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=== release 1.8.3 ===

2016-08-19  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.8.3

2016-08-19 11:54:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* 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/hr.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

2016-06-16 10:01:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: remove eos avoidance workaround
	  This workaround tried to avoid an EOS event when seeking to the
	  end of an Ogg stream in order to find its duration. At some point,
	  an EOS event there would cause any queue2 upstream to pause and
	  not restart on a seek back to the beginning. This now appears to
	  not be the case anymore, and so the workaround can be removed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767689

2016-05-24 00:44:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/allocators/Makefile.am:
	* gst-libs/gst/app/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/fft/Makefile.am:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/riff/Makefile.am:
	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/sdp/Makefile.am:
	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/video/Makefile.am:
	  g-i: pass compiler env to g-ir-scanner
	  It's what introspection.mak does as well. Should
	  fix spurious build failures on gnome-continuous.

2016-07-20 11:47:48 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: fix unknown duration playing Ogg over HTTP
	  If the duration is not known from the chain, it might be known
	  by the startup seek.
	  This fixes failure to seek.
	  Merged with a patch from Tim-Philipp Müller <tim@centricular.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=768991

2016-07-18 19:59:23 +1000  Duncan Palmer <dpalmer@digisoft.tv>

	* sys/xvimage/xvimageallocator.c:
	* sys/xvimage/xvimageallocator.h:
	  xvimageallocator: const correctness in gst_xvimage_allocator_alloc().
	  https://bugzilla.gnome.org/show_bug.cgi?id=767712

2016-07-18 14:20:11 +0100  Tim-Philipp Müller <tim@centricular.com>

	* sys/xvimage/xvimageallocator.c:
	  xvimagesink: only error out if the allocated memory is too small
	  https://bugzilla.gnome.org/show_bug.cgi?id=767712

2016-07-07 22:27:15 +1000  Duncan Palmer <dpalmer@digisoft.tv>

	* sys/xvimage/xvimageallocator.c:
	* sys/xvimage/xvimageallocator.h:
	* sys/xvimage/xvimagepool.c:
	  xvimagesink: error out on buffer size sanity check failure.
	  If sanity checks on the buffer size allocated by XvShmCreateImage() fail,
	  call on g_set_error(), rather than just logging a warning, as this
	  failure is fatal.
	  Add a sanity check on buffer size when the video format is RGB. This adds to
	  existing checks on various YUV pixel formats.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767712

2016-07-08 16:43:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Remove some more fields from the caps when creating from discoverer info
	  parsed, framed, stream-format and alignment are only relevant for parsers and
	  should not matter here. We still want to be able to use an encoder that can
	  only output byte-stream if the input was avc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768566

2016-07-08 15:45:25 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/missing-plugins.c:
	  missing-plugins: Remove some other fields when cleaning up caps
	  Caps are cleaned up for missing plugins, and for creating encoding profiles
	  and caps descriptions.
	  Fields like streamheader, parsed, framed, stream-format and alignment are not
	  relevant here. The last ones all because a parser will take care of them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768566

2016-07-04 17:19:08 +0100  Sergio Torres Soldado <torres.soldado@gmail.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Fix potential deadlock caused by blocking read forever
	  Reset the connection "may_cancel" property to avoid a potential deadlock
	  if there is no data to read and the socket stays blocked forever.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768249

2016-07-04 11:16:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: fix criticals fixating a non existent field
	  https://bugzilla.gnome.org/show_bug.cgi?id=766970

2016-07-04 11:07:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: Protect samples_in/bytes_out and audio info with object lock
	  It might cause invalid calculations during the CONVERT query otherwise.

2016-07-04 11:12:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: Protect samples_in/bytes_out and audio info with object lock
	  It might cause invalid calculations during the CONVERT query otherwise.

2016-07-04 11:00:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/audio/gstaudioutilsprivate.c:
	* gst-libs/gst/audio/gstaudioutilsprivate.h:
	  audioencoder/decoder: Move encoded audio conversion function to a common place
	  No need to duplicate this non-trivial function.

2016-07-04 09:15:03 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: fix criticals fixating a non existent field
	  https://bugzilla.gnome.org/show_bug.cgi?id=766970

2016-07-04 10:55:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Use the object lock to protect bytes/time tracking
	  And especially don't use the stream lock for that, as otherwise non-serialized
	  queries (CONVERT) will cause the stream lock to be taken and easily causes the
	  application to deadlock.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768361

2016-07-04 10:47:36 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	* gst-libs/gst/video/gstvideoencoder.c:
	* gst-libs/gst/video/gstvideoutilsprivate.c:
	* gst-libs/gst/video/gstvideoutilsprivate.h:
	  videoencoder/decoder: Move conversion utility functions to a common header and use consistently in encoder/decoder

2016-07-04 10:52:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideoencoder.c:
	  videoencoder: Use the object lock to protect bytes/time tracking

2016-06-30 18:53:07 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/tag/gsttagdemux.c:
	  tagdemux: fix handling of very short files in push mode
	  By default we'll wait for a certain amount of data before
	  attempting typefinding. However, if the stream is fairly
	  short, we might get EOS before we ever attempted any
	  typefinding, so at this point we should force typefinding
	  and output any pending data if we manage to detect the
	  type.
	  https://bugzilla.gnome.org//show_bug.cgi?id=768178

2016-06-30 17:30:34 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/tag/gsttagdemux.c:
	  tagdemux: fix erroring out if we reach EOS without detecting type
	  In 0.10 the source pad was a dynamic pad that was only added once
	  the type had been detected, but in 1.x it's an always source pad,
	  so checking whether it's still NULL won't work to detect if the
	  type has been detected.
	  Makes tagdemux error out when we get EOS but haven't managed to
	  identify the format of the data after the tag.
	  https://bugzilla.gnome.org//show_bug.cgi?id=768178

2016-06-29 18:14:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/audio-channels.c:
	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: Handle fallback channel mask for mono correctly
	  It's 0 and no mask should be set for mono at all.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757472

2016-06-27 20:49:38 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Force STEP events on the video-sink for GST_FORMAT_BUFFERS
	  It does not make much sense for audio sinks.

2016-06-27 20:53:37 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Don't send another step event to the audio-sink if we got step-done from there
	  Otherwise we would end up with a deadlock as the audio-sink emits step-done
	  from its streaming thread.

2016-06-21 10:24:15 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* tests/check/libs/discoverer.c:
	  discoverer: Only allow serializing OK discoverer infos to GVariants
	  They will be incomplete otherwise and we can't generate the full serialized
	  information, and instead will crash somewhere on the way.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767859

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=== release 1.8.2 ===

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2016-06-09 11:50:43 +0300  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* docs/plugins/gst-plugins-base-plugins.args:
	* 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-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-opus.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-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/config.h:
	  Release 1.8.2

2016-06-09 11:17:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* 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/hr.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
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2016-06-09 10:05:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/hr.po:
	* po/pt_BR.po:
	* po/sk.po:
	  po: Update translations

2016-05-10 13:56:13 +0200  Stian Selnes <stian@pexip.com>

	* gst-libs/gst/video/video-color.c:
	* tests/check/libs/video.c:
	  video-color: Fix colorimetry IS_UNKNOWN
	  Fix issue with colorimetry default indicies not being in sync with the
	  actual table causing IS_UNKNOWN() to sometimes fail.
	  https://bugzilla.gnome.org/show_bug.cgi?id=767163

2016-05-14 14:41:28 +0200  Olivier Crête <olivier.crete@collabora.com>

	* ext/opus/gstopusdec.c:
	  opusdec: Use GST_AUDIO_DECODER_ERROR
	  This way, the first invalid stream won't break all decoding.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766265

2016-05-16 12:52:50 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst-libs/gst/video/gstvideosink.c:
	  videosink: ensure the debug category is always initialized
	  gst_video_sink_center_rect() can be called without a GstVideoSink
	  having been instantiated so we can't relly on the video sink
	  class_init function to init the category.
	  Fix a warning when running:
	  GST_CHECKS=test_video_center_rect GST_DEBUG=6 G_DEBUG=fatal_warnings make libs/video.check-norepeat
	  https://bugzilla.gnome.org/show_bug.cgi?id=766510

2016-05-16 15:39:02 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  playbin: fix suburidecodebin leak
	  We take a ref before removing which was never freeded.
	  The element is still alive anyway because the group has its own ref as
	  well.
	  Fix a leak with the 'test_suburi_error_wrongproto' test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=766515

