
* Sebastian Rasmussen
* Simon Berg
* Sreerenj Balachandran
* Stefan Sauer
* Thiago Santos
* Thibault Saunier
* Thijs Vermeir
* Thomas Scheuermann
* Tim-Philipp Müller
* Tom Greenwood
* Vincent Penquerc'h
* Víctor Manuel Jáquez Leal