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=== 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

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=== release 1.1.1 ===

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2013-06-05 17:58:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
	* common:
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	* configure.ac:
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	* 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
Sebastian Dröge's avatar
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2013-06-05 16:20:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/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:
	  Update .po files

2013-06-05 15:14:43 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* common:
	  Automatic update of common submodule
	  From 098c0d7 to 01a7a46

2013-06-04 17:49:55 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Change GST_WARNING to a GST_DEBUG
	  It's completely normal for some decoders to queue 50-60 frames without
	  it causing any problems, e.g. RPi.

2013-06-01 09:05:16 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst-libs/gst/audio/audio-info.c:
	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: Remove private copy of gst_audio_info_is_equal()
	  And improve the public one a bit based on it.

2013-05-30 16:00:35 -0600  Brendan Long <b.long@cablelabs.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: remove functions added in GLib 2.34
	  g_pollable_stream_read and g_pollable_stream_write were added in GLib 2.34,
	  but Ubuntu 12.04 and Debian Wheezy still use GLib 2.32.
	  Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=701316

2013-05-30 18:48:19 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	  adder: Add GstChildProxy interface for the sinkpads
	  This allows to set the sinkpad properties more easily.
	  Next step: Implement proper synchronization in adder, almost done!

2013-05-30 18:41:22 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	  adder: Hold object lock in setcaps a bit longer to prevent race conditions

2013-05-30 14:57:04 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	  adder: Simplify segment event handling
	  We don't care about upstream segments but generate our own. This
	  makes the code more similar to videomixer again.

2013-05-30 14:45:58 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	  adder: Use gst_audio_info_is_equal() to check if we get the same caps

2013-05-30 14:45:31 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/audio-info.c:
	* gst-libs/gst/audio/audio-info.h:
	* win32/common/libgstaudio.def:
	  audio: Add gst_audio_info_is_equal()

2013-05-30 14:32:03 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	  adder: Don't calls gst_pad_set_caps() on sinkpads
	  It doesn't make much sense and the CAPS query handling
	  on the sinkpads should handle this.

2013-05-30 12:57:11 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	  adder: Set GAP flag on silence buffers we created

2013-05-30 12:54:37 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	* gst/adder/gstadderorc-dist.c:
	* gst/adder/gstadderorc-dist.h:
	* gst/adder/gstadderorc.orc:
	  adder: Remove caching of the processing function
	  The compiler will generate a hashtable from the switch-case, and
	  we need to call functions explicitely for the volume!=1.0 cases
	  anyway.

2013-05-30 12:46:56 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	* gst/adder/gstadderorc-dist.c:
	* gst/adder/gstadderorc-dist.h:
	* gst/adder/gstadderorc.orc:
	  adder: Add support for per-stream volumes

2013-05-30 12:21:06 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	* gst/adder/gstadderorc-dist.c:
	* gst/adder/gstadderorc-dist.h:
	* gst/adder/gstadderorc.orc:
	  adder: Add optimized orc code for F64 processing

2013-05-30 12:05:02 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	  adder: The output buffer must be readable and writable

2013-05-30 12:02:53 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	  adder: Add support for muting individual pads

2013-05-30 11:45:10 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	  adder: Sync pad properties with the GstController

2013-05-30 11:40:01 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	  adder: Add custom GstPad subclass to hold additional data and properties
	  This will later allow to set per-stream volumes and mute status.

2013-05-30 17:31:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	* win32/common/libgstrtsp.def:
	  rtsp: add method to get the TLS connection

2013-05-30 13:14:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: let the sockets be reffed by the connection
	  Don't add an extra ref to the sockets but use that of the connection.
	  Keep the connection around as an IOStream.

2013-05-30 10:50:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: Cleanup the error path
	  Make sure the watch is removed when we close the read socket because of
	  an error.

2013-05-30 10:45:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: cleanup the watch reset function

2013-05-30 10:30:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: check if the streams are still active
	  Don't try to read/write from an inactive stream. When we, for example,
	  transfer the second connection in tunneling mode, we are not interested anymore
	  on read/write activity on the old connection.

2013-05-29 17:44:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: use child sources instead of using the sockets
	  Use the source of the pollable input/output streams instead of
	  accessing the sockets directly.

2013-05-29 16:15:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: fix input/output streams for tunneling

2013-05-29 15:27:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: don't use sockets for blocking
	  Use the blocking and non-blocking API of the input/output streams instead
	  of polling the sockets directly. This also allows us to simplify some
	  code.