2016-05-10 21:34:53 +0900  Hyunjun Ko <zzoon@igalia.com>

	* gst-libs/gst/sdp/gstsdpmessage.c:
	  sdp: parse sdp attributes in case that sdp message doesn't contain mikey message
	  https://bugzilla.gnome.org/show_bug.cgi?id=766204

2016-04-29 11:06:49 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Fix caps memory leak

2016-04-28 11:18:23 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Fail to create encoding profile from discoverer info if no streams could be added
	  https://bugzilla.gnome.org/show_bug.cgi?id=765708

2016-04-28 11:21:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Recurse into nested container profiles and only add the final audio/video streams
	  If we e.g. have AVI with DV container with video/audio inside the DV
	  container, we can't handle this at this point with an encoding profile.
	  Instead of erroring out, flatten the container hierarchy.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765708

2016-04-28 11:15:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Move adding of each stream to a helper function
	  https://bugzilla.gnome.org/show_bug.cgi?id=765708

2016-05-02 14:21:55 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/opus/gstopusdec.c:
	* tests/check/elements/opus.c:
	  opusdec: intersect with the filter before returning on getcaps
	  So upstream gets a smaller set to decide upon as it is what it requested
	  with the filter
	  https://bugzilla.gnome.org/show_bug.cgi?id=765684

2016-05-02 10:23:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/opus/gstopusdec.c:
	* tests/check/elements/opus.c:
	  opusdec: improve getcaps to return all possible rates
	  The library is capable of converting to different rates.
	  Includes tests.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765684

2016-05-02 10:21:52 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/opus/gstopusdec.c:
	  opusdec: remove artificial restriction on rate negotiation
	  Remove restrictions when rate is 48000, the underlying lib supports
	  converting any of the input to any of the output rates.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765684

2016-05-01 23:19:57 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/opus/gstopusdec.c:
	  opusdec: refactor getcaps repeated code into a function
	  Easier to read and maintain

2016-05-02 10:36:07 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/opus.c:
	  tests: opus: remove apparently useless macro in tests

2016-04-28 09:59:25 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* ext/opus/gstopusdec.c:
	  opusdec: fix caps leaks
	  The caps returned by gst_pad_get_allowed_caps() was leaked.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765706

2016-04-25 17:16:04 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/encoding/gstsmartencoder.c:
	  smartencoder: Only accept TIME segments for real
	  ... and don't try to push pending data without ever having received a SEGMENT
	  event before EOS
	  https://bugzilla.gnome.org/show_bug.cgi?id=765541

2016-04-25 16:47:00 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: H264 level idc 0 is not valid
	  Don't put level=0 into the caps, it confuses other elements.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765538

2016-04-25 16:48:36 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: H265 level idc 0 is not valid
	  Don't put level=0 into the caps, it confuses other elements.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765538

2016-04-25 16:06:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Remove codec_data and streamheader fields from constraint caps
	  When converting discoverer output to an encoding profile, it makes sense to
	  omit these. It's very very unlikely that our encoder is going to produce bit
	  by bit the same codec_data or streamheader.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765534

2016-04-25 15:05:36 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.h:
	  encoding-profile: Don't put G_BEGIN_DECLS around #include statements
	  It should only be around our own declarations.

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=== release 1.8.1 ===

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2016-04-20 18:15:39 +0300  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* 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-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-opus.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-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/config.h:
	  Release 1.8.1

2016-04-20 18:02:22 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.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/hr.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
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2016-04-15 17:48:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* win32/common/libgstsdp.def:
	  win32: update .def for new API

2016-04-20 15:30:04 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/da.po:
	  po: Update translations

2016-04-19 17:36:20 +0200  Josep Torra <n770galaxy@gmail.com>

	* gst-libs/gst/sdp/gstmikey.c:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	  sdp: update since markers to 1.8.1 for some new APIs
	  As we decided to backport some fixes we update the since markers.

2016-04-15 00:18:50 -0700  Aleix Conchillo Flaqué <aconchillo@gmail.com>

	* gst-libs/gst/sdp/gstsdpmessage.c:
	* gst-libs/gst/sdp/gstsdpmessage.h:
	  sdpmessage: new gst_sdp_media_parse_keymgmt/gst_sdp_media_parse_keymgmt
	  We add a couple of new functions gst_sdp_media_parse_keymgmt and
	  gst_sdp_media_parse_keymgmt. We also implement
	  gst_sdp_message_attributes_to_caps and gst_sdp_media_attributes_to_caps
	  in terms of these new functions and also gst_mikey_message_to_caps.

2016-04-14 23:29:34 -0700  Aleix Conchillo Flaqué <aconchillo@gmail.com>

	* gst-libs/gst/sdp/gstmikey.c:
	* gst-libs/gst/sdp/gstmikey.h:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	  mikey: add new function gst_mikey_message_to_caps

2016-04-16 02:11:59 +1000  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/audio/gstaudioringbuffer.c:
	  Revert "audioringbuffer: start ringbuffer if needed upon commit"
	  This reverts commit 13ee94ef1091f8a8a90dbd395b39876c26c5188e.
	  Causes audio glitches at startup by starting to output segments
	  from the ringbuffer before it has been filled / fully prerolled.
	  https://bugzilla.gnome.org/show_bug.cgi?id=657076

2016-04-14 17:26:54 -0700  Aleix Conchillo Flaqué <aconchillo@gmail.com>

	* gst-libs/gst/sdp/gstmikey.c:
	  mikey: allow passing srtp or srtcp to create mikey message
	  Current implementation requires all srtp and srtcp parameters to be
	  given in the caps. MIKEY uses only one algorithm for encryption and one
	  for authentication so we now allow passing srtp or srtcp parameters. If
	  both are given srtp parametres will be preferred.
	  https://bugzilla.gnome.org/show_bug.cgi?id=765027

2016-04-11 11:28:09 +0200  Fabrice Bellet <fabrice@bellet.info>

	* gst-libs/gst/audio/gstaudiosink.c:
	* gst-libs/gst/audio/gstaudiosrc.c:
	  audio: Fix a race with the audioringbuffer thread
	  There is a small window of time where the audio ringbuffer thread
	  can access the parent thread variable, before it's initialized
	  by the parent thread. The patch replaces this variable use by
	  g_thread_self().
	  https://bugzilla.gnome.org/show_bug.cgi?id=764865

2016-03-24 14:59:48 +1100  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: Hold new buffering_post lock while posting msgs
	  There's a small window between decodebin choosing a buffering level
	  to post and another thread choosing a different buffering level
	  where things can race. Close that window by holding a new lock
	  that's only for posting buffering messages - like what was done
	  in multiqueue.
	  https://bugzilla.gnome.org/show_bug.cgi?id=764020

2016-02-26 02:56:15 +1100  Jan Schmidt <jan@centricular.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: Reduce URI typefinder from MAX to LIKELY
	  Don't claim maximum likelihood for anything that starts
	  with text that looks like a uri, it's too broad.

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=== release 1.8.0 ===

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2016-03-24 12:19:23 +0200  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* 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-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-opus.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-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/config.h:
	  Release 1.8.0

2016-03-24 11:43:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* 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/hr.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
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2016-03-08 13:22:32 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst-libs/gst/pbutils/install-plugins.c:
	  install-plugins: update documentation
	  Use gst-inspect-1.0 instead of gst-inspect-0.10
	  https://bugzilla.gnome.org/show_bug.cgi?id=763316

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=== release 1.7.91 ===

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2016-03-15 12:02:20 +0200  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* 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-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-opus.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-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/audio-enumtypes.c:
	* win32/common/config.h:
	  Release 1.7.91

2016-03-15 11:48:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.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/gl.po:
	* po/hr.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/pt_BR.po:
	* po/ro.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/tr.po:
	  Update .po files
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2016-03-15 11:40:06 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/cs.po:
	* po/fr.po:
	* po/hu.po:
	* po/pl.po:
	* po/ru.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: Update translations

2016-03-14 17:06:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Shut down all elements explicitly to NULL state before freeing the decode chain
	  Due to transient locked state during autoplugging, some elements might be
	  ignored by the GstBin::change_state() and might still be running. Which could
	  then cause pad-added and similar accessing decodebin state that does not exist
	  anymore, and crash.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763625

2016-03-13 13:59:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/tcp/gstmultihandlesink.c:
	* gst/tcp/gstmultihandlesink.h:
	* tests/check/elements/multifdsink.c:
	* tests/check/elements/multisocketsink.c:
	  multihandlesink: Remove useless streamheader storage
	  We don't do anything with it but always get them from the caps anyway, so
	  stop storing them and having complicated logic around that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763278

2016-03-13 10:51:30 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/tcp/gstmultihandlesink.c:
	* gst/tcp/gstmultihandlesink.h:
	  multihandlesink: Only don't send HEADER buffers normally if they are actually streamheaders from the caps
	  And also consider HEADER buffers without DELTA_UNIT flag as sync points. This
	  fixes sync-mode=2 with mpegtsmux for example, which has no streamheaders but
	  puts the HEADER flag on its keyframes.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763278

2016-03-12 19:47:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: expose_pad() is always called with lock==TRUE, simplify code
	  This basically reverts ee44337fc3e3030a5155d28b3561af157e6c6003 .
	  https://bugzilla.gnome.org/show_bug.cgi?id=763491

2016-03-12 19:46:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Don't check twice if the decode chain is complete in pad_added_cb()
	  expose_pad() already does the same.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763491

2016-03-12 19:45:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Don't hold EXPOSE_LOCK in type_found() outside the stream lock
	  In other places we lock it the other way around, leading to possible
	  deadlocks. Also this will deadlock if analyze_pad() causes a new element to be
	  autoplugged that adds new pads on itself when its state is changed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763491

2016-03-13 10:58:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/tcp/gstmultioutputsink.c:
	  tcp: Remove unused file
	  It's a copy of multihandlesink, but completely outdated. Let's get rid of it
	  before it gets even more outdated.
	  https://bugzilla.gnome.org/show_bug.cgi?id=763278

2016-03-08 19:22:34 +0100  Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>

	* ext/pango/gstbasetextoverlay.c:
	* ext/pango/gstbasetextoverlay.h:
	  basetextoverlay: Add new properties and alignment type for unclamped absolute positions
	  Introduces [x-absolute, y-absolute] properties
	  for positioning in +/- MAX_DOUBLE range.
	  Adds new (h/v)alignment type "absolute" where coordinates
	  map the text area to be exactly inside of video canvas for [0, 0] - [1, 1]:
	  [0, 0]: Top-Lefts of video and text are aligned
	  [0.5, 0.5]: Centers are aligned
	  [1, 1]: Bottom-Rights are aligned
	  https://bugzilla.gnome.org/show_bug.cgi?id=761251

2016-03-11 13:15:03 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/pango/gstbasetextoverlay.c:
	  Revert "textoverlay: Do not limit positioning to video area."
	  This reverts commit a48daf6dd8cb69b4260a03aa7f3cdf227d4f1602.
	  This changed behaviour in a way that's not always
	  backwards-compatible.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761251

2016-02-25 05:07:04 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* win32/common/libgstfft.def:
	  win32: Add a module definitions file for gstfft

2016-03-09 09:56:52 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/theora/gsttheoradec.c:
	* ext/theora/gsttheoradec.h:
	* ext/theora/gsttheoraenc.c:
	* ext/theora/gsttheoraenc.h:
	  theora: fix performance category initialisation
	  Remove unused _register() functions and look up the performance
	  debug category in a function that's actually called at some point.

2016-03-04 17:13:59 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channel-mixer.h:
	  audio-channel-mixer: improve non-interleaved flags
	  Make separate flags for non-interleaved input and output because the
	  channel mixer should be able to convert between the two layouts in the
	  future.

2016-03-04 12:12:56 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tools/gst-play.c:
	  gst-play: remove peculiar setting of invalid -v property

2016-02-05 14:14:37 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: fix chaining causing running time to restart from 0
	  This fixes:
	  gst-play-1.0 http://relay-nyc.gameowls.com:8000/chiptune.ogg
	  https://bugzilla.gnome.org/show_bug.cgi?id=758282

2016-03-03 20:10:17 +0100  Havard Graff <havard.graff@gmail.com>

	* ext/opus/gstopusdec.c:
	  opusdec: plug caps leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=763059

2016-03-02 20:47:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  Revert "playbin: use avdeinterlace for deinterlacing until deinterlace is ported"
	  This reverts commit 0615794300234e3efbcb49a524efdee11171ab4c.
	  deinterlace was ported at some point in the last 4 years and has better video
	  format support, and especially better negotiation than avdeinterlace. Having
	  avdeinterlace but not deinterlace causes various problems in zerocopy
	  scenarios.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760553

2016-03-02 18:47:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/encoding/gstencodebin.c:
	  encodebin: Make dispose() function safe to be called multiple times

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=== release 1.7.90 ===

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2016-03-01 18:14:54 +0200  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* docs/plugins/gst-plugins-base-plugins.hierarchy:
	* 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-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-opus.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-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/config.h:
	  Release 1.7.90
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2016-03-01 16:53:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* 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/hr.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:
	  po: Update translations

2016-01-28 16:26:47 +0100  Tom Deseyn <tom.deseyn@gmail.com>

	* gst/tcp/gstmultisocketsink.c:
	  multisocketsink: handle client close correctly and EWOULDBLOCK
	  Fixes 100% cpu usage when client disconnects. Commit 6db2ee56
	  would just make multisocketsink ignore reads of 0 bytes without
	  removing the client, so we'd get woken up over and over again
	  for the client.
	  Fix the original issue differently by handling the non-fatal error code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761257
	  https://bugzilla.gnome.org/show_bug.cgi?id=743834

2016-02-27 00:11:02 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/video/video-orc-dist.c:
	* gst-libs/gst/video/video-orc-dist.h:
	  video: update disted orc backup file
	  https://bugzilla.gnome.org/show_bug.cgi?id=761851

2016-02-11 11:27:57 +0100  Göran Jönsson <goranjn@axis.com>

	* gst-libs/gst/video/video-converter.c:
	* gst-libs/gst/video/video-orc.orc:
	  video-converter: add direct UYVY to GRAY8 conversion function
	  https://bugzilla.gnome.org/show_bug.cgi?id=761851

2016-02-04 16:01:00 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/opus/gstopusdec.c:
	  opus: fix mono<->stereo up/down-mixing
	  https://bugzilla.gnome.org/show_bug.cgi?id=761588

2016-02-26 17:09:06 +0800  Lim Siew Hoon <siew.hoon.lim@intel.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  pbutils: docs: Remove the empty lines in between <refsect2> and </refsect2>
	  They are converted into <para></para> by gtk-doc...
	  https://bugzilla.gnome.org/show_bug.cgi?id=762674

2016-02-26 12:41:01 +0200  Sebastian Dröge <sebastian@centricular.com>

	* common:
	  Automatic update of common submodule
	  From b64f03f to 6f2d209

2016-02-26 00:53:05 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/opus/gstopusenc.c:
	  opusenc: remove deprecated "cbr", "audio", and "constrained-vbr" properties
	  They have been replaced by "audio-type" and "bitrate-type".
	  https://bugzilla.gnome.org/show_bug.cgi?id=756282

2016-02-26 00:37:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/gst-plugins-base-plugins.hierarchy:
	* docs/plugins/gst-plugins-base-plugins.interfaces:
	* docs/plugins/inspect/plugin-opus.xml:
	  docs: add Opus to docs

2016-02-26 00:20:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* ext/Makefile.am:
	* ext/opus/Makefile.am:
	* ext/opus/gstopus.c:
	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	  opus: move Opus audio decoder and encoder from -bad to -base
	  Hook into build system after moving history.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756282

2016-02-25 23:51:42 +0000  Tim-Philipp Müller <tim@centricular.com>

	  Merge branch 'plugin-move-opus'
	  Move Opus decoder and encoder from -bad to -base.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756282

2016-02-25 23:13:39 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-play-1.0.1:
	* tools/gst-play.c:
	  tools: gst-play: add 'n' and 'b' as additional shortcuts for next/previous item
	  < and > are composed with shift + something else on many keyboards
	  layouts, so don't work well when injecting them via windowing systems
	  which will send them as shift key press and separate other key, and
	  we the don't combine that to < or > properly. n/b are easier.

2016-02-26 00:02:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/libs/baseaudiovisualizer.c:
	  audiovisualizer: Use the library instead of including the source file
	  Fixes build now that the shader enum GType has moved to a different file.

2016-02-25 20:39:04 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	  audiovisualizer: Let GstAudioVisualizerShader enum GType be autogenerated by glib-mkenums
	  That happens automatically already anyway.

2016-02-25 17:46:31 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/video/video-frame.c:
	  video: flesh out docs for gst_video_frame_map()

2016-02-25 10:47:17 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	  visual: correct type name
	  Base class type name should not reference libvisual since not all child
	  elements use this. This was an oversight when merging audiovisualizers into
	  a common base class.

2016-02-24 14:05:03 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-quantize.c:
	  audio-quantize: fix feedback dither
	  Make sure we allocated enough extra space in the error buffer to
	  store the feedback error.