2013-05-28 17:06:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* gst-libs/gst/rtsp/gstrtsptransport.c:
	* gst-libs/gst/rtsp/gstrtsptransport.h:
	* gst-libs/gst/rtsp/gstrtspurl.c:
	  rtsp: add TLS support
	  Add flag to select TLS in the transport.
	  Enable TLS on the socketclient when we use a TLS uri.

2013-05-28 16:45:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: use the input/output stream of clientconnection
	  Don't use the raw sockets for RTSP communication but use the IOStream.
	  This is needed if we are going to use TLS later.

2013-05-28 11:16:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: set sockets non-blocking

2013-04-05 16:50:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: use GSocketClient for making connections
	  Use the GSocketClient API for making connections with the server. This removes a
	  bit of code and gives us the ability to do TLS later.

2013-05-27 15:32:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  Revert "rtspconnection: Use a GSocketAddressNumerator to resolve the addresses"
	  This reverts commit 15a0bb0a10dcbc99c7f52e28ec9d0395699851ae.
	  We should be using GSocketClient

2013-05-30 05:24:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videoconvert/videoconvert.c:
	* gst/videoconvert/videoconvert.h:
	  videoconvert: free tmplines correctly
	  Keep track of how many tmplines we allocated and use that to free the
	  correct amount of lines.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701234

2013-05-29 10:33:48 -0600  Daniel Drake <dsd@laptop.org>

	* gst/playback/gstplaysink.c:
	  playsink: pass translated color balance value to channel
	  We found a case where untranslated values were being passed from the
	  proxy to the underlying channel, causing bad color balance values
	  in some setups.
	  Thanks to Sebastian Dröge for clarifying how the code works, and
	  suggesting the fix.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701202

2013-05-29 10:15:36 -0600  Brendan Long <b.long@cablelabs.com>

	* gst/playback/gstplaybin2.c:
	  playbin: Don't take an extra reference to the custom stream combiners
	  They are automatically reffed when added to the bin because they're
	  already not floating anymore.

2013-05-29 16:41:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	  alsasrc: Dump some more debug output about the device configuration

2013-05-29 16:39:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/alsa/gstalsasink.c:
	  alsasink: Update internal buffer/period times with the values that were configured on the device

2013-05-29 10:37:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* tests/check/Makefile.am:
	* tests/check/elements/playbin-complex.c:
	* tests/check/elements/playbin-compressed.c:
	  playbin: Rename compressed unit test to complex
	  It's not really about compressed streams anymore, but also
	  about stream switching and stream combiners.

2013-05-29 10:35:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	* tests/check/elements/playbin-compressed.c:
	  playbin: Set custom stream-combiners to NULL and unref before finalizing

2013-05-28 10:59:22 -0600  Brendan Long <b.long@cablelabs.com>

	* tests/check/elements/playbin-compressed.c:
	  playbin: Add playbin audio-stream-combiner test using adder

2013-05-28 11:23:56 -0600  Brendan Long <b.long@cablelabs.com>

	* gst/playback/gstplaybin2.c:
	  playbin: Rename select to combine and selector to combiner in playbin

2013-05-17 17:23:46 -0600  Brendan Long <b.long@cablelabs.com>

	* gst/playback/gstplaybin2.c:
	  playbin: Add support for custom stream-combiners
	  This allows to chose something else than input-selector
	  for multiple audio/video/text streams, e.g. an adder could
	  be used for audio.
	  It is needed for example to implement some of the more
	  advanced HTML5 video features.
	  https://bugzilla.gnome.org/show_bug.cgi?id=698851

2013-05-28 13:32:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Don't call autoplug-query on shutdown
	  And remove leftover debug code

2013-05-28 13:23:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin: In autoplug-queries, add the actual decoder/parser/etc template caps
	  Add the actual decoder/parser/etc caps at the very end to
	  make sure we don't cause empty caps to be returned, e.g.
	  if a parser asks us but a decoder is required after it
	  because no sink can handle the format directly.

2013-05-28 13:14:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin: Forward CONTEXT queries to the corresponding sink if we have one
	  https://bugzilla.gnome.org/show_bug.cgi?id=700967

2013-05-28 13:08:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybin2.c:
	  playbin: Refactor autoplug-query handling
	  We now only check sinks and factories of the corresponding media
	  type. It doesn't make sense to pass audio/subtitle caps to a video
	  decoder.