2016-02-24 12:54:39 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: perform dithering on the current format
	  Use the current (intermediate) format to decide how to set up dithering
	  instead of the input format.

2016-02-23 18:23:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasepayload.c:
	  rtpbasepayload: Handle gst_pad_get_current_caps() returning NULL gracefully

2016-02-23 09:35:14 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstplaysink.c:
	  Revert "playsink: Properly mark pending blocked pads"
	  This reverts commit 62053852de01fb324a915b27c00f5b8dc0f66fb3.
	  The issue that the patch fixes is only noticeable when using decodebin3,
	  which isn't yet in master.

2015-12-10 15:32:06 +0100  Adam Miartus <adam.miartus@streamunlimited.com>

	* gst-libs/gst/tag/gstid3tag.c:
	  tag: id3v2: read conductor tag
	  ID3v2 features the TPE3 info frame, which contains information
	  about the conductor.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762451

2016-02-20 11:31:43 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/theora/gsttheoradec.c:
	* gst-libs/gst/video/video-frame.c:
	* gst/videoconvert/gstvideoconvert.c:
	* gst/videoscale/gstvideoscale.c:
	* sys/ximage/ximage.c:
	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvcontext.c:
	* sys/xvimage/xvimage.c:
	* sys/xvimage/xvimagesink.c:
	  Fix use of undeclared core debug category symbols
	  libgstreamer currently exports some debug category
	  symbols GST_CAT_*, but those are not declared in any
	  public headers.
	  Some plugins and libgstvideo just use GST_DEBUG_CATEGORY_EXTERN()
	  to declare and use those, but that's just not right at
	  all, and it won't work on Windows with MSVC. Instead look
	  up the categories via the API.

2016-02-20 10:05:17 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/audio/audio.def:
	* gst-libs/gst/audio/audio.vcproj:
	* gst-libs/gst/audio/audiofilter.vcproj:
	* gst-libs/gst/riff/riff.def:
	* gst-libs/gst/riff/riff.vcproj:
	* gst-libs/gst/video/video.vcproj:
	* gst/adder/adder.vcproj:
	* gst/audioconvert/audioconvert.vcproj:
	* gst/audiorate/audiorate.vcproj:
	* gst/tcp/tcp.vcproj:
	* gst/typefind/typefindfunctions.vcproj:
	* gst/videoconvert/videoconvert.vcproj:
	* gst/videorate/videorate.vcproj:
	* gst/videoscale/videoscale.vcproj:
	* gst/videotestsrc/videotestsrc.vcproj:
	* gst/volume/volume.vcproj:
	* win32/MANIFEST:
	* win32/vs6/grammar.dsp:
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstadder.dsp:
	* win32/vs6/libgstaudio.dsp:
	* win32/vs6/libgstaudioconvert.dsp:
	* win32/vs6/libgstaudiorate.dsp:
	* win32/vs6/libgstaudioresample.dsp:
	* win32/vs6/libgstaudioscale.dsp:
	* win32/vs6/libgstaudiotestsrc.dsp:
	* win32/vs6/libgstdecodebin.dsp:
	* win32/vs6/libgstdecodebin2.dsp:
	* win32/vs6/libgstdirectsound.dsp:
	* win32/vs6/libgstfft.dsp:
	* win32/vs6/libgstgdp.dsp:
	* win32/vs6/libgstinterfaces.dsp:
	* win32/vs6/libgstogg.dsp:
	* win32/vs6/libgstpbutils.dsp:
	* win32/vs6/libgstplaybin.dsp:
	* win32/vs6/libgstriff.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstsdp.dsp:
	* win32/vs6/libgstsinesrc.dsp:
	* win32/vs6/libgstsubparse.dsp:
	* win32/vs6/libgsttag.dsp:
	* win32/vs6/libgsttheora.dsp:
	* win32/vs6/libgsttypefindfunctions.dsp:
	* win32/vs6/libgstvideo.dsp:
	* win32/vs6/libgstvideorate.dsp:
	* win32/vs6/libgstvideoscale.dsp:
	* win32/vs6/libgstvideotestsrc.dsp:
	* win32/vs6/libgstvolume.dsp:
	* win32/vs6/libgstvorbis.dsp:
	* win32/vs7/gst-plugins-base.sln:
	* win32/vs7/libgstadder.vcproj:
	* win32/vs7/libgstaudio.vcproj:
	* win32/vs7/libgstaudioconvert.vcproj:
	* win32/vs7/libgstaudiorate.vcproj:
	* win32/vs7/libgstaudioresample.vcproj:
	* win32/vs7/libgstaudiotestsrc.vcproj:
	* win32/vs7/libgstdecodebin.vcproj:
	* win32/vs7/libgstinterfaces.vcproj:
	* win32/vs7/libgstogg.vcproj:
	* win32/vs7/libgstplaybin.vcproj:
	* win32/vs7/libgstriff.vcproj:
	* win32/vs7/libgstsubparse.vcproj:
	* win32/vs7/libgsttag.vcproj:
	* win32/vs7/libgsttcp.vcproj:
	* win32/vs7/libgsttheora.vcproj:
	* win32/vs7/libgsttypefind.vcproj:
	* win32/vs7/libgstvideo.vcproj:
	* win32/vs7/libgstvideorate.vcproj:
	* win32/vs7/libgstvideoscale.vcproj:
	* win32/vs7/libgstvideotestsrc.vcproj:
	* win32/vs7/libgstvolume.vcproj:
	* win32/vs7/libgstvorbis.vcproj:
	* win32/vs8/gst-plugins-base.sln:
	* win32/vs8/libgstadder.vcproj:
	* win32/vs8/libgstaudio.vcproj:
	* win32/vs8/libgstaudioconvert.vcproj:
	* win32/vs8/libgstaudiorate.vcproj:
	* win32/vs8/libgstaudioresample.vcproj:
	* win32/vs8/libgstaudiotestsrc.vcproj:
	* win32/vs8/libgstdecodebin.vcproj:
	* win32/vs8/libgstinterfaces.vcproj:
	* win32/vs8/libgstogg.vcproj:
	* win32/vs8/libgstplaybin.vcproj:
	* win32/vs8/libgstriff.vcproj:
	* win32/vs8/libgstsubparse.vcproj:
	* win32/vs8/libgsttag.vcproj:
	* win32/vs8/libgsttcp.vcproj:
	* win32/vs8/libgsttheora.vcproj:
	* win32/vs8/libgsttypefind.vcproj:
	* win32/vs8/libgstvideo.vcproj:
	* win32/vs8/libgstvideorate.vcproj:
	* win32/vs8/libgstvideoscale.vcproj:
	* win32/vs8/libgstvideotestsrc.vcproj:
	* win32/vs8/libgstvolume.vcproj:
	* win32/vs8/libgstvorbis.vcproj:
	  win32: remove outdated build cruft
	  This hasn't been touched for generations, doesn't work,
	  and is just causing confusion. We also don't want to
	  maintain these files manually.

2016-02-19 12:38:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

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=== release 1.7.2 ===

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2016-02-19 11:48:30 +0200  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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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.prerequisites:
	* 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-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-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/audio-enumtypes.c:
	* win32/common/audio-enumtypes.h:
	* win32/common/config.h:
	* win32/common/video-enumtypes.c:
	  Release 1.7.2
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2016-02-19 10:31:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* 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/hr.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:
	  po: Update translations

2016-02-18 14:31:28 +0000  Julien Isorce <j.isorce@samsung.com>

	* pkgconfig/gstreamer-allocators-uninstalled.pc.in:
	* pkgconfig/gstreamer-app-uninstalled.pc.in:
	* pkgconfig/gstreamer-audio-uninstalled.pc.in:
	* pkgconfig/gstreamer-fft-uninstalled.pc.in:
	* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-riff-uninstalled.pc.in:
	* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
	* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
	* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
	* pkgconfig/gstreamer-tag-uninstalled.pc.in:
	* pkgconfig/gstreamer-video-uninstalled.pc.in:
	  uninstalled.pc: add support for non libtool build systems
	  Currently the .la path is provided which requires to use libtool as
	  mentioned in the GStreamer manual section-helloworld-compilerun.html.
	  It is fine as long as the application is built using libtool.
	  So currently it is not possible to compile a GStreamer application
	  within gst-uninstalled with CMake or other build system different
	  than autotools.
	  This patch allows to do the following in gst-uninstalled env:
	  gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
	  gstreamer-video-1.0)
	  Previously it required to prepend libtool --mode=link
	  https://bugzilla.gnome.org/show_bug.cgi?id=720778

2016-01-22 18:26:01 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: strengthen check for valid H.263 picture layer
	  Avoids some false positives leading to miss identification:
	  * Prevent picture start code emulation for the first 2 bytes read
	  * Add check for valid "picture coding type" and "PB-frames mode" combination
	  Additionally, change name on confusingly named TR var to what
	  it is, the layer's PTYPE.
	  https://bugzilla.gnome.org/show_bug.cgi?id=693263

2015-11-23 15:06:02 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: return incomplete topology if decode chains' cap could not be obtained
	  When getting caps of the decode chain, in get_topology, the caps are being
	  checked if fixed or not. But get_topology will be called when the decode is
	  chain is being exposed and hence it will always be fixed. Hence removing the
	  check for fixed caps. Removing gst_pad_get_current_caps for the chain->pad, as
	  get_pad_caps will again call the same api.
	  And get_topology can return NULL value if currently shutting down the
	  pipeline, which on being passed to create message will result in assertion
	  error. Check if topology is valid before using it
	  https://bugzilla.gnome.org/show_bug.cgi?id=755918

2016-02-05 10:10:40 +0100  Havard Graff <havard.graff@gmail.com>

	* gst-libs/gst/Makefile.am:
	  rtp: build audio library before rtp
	  Because audio-enumtypes.h needs to be available for
	  gstrtpbaseaudiopayload.c
	  https://bugzilla.gnome.org/show_bug.cgi?id=761949

2016-02-15 21:28:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Fix documentation of the autoplug-query signal

2016-01-26 13:54:46 +0100  Stian Selnes <stian@pexip.com>

	* gst-libs/gst/video/gstvideoencoder.c:
	* tests/check/libs/videoencoder.c:
	  videoencoder: Fix leak when pre_push does not return OK
	  https://bugzilla.gnome.org/show_bug.cgi?id=761951

2016-02-11 19:47:04 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioresample/resample.c:
	  resample: avoid overflows
	  Avoid overflow in rate calculation. This can cause the resampler to
	  start on the wrong phase after a rate change.
	  Avoid overflow in cubic fraction calculation. This can cause noise when
	  dealing with higher samplerates.

2016-02-11 18:01:40 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioresample/resample_sse.h:
	  resample: fix double interpolation sse code
	  We were only reading 2 filter taps and we need to read 4 to do cubic
	  interpolation.

2016-02-10 12:48:15 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: make a copy if we can't write in unpack
	  If we don't have writable memory, make sure to make a copy of the input
	  samples into a temporary (writable) buffer, even if we are dealing with
	  a native intermediate format that we don't need to call the unpack
	  function for.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=761655

2016-02-05 19:15:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/Makefile.am:
	  tests: extend the AM_TESTS_ENVIRONMENT from check.mak
	  To get the CK_DEFAULT_TIMEOUT defined for all tests.
	  Also replaces a 120 timeout that was set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761472

2016-02-05 18:03:07 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From 86e4663 to b64f03f

2016-01-21 09:43:35 +0100  Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>

	* ext/pango/gstbasetextoverlay.c:
	* ext/pango/gstbasetextoverlay.h:
	  textoverlay: Expose rendering dimensions as properties.
	  In order to detect graphical user input on the
	  textoverlay, the resulting rendering properties
	  need to be exposed to applications.
	  Fixes delayx property declaration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761251

2016-01-20 15:37:44 +0100  Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>

	* ext/pango/gstbasetextoverlay.c:
	  textoverlay: Do not limit positioning to video area.
	  The current position property is limited to X,Y positions
	  in the range of [0, 1]. This patch allows full control
	  over the overlay position, including partially outside
	  of the video area.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761251

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2016-02-03 16:28:42 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/opus/gstopusdec.c:
	  opus: fix FEC
	  FEC may only be used when PLC is enabled on the audio decoder,
	  as it relies on empty buffers to generate audio from the next
	  buffer. Hooking to the gap events doesn't work as the audio
	  decoder does not like more buffers output than it sends.
	  The length of data to generate using FEC from the next packet
	  is determined by rounding the gap duration to nearest. This
	  ensures that duration imprecision does not cause quantization
	  to 2.5 milliseconds less than available. Doing so causes the
	  Opus API to fail decoding. Such duration imprecision is common
	  in live cases.
	  The buffer to consider when determining the length of audio
	  to be decoded is the previous buffer when using FEC, and the
	  new buffer otherwise. In the FEC case, this means we determine
	  the amount of audio from the previous buffer, whether it was
	  missing or not (and get the data either from this buffer, or
	  the current one if the previous one was missing).

2016-02-02 15:20:48 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* ext/opus/gstopusdec.c:
	  opusdec: fix wrong buffer being checked for missing data
	  This caused a decoding error if the resulting (wrong) buffer size
	  was passed to the Opus decoding API.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758158

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2016-01-28 13:29:39 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiorate/gstaudiorate.c:
	  audiorate: Use gst_audio_format_fill_silence() instead of memset with 0 for generating silence
	  For unsigned formats, silence is not all bits 0.

2016-01-28 13:21:33 +0100  HoonHee Lee <hoonhee.lee@lge.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/video/gstvideodecoder.c:
	  audio/videodecoder: Minor cleanup of last commit
	  https://bugzilla.gnome.org/show_bug.cgi?id=761218

2016-01-28 18:06:44 +0900  HoonHee Lee <hoonhee.lee@lge.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/video/gstvideodecoder.c:
	  audio/videodecoder: use gst_pad_peer_query_caps to make output caps
	  gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
	  In that case, use gst_pad_peer_query_caps() with template caps as filter
	  to have negotiated output caps properly before forwarding GAP event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761218

2016-01-26 19:23:04 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/encoding/gstencodebin.c:
	  encodebin: Allow streamheader update when profile.allow_dynamic_output == FALSE
	  Some encoders can update the stream header through time (for example
	  vp8 might do that) but it does not strictly changes the output format.

2016-01-26 14:09:42 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst-libs/gst/video/video-format.h:
	  video-format: fix GstVideoFormatInfo documentation warnings
	  Add missing ':' to tile_ws and tile_hs fields documentation to avoid
	  bad render of these two fields, mark reserved bytes as private to hide
	  field and avoid gtkdoc warning and add parameters description to
	  documented macro to avoid gtkdoc warnings.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761132

2016-01-26 16:56:57 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* win32/common/libgstaudio.def:
	  audio-converter: add reset function

2016-01-26 16:36:41 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: handle NULL input
	  Allow NULL as input to mean silence samples.

2016-01-26 17:16:52 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: improve _update_config
	  Allow NULL config to keep the existing parameters.
	  Fix the docs.

2016-01-26 17:14:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	  audio-converter: audio-converter: make some optimized functions
	  Make optimized functions for generic and passthrough conversion.

2016-01-26 16:34:35 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-quantize.c:
	* gst-libs/gst/audio/audio-quantize.h:
	  audio-quantize: add _reset function
	  Add a reset function that clears any history.

2016-01-25 17:40:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* m4/Makefile.am:
	* m4/freetype2.m4:
	* tests/examples/Makefile.am:
	  build: remove nonsensical check for freetype
	  The examples need Gtk+, nothing uses freetype directly.

2016-01-25 16:22:17 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/libvisual.c:
	  tests: libvisual: make run faster
	  Reduce resolution, which shouldn't make any difference
	  to what's tested here. Makes test finish in less than
	  half the time it took before (8s vs. 21s).

2016-01-25 18:30:30 +0530  Arun Raghavan <git@arunraghavan.net>

	* ext/alsa/gstalsasink.c:
	  alsa: Trivial doc update
	  alsasink now does more than just raw audio.

2016-01-21 18:30:40 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Correctly expose pads from elements that have directly exposable pads
	  analyze_new_pad() can return a new decode chain, which might have a new
	  GstDecodePad in the end. We should use those two for expose_pad() and not the
	  original ones that were passed to analyze_new_pad().
	  This fails when having a demuxer element that has raw pads immediately or
	  if a decoder with raw caps is after an adaptive demuxer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760949

2016-01-21 16:08:46 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: ensure correct alignment of samples
	  Make sure that the data we allocate for our temporary buffers is
	  properly aligned.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=760938

2016-01-21 10:45:40 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/video/video-color.c:
	* gst-libs/gst/video/video-color.h:
	  video-color: add Adobe RGB primaries and transfer function

2016-01-20 10:19:34 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/video/video-info.c:
	  video-info: enfore RGB matrix for RGB formats
	  In gst_video_info_to_caps(), make sure we end up with an RGB matrix for
	  RGB formats and warn when the GstVideoInfo colorimetry is wrong.
	  In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
	  for RGB formats and warn about inconsistent caps.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-20 10:02:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/video/video-converter.c:
	  video-converter: ignore matrix for RGB formats
	  For RGB formats, the matrix in the colorimetry (conversion from YUV to
	  RGB) is irrelevant and we should ignore it and assume the identity
	  transform for everything we do.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-19 23:26:57 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/video/gstvideoencoder.h:
	  videoencoder: Deprecate GST_VIDEO_ENCODER_FLOW_DROPPED
	  It was never actually supported or used
	  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-19 23:22:35 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/video/gstvideoencoder.c:
	  Revert "videoencoder: Release video frame when ->handle return ERROR or DROPPED"
	  This reverts commit 63517d0ed348784cce4ab4b295c2c0f1b78baa81.
	  It was wrong ref counting wise and we decided to deprecated DROPPED
	  return value
	  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-18 11:40:36 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* tests/check/elements/audioconvert.c:
	  tests:audioconvert: Fix integer overflow build error
	  value of 32768L << 16 and 1L << 31 is 2147483648
	  but it exceeds the positive range of int which is 2147483647
	  resulting in integer overflow error. Use G_GINT64_CONSTANT instead of L.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760769

2016-01-19 12:39:22 +0530  Arun Raghavan <git@arunraghavan.net>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: Minor documentation cleanup

2016-01-14 23:14:27 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-play.c:
	  tools: gst-play: allow setting of flags in serialized foo+bar format
	  https://bugzilla.gnome.org/show_bug.cgi?id=751901

2015-07-02 17:58:00 +0200  Hugues Fruchet <hugues.fruchet@st.com>

	* tools/gst-play.c:
	  tools: gst-play: add command line options for verbose output and playbin flags
	  https://bugzilla.gnome.org/show_bug.cgi?id=751901

2016-01-18 15:51:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* win32/common/libgstapp.def:
	  win32: Update exports

2015-10-15 10:38:16 -0400  Evan Callaway <evan.callaway@ipconfigure.com>

	* gst-libs/gst/app/gstappsink.c:
	* gst-libs/gst/app/gstappsink.h:
	  Add WAIT_ON_EOS flag to gstappsink.
	  If set, an appsink that receives an EOS will wait until all of its buffers have been processed before continuing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756187

2016-01-16 10:17:50 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: Add note to the documentation about various settings being reset before set_format()
	  It's quite unexpected behaviour that various subclass settings are just
	  reset before set_format(). Unfortunately changing this now has the risk
	  of breaking existing code but we should reconsider this for 2.0.

2016-01-09 04:35:23 +0100  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Ignore flushing streams [..]
	  [..] when resetting group start time. In GES, we are usually connected
	  to the streamsynchronizer on one audio and one video pad.
	  When seeking the timeline, both nlecompositions often output their flush_start
	  before any of them has output its flush_stop.
	  The current code, when receiving the first flush stop was using the
	  running time of the start of the second composition, which could
	  be pretty much anything, and means nothing at that point.
	  This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
	  both when setting flushing and when checking it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750013

2016-01-08 18:53:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaybin2.c:
	  playbin: Only append non-raw and sysmem pad template caps to the autoplug-query result
	  Otherwise a decoder supporting GL memory will think that all downstream can
	  support GL memory because of seeing its own template caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758212

2016-01-08 18:37:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaybin2.c:
	  Revert "playbin: only add the template caps when the result is empty"
	  This reverts commit 023af2d3b192f8ebf1bd4fe75a22a4adaedc1e05.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758212

2016-01-15 13:35:22 +0000  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/video/gstvideoencoder.c:
	  videoencoder: Release video frame when ->handle return ERROR or DROPPED
	  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-15 09:50:29 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Properly mark pending blocked pads
	  When blocking input pads, we also need to properly set the appropriate
	  pending flag.
	  Without this, when switching stream types after initial configuration
	  (like going from Audio+Video to Audio+Video+Sub) playsink would never
	  wait for *all* input streams to be blocked (it would just wait for the
	  new input pad (text in this case) to be blocked).
	  Since the reconfiguration might introduce unlinking/relinking of elements,
	  we need to ensure that *ALL* input streams are blocked.
	  Failure to do so would result in having some input streams pushing data
	  to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
	  (returning GST_FLOW_NOT_LINKED).
	  A later optimization could involve only blocking the input pads that
	  might be involved in reconfiguration. But better be safe than sorry for
	  now :)

2016-01-06 10:12:43 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  gst-device-monitor: Use g_printerr instead of g_error
	  g_error is meant to be used for programmer errors (causes an abort),
	  not for expected runtime errors.

2016-01-13 16:32:25 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: replace gst_caps_can_intersect() with is_subset()
	  Subset check verifies also that all required fields are present
	  and is mostly commonly used when checking if an element accepts
	  a certain caps

2016-01-12 11:31:50 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/playback/gstplaybin2.c:
	  playbin: use subset check instead of intersect
	  Elements usually require that all fields on their caps are present
	  on the fixed caps they receive. Using intersection won't verify it,
	  resort to using is_subset() checks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760477

2016-01-12 15:56:36 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channel-mixer.c:
	  audio-channel-mixer: round before truncating
	  Round the result before truncating for int channel mixing.

2016-01-12 15:27:16 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: Avoid conversion when possible
	  When the input and output formats are the same and in a possible
	  intermediate format, avoid unpack and pack.
	  Never do passthrough channel mixing.
	  Only do dithering and noise shaping in S32 format

2016-01-12 11:43:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channel-mixer.c:
	  audio-channel-mixer: add more formats
	  Add support for float and int16 mixing
	  Remove in-place processing, this simplifies things as we won't be using it.
	  Don't do clipping for float audio formats

2016-01-12 11:37:17 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: improve processing loop
	  Process as many samples as we can from the input and return the number
	  of processed samples from the chain. This simplifies some code.
	  Fix the IN_WRITABLE handling, don't overwrite the flags.

2016-01-11 18:24:48 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: replace accept-caps with caps query
	  Those accept caps are actually checking if downstream supports
	  some particular caps to check if it need to negotiate a different
	  format. Checking only the next element with accept-caps is not enough
	  to guarantee that it is supported.
	  Using a caps query makes it obtain the supported caps for downstream
	  as a whole instead of only the next element.

2016-01-08 21:27:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* win32/common/libgstaudio.def:
	  audio: Update exported symbols list

2016-01-08 15:05:38 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/videorate/gstvideorate.c:
	  videorate: replace accept-caps with a caps query
	  accept-caps is only a shallow check, it needs to know
	  whether downstream as a whole accepts the framerate

2016-01-08 16:08:47 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: fix up for GstAudioChannelMix rename as well

2016-01-08 17:34:50 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio-converter: small API tweaks
	  Pass flags in _converter_new() so that we can configure ourselves
	  differently depending on some options.
	  SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'

2016-01-08 17:28:31 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	  audio-converter: prepare API for rate changes
	  Use the update function to update the sample rates along with the config
	  once we implement resampling.

2016-01-08 17:17:44 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio-convert: simplify API
	  Simplify the API, we don't need the consumed and produced output
	  arguments. The caller needs to use the _get_in_frames/get_out_frames API
	  to check how much input is needed and how much output will be produced.

2016-01-08 17:50:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudioutilsprivate.h:
	* gst-libs/gst/video/gstvideoutilsprivate.h:
	  audio/video: Use G_GNUC_INTERNAL for internal functions

2016-01-08 16:22:25 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio-channel-mixer.c:
	* gst-libs/gst/audio/audio-channel-mixer.h:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio.h:
	* win32/common/libgstaudio.def:
	  audio: GstAudioChannelMix -> GstAudioChannelMixer
	  Rename the GstAudioChannelMix object to GstAudioChannelMixer because it
	  looks better and to avoid a conflict with a library in -bad.