2013-05-28 13:06:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Block on serialized queries too
	  Otherwise we will only block after the serialized, non-sticky event
	  after the CAPS event or the first buffer. If we're waiting for another
	  pad to finish autoplugging after we got final caps on this pad, it
	  will mean that we will let the ALLOCATION query pass although the
	  pad is not exposed yet.

2013-05-28 12:03:49 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gsturidecodebin.c:
	  decodebin: Pass the element in the autoplug-query signal too

2013-05-28 11:40:51 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Need to lock the chain mutex in autoplug_query

2013-05-28 11:36:58 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/playback/gstplaysinkconvertbin.c:
	  playsinkconvertbin: Fix leak of the downstream caps filter

2013-05-28 11:05:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin: Refactor autoplug-query handling a bit

2013-05-27 14:53:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Use a GSocketAddressNumerator to resolve the addresses
	  Instead of just trying the first possible resolution we're trying all
	  resolutions until one works.

2013-05-27 13:04:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/theora/gsttheoradec.c:
	  theoradec: Require caps to be set before data flow happens

2013-05-27 11:53:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video-format.c:
	* gst-libs/gst/video/video-orc.orc:
	  video-format: fix NV16 unpack
	  We can just use the NV12 functions, the only difference is the
	  vertical subsampling.

2013-05-27 11:25:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video-chroma.h:
	  video-chroma: add interlaced flag

2013-05-17 16:34:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videoconvert/videoconvert.c:
	* gst/videoconvert/videoconvert.h:
	  videoconvert: run chroma resamplers
	  Run the chroma upsampler after unpack and the chroma subsampler
	  before pack for higher quality conversions and correct chroma siting.

2013-05-17 16:26:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.c:
	* gst/videotestsrc/videotestsrc.h:
	  videotestsrc: subsample chroma before packing
	  Run the chroma subsampler before packing.

2013-05-17 16:22:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video-chroma.c:
	* gst-libs/gst/video/video-chroma.h:
	* win32/common/libgstvideo.def:
	  video-chroma: add chroma resampler
	  Add functions to up/downsample chroma in horizontal and vertical
	  directions. These functions work in-placeand are meant to be used on the
	  input/output of the pack/unpack functions.

2013-04-01 16:16:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/video-format.c:
	* gst-libs/gst/video/video-format.h:
	* gst-libs/gst/video/video-orc.orc:
	  video: don't perform subsampling while packing
	  Don't perform subsampling when packing but let this be done by a
	  separate subsampling step.

2013-04-01 16:05:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videoconvert/videoconvert.c:
	  videoconvert: reformat

2013-05-17 15:45:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/Makefile.am:
	* gst-libs/gst/video/video-chroma.c:
	* gst-libs/gst/video/video-chroma.h:
	* gst-libs/gst/video/video-format.c:
	* gst-libs/gst/video/video-format.h:
	  video: move chroma functions to separate file

2013-05-17 15:41:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/videoconvert/videoconvert.c:
	  videoconvert: actually use the input pixels
	  Operate on the provided pixels array instead of the temp array.

2013-05-17 15:40:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst-libs/gst/video/gstvideometa.h:
	  videometa: fix docs

2013-05-25 16:08:06 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst-libs/gst/video/gstvideoencoder.c:
	  videoencoder: Don't require an output state to be set before allocating output buffers

2013-05-24 17:43:53 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: Ensure we have enough data when reading the sync marker in the AAC/LOAS typefinder

2013-05-24 16:52:50 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudioencoder.c:
	  audio: Always provide a buffer in gst_audio_(enc|dec)oder_allocate_output_buffer()
	  We have no way of tell the caller of the exact error (e.g. if we're flushing),
	  so will have to wait until the caller uses API that returns a GstFlowReturn,
	  for example when pushing this buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700006

2013-05-24 16:51:17 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst-libs/gst/video/gstvideodecoder.c:
	* gst-libs/gst/video/gstvideoencoder.c:
	  video: Always provide a buffer in gst_video_(enc|dec)oder_allocate_output_buffer()
	  We have no way of tell the caller of the exact error (e.g. if we're flushing),
	  so will have to wait until the caller uses API that returns a GstFlowReturn,
	  for example when pushing this buffer.
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