2016-01-07 15:24:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaybin2.c:
	  playbin: Use the caps query instead of accept-caps to detect if a sink accepts caps
	  accept-caps is only for one element, caps query is recursive. Fixes playback
	  with totem and other situations.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760234

2016-01-06 15:49:59 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst-libs/gst/video/gstvideopool.c:
	  videopool: store videoinfo after choosing the biggest buffer size
	  Otherwise, pool could be negotiated with a size which will be different
	  from the one used in allocation which is the GstVideoInfo.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760222

2016-01-06 12:14:39 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst/videotestsrc/gstvideotestsrc.c:
	  videotestsrc: add missing break in set_property switch case
	  To avoid future issue when adding new properties.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760204

2016-01-06 01:04:31 +0000  Koop Mast <kwm@FreeBSD.org>

	* tests/check/elements/audioconvert.c:
	  tests: audioconvert: fix test compilation with clang
	  With clang 3.7.1 on FreeBSD:
	  elements/audioconvert.c:650:12: error: shifting a negative signed value is
	  undefined [-Werror,-Wshift-negative-value]
	  (-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
	  ~~~ ^
	  https://bugzilla.gnome.org/show_bug.cgi?id=760134

2016-01-06 01:06:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/libs/audiodecoder.c:
	* tests/check/libs/audioencoder.c:
	* tests/check/libs/rtp.c:
	* tests/check/libs/rtpbasepayload.c:
	  tests: fix indentation of various unit tests

2016-01-05 22:52:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: add new audio API

2016-01-03 17:21:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/sdp/gstmikey.h:
	* gst-libs/gst/video/video-overlay-composition.h:
	  docs: remove dummy function declarations with G_INLINE_FUNCTION for gtk-doc
	  gtk-doc can handle static inline functions just fine these days,
	  there's no need for this stuff any more.

2016-01-03 10:33:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/riff/riff-ids.h:
	  riff: Add missing closing parenthesis to GST_RIFF_WAVE_FORMAT_ANTEX_ADPCME
	  Apparently this #define is unused.

2016-01-02 23:29:22 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
	  riff-ids: remove trailing whitespace

2016-01-02 23:27:44 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
	  riff-ids: fix two swapped ids
	  For these fourcc ids the name and value is swapped. This was causing a warning
	  when registering the avi ids.

2015-12-31 20:43:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/Makefile.am:
	  sdp: Also reorder SUBDIRS to try even harder to build the RTP library first

2015-12-31 20:41:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/Makefile.am:
	  sdp: The SDP library depends on the RTP library now and is not independent anymore
	  Fix up the build dependencies.

2015-10-07 18:50:18 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/sdp/Makefile.am:
	* gst-libs/gst/sdp/gstmikey.c:
	* gst-libs/gst/sdp/gstmikey.h:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	* gst-libs/gst/sdp/gstsdpmessage.h:
	* tests/check/libs/sdp.c:
	* win32/common/libgstsdp.def:
	  sdp: add helper fuctions from/to sdp from/to caps
	  <gstsdpmessage.h>
	  GstCaps*       gst_sdp_media_get_caps_from_media   (const GstSDPMedia *media, gint pt);
	  GstSDPResult   gst_sdp_media_set_media_from_caps   (const GstCaps* caps, GstSDPMedia *media);
	  gchar *        gst_sdp_make_keymgmt                (const gchar *uri, const gchar *base64);
	  GstSDPResult   gst_sdp_message_attributes_to_caps  (GstSDPMessage *msg, GstCaps *caps);
	  GstSDPResult   gst_sdp_media_attributes_to_caps    (GstSDPMedia *media, GstCaps *caps);
	  <gstmikey.h>
	  GstMIKEYMessage * gst_mikey_message_new_from_caps  (GstCaps *caps);
	  gchar *           gst_mikey_message_base64_encode  (GstMIKEYMessage* msg);
	  https://bugzilla.gnome.org/show_bug.cgi?id=745880

2015-12-29 18:14:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
	  In this specific case it wouldn't cause problems as we only ever access the
	  first array element, but let's make explicit what is happening here.
	  CID 1346530 and 1346529

2015-12-29 17:56:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Check for FALSE'ness directly, not by comparing with FALSE

2015-12-29 17:54:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Don't use preset_name string after free
	  When we run the loop for another time and do not have a preset name, we would
	  try to print the preset name of a previous iteration that is already freed.
	  Also move some other variables into the block where they are actually used
	  to prevent similar mistakes in the future.
	  CID 1346536

2015-12-29 14:40:04 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/audioconvert.c:
	  audioconvert: add a test for gap handling

2015-12-29 14:23:59 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst-libs/gst/audio/audio-converter.c:
	* tests/check/elements/audioconvert.c:
	  audioconvert: fix passthrough operation
	  We did not take the sample size into account. Rearrange the tests to have more
	  conversion test and an extra test case for passthrough operations.
	  Fixes #759890

2015-12-29 11:29:31 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-device-monitor.c:
	  tools: gst-device-monitor: print uint properties in both decimal and hex
	  Some values are easier to read and make sense of in hex.
	  https://bugzilla.gnome.org//show_bug.cgi?id=759780

2015-11-12 14:01:03 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst-libs/gst/video/video-blend.c:
	  videoblend: special case 1x1 src dims on increment computation
	  Fix crash with 1x1 overlay pixmap
	  https://bugzilla.gnome.org/show_bug.cgi?id=757290

2015-12-28 12:28:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
	  We would otherwise read beyond the array bounds and crash every now and then.
	  This was introduced with 5640ba17c8db80976b7718904e4024dcfe9ee1a0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759910

2015-12-27 19:41:43 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/audioconvert.c:
	  tests: remove commented code from audioconvert test
	  This is just what we have in gst_check_buffer_data().

2015-12-27 19:25:20 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: code cleanup
	  Rename samples to num_samples, since we also have samples in chain, but that is
	  the data pointer. Always use gzize for num_samples. Make the log output a bit
	  more homogenous.

2015-12-26 11:34:47 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-device-monitor.c:
	  tools: gst-device-monitor: print non-string device properties too

2015-12-26 09:43:56 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/audio-channel-mix.c:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-quantize.c:
	  audio: Fix some documentation warnings
	  Remove/rename function parameters and skip some functions that can't
	  be used by bindings as they are now.

2015-12-26 09:43:51 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideoaffinetransformationmeta.c:
	  videoaffinetransformmeta: Add (transfer none) annotation for return value

2015-12-25 11:34:10 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Don't leak audio/video filters due to floating references weirdness
	  The filters' floating references are sinked during set_property() already,
	  which means that GstBin takes a new reference when adding the filter to it.
	  Get rid of the additional reference after adding the filter to the bin.

2015-12-25 10:36:44 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Allow reuse of audio/video filters by unparenting them from their bins
	  And also recreate the chains if the filter is changing.

2015-12-25 10:28:02 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Don't leak audio/video filters when using non-raw media

2015-12-24 15:27:43 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

2015-12-24 13:59:52 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/Makefile.am:
	  pbutils: Link to libgstbase for bytewriter and adapter

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=== release 1.7.1 ===

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2015-12-24 13:59:15 +0100  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* 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-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-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/audio-enumtypes.c:
	* win32/common/audio-enumtypes.h:
	* win32/common/config.h:
	* win32/common/pbutils-enumtypes.c:
	* win32/common/pbutils-enumtypes.h:
	  Release 1.7.1

2015-12-24 13:10:08 +0100  Sebastian Dröge <sebastian@centricular.com>

	* 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/hr.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
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2015-12-24 12:22:04 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/nl.po:
	* po/sv.po:
	* po/zh_CN.po:
	  po: Update translations

2015-12-11 15:38:00 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encodebin: Implement an encoding profile serialization format
	  https://bugzilla.gnome.org/show_bug.cgi?id=759356

2015-12-21 00:43:49 +0100  Koop Mast <kwm@rainbow-runner.nl>

	* configure.ac:
	  configure: Make -Bsymbolic check work with clang.
	  Update the -Bsymbolic check with the version glib has. This version
	  works with clang.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759713

2015-12-03 11:53:05 +0900  Kazunori Kobayashi <kkobayas@igel.co.jp>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: Clear is_eos flag when receiving the flush-stop event
	  The EOS event can be propagated to the downstream elements when
	  is_eos flag remains set even after leaving the flushing state.
	  This fix allows this element to normally restart the streaming
	  after receiving the flush event by clearing the is_eos flag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759110

2015-12-16 18:11:05 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/examples/playback/playback-test.c:
	  examples: playback-test: remove unused variables
	  audiosink and videosink string variables are unused

2015-11-30 10:28:55 +1100  Matthew Waters <matthew@centricular.com>

	* gst/playback/gstplaybin2.c:
	  playbin: only add the template caps when the result is empty
	  Unconditionally adding the template caps when proxying the caps query will play
	  havoc with decoders that attempt to choose an output format based on some caps
	  features.  Creating a sink that does not include those caps features and a
	  decoder/parser/etc that preferentially chooses some specific caps feature when
	  available, will always return the decoder/parser/etc template caps and choose a
	  feature that downstream will be unable to support.
	  Fix by limiting the addition of the template caps to when the result is actually
	  empty.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758212

2015-12-17 13:39:01 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  configure: Don't use AG_GST_CHECK_FEATURE for checking for gio-unix-2.0
	  It's meant to be used for external plugins that can then all be disabled via
	  --disable-external. gio-unix-2.0 however is just an optional dependency for
	  the TCP unit test.
	  Also when using AG_GST_CHECK_FEATURE like this, in the --disable-external part
	  there needs to be an AM_CONDITIONAL for the feature with FALSE.

2015-12-16 17:07:54 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  Revert "decodebin2: fix deadlock on chain shutdown"
	  This reverts commit 77dc09c3a9a5e5e371e189f39b5557db440a8dc9.
	  It can cause the FLUSH_START/STOP events to go to the sink elements, which
	  then causes state changes and various other problems. We shouldn't really
	  flush downstream here, the idea is to do *draining*.
	  Apart from that the testcase for the original bug here works without this
	  commit now.

2015-12-16 11:12:00 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/tcp/gstmultifdsink.c:
	  multifdsink: fix typo in GST_WARNING_OBJECT
	  This should make easier to parse the debug logs.
	  s/fnctl/fcntl

2014-04-10 15:36:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: remove dead code
	  Since the loops increasing count from 0 are always run at least
	  once (if count < 1), count will always be at least one when
	  compared to the drop/dup conditions.
	  Coverity 1139674

2015-12-16 10:45:48 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* win32/common/libgstaudio.def:
	  audio-converter: rework the main processing loop
	  Rework the main processing loop. We now create an audio processing
	  chain from small core functions. This is very similar to how the
	  video-converter core works and allows us to statically calculate an
	  optimal allocation strategy for all possible combinations of operations.
	  Make sure we support non-interleaved data everywhere.
	  Add functions to calculate in and out frames and latency.

2015-12-16 10:44:16 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: clear convert object

2015-12-16 09:35:38 +0100  Sebastian Dröge <sebastian@centricular.com>

	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/gst-plugins-base-plugins.hierarchy:
	* docs/plugins/gst-plugins-base-plugins.signals:
	* docs/plugins/inspect/plugin-adder.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-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:
	  docs: update to git

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2015-12-14 11:09:46 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/opus/gstopusdec.c:
	* ext/opus/gstopusenc.c:
	  plugins-bad: Fix example pipelines
	  rename gst-launch --> gst-launch-1.0
	  replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
	  fix caps in examples
	  https://bugzilla.gnome.org/show_bug.cgi?id=759432

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2015-12-14 13:59:02 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/alsa/gstalsasrc.c:
	  Revert "alsasrc: Disable HW timestamp"
	  This reverts commit 3642e9a3913a35c00f379034780c27298d09929c.

2015-11-10 12:54:23 -0500  Xavier Claessens <xavier.claessens@collabora.com>

	* gst-libs/gst/allocators/gstfdmemory.h:
	* gst-libs/gst/app/gstappsink.h:
	* gst-libs/gst/app/gstappsrc.h:
	* gst-libs/gst/audio/audio-info.h:
	* gst-libs/gst/audio/gstaudiobasesink.h:
	* gst-libs/gst/audio/gstaudiobasesrc.h:
	* gst-libs/gst/audio/gstaudiocdsrc.h:
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	* gst-libs/gst/audio/gstaudioencoder.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/audio/gstaudioringbuffer.h:
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.h:
	* gst-libs/gst/pbutils/encoding-profile.h:
	* gst-libs/gst/pbutils/encoding-target.h:
	* gst-libs/gst/pbutils/gstdiscoverer.h:
	* gst-libs/gst/pbutils/install-plugins.h:
	* gst-libs/gst/rtp/gstrtpbaseaudiopayload.h:
	* gst-libs/gst/rtp/gstrtpbasedepayload.h:
	* gst-libs/gst/rtp/gstrtpbasepayload.h:
	* gst-libs/gst/rtsp/gstrtspurl.h:
	* gst-libs/gst/sdp/gstmikey.h:
	* gst-libs/gst/sdp/gstsdpmessage.h:
	* gst-libs/gst/tag/gsttagdemux.h:
	* gst-libs/gst/tag/gsttagmux.h:
	* gst-libs/gst/video/colorbalancechannel.h:
	* gst-libs/gst/video/gstvideodecoder.h:
	* gst-libs/gst/video/gstvideoencoder.h:
	* gst-libs/gst/video/gstvideofilter.h:
	* gst-libs/gst/video/gstvideopool.h:
	* gst-libs/gst/video/gstvideosink.h:
	* gst-libs/gst/video/gstvideoutils.h:
	* gst-libs/gst/video/video-info.h:
	* gst-libs/gst/video/video-overlay-composition.h:
	  base: Add g_autoptr() support to all types
	  https://bugzilla.gnome.org/show_bug.cgi?id=754464

2015-09-24 18:26:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	  alsasrc: Disable HW timestamp
	  This is a workaround for broken pulse module.

2015-12-14 19:03:33 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Properly initialize stack-allocated RTSP message to all-zeroes

2015-12-14 10:57:19 -0500  Evan Callaway <evan.callaway@ipconfigure.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Use relative URI for non-proxy tunneled requests
	  Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we
	  are using a proxy server. Also, send Host header for compatability with
	  HTTP/1.1 and some HTTP/1.0 servers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758922

2015-12-14 09:10:16 -0500  Evan Callaway <evan.callaway@ipconfigure.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	* win32/common/libgstrtsp.def:
	  rtspconnection: Support authentication during tunneling setup
	  gst_rtsp_connection_connect_with_response accepts a response pointer
	  which it fills with the response from setup_tunneling if the
	  connection is configured to be tunneled.  The motivation for this is to
	  allow the caller to inspect the response header to determine if
	  additional authentication is required so that the connection can be
	  retried with the appropriate authentication headers.
	  The function prototype of gst_rtsp_connection_connect has been
	  preserved for compatability with existing code and wraps
	  gst_rtsp_connection_connect_with_response.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749596

2015-12-14 13:11:21 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepayload: Check if the packet loss event actually has timestamp and duration fields
	  CID 1139615

2015-12-10 17:46:26 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channel-mix.c:
	* gst-libs/gst/audio/audio-channel-mix.h:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-quantize.c:
	* gst-libs/gst/audio/audio-quantize.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio: adapt API for non-interleaved formats
	  Allow an array of sample blocks to be passed to the channel mix and
	  quantizer functions to support non-interleaved formats.

2015-12-10 16:26:40 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	  audio-converter: improve API for non-interleaved formats
	  Make it possible to pass an array of sample blocks when dealing with
	  non-interleaved formats.

2015-12-12 17:49:28 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: add FourCC aliases
	  Support media using the aliases defined in http://www.fourcc.org/ that are
	  exact duplicates of already known codes.

2015-12-12 17:04:21 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: use defined FourCC
	  Make gst_riff_create_video_caps() use the FourCC available in riff-ids.h,
	  like gst_riff_create_audio_caps() does.

2015-12-11 14:42:09 +0000  Julien Isorce <j.isorce@samsung.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: add some debug around pool negotiation
	  It lets us know easily which pool is activated or
	  inactivated during the negotiation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720597

2015-12-11 21:42:00 +0800  Song Bing <b06498@freescale.com>

	* gst-libs/gst/video/convertframe.c:
	  video/convertframe: Add crop meta support via videocrop
	  https://bugzilla.gnome.org/show_bug.cgi?id=759329

2015-12-11 11:01:53 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepay: when setting discont flag make sure rtpbuffer is current
	  Depayloaders will look at rtpbuffer->buffer for the discont flag.
	  When we set the discont flag on a buffer in the rtp base depayloader
	  and we have to make the buffer writable, make sure the rtpbuffer
	  actually contains the newly-flagged buffer, not the original input
	  buffer. This was introduced with the addition of the process_rtp_packet
	  vfunc, but would only trigger if the input buffer wasn't flagged
	  already and was not writable already.

2015-12-11 00:18:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/libs/rtpbasedepayload.c:
	  tests: rtpbasedepayload: add test for seqnum gap discont setting
	  The problem was triggered only when the input buffers were not
	  writable, so add extra ref to test this code path.

2015-12-11 10:25:00 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepay: fix possible refcounting issue when detecting a discont
	  When we detect a discont and the input buffer isn't already flagged
	  as discont, handle_buffer() does a gst_buffer_make_writable() on the
	  input buffer in order to set the flag. This assumed it had ownership
	  of the input buffer though, which it didn't. This would still work
	  fine in most scenarios, but could lead to crashes or mini object
	  unref criticals in some cases when a discont is detected, e.g. when
	  using pcapparse in front of a depayloader. This problem was
	  introduced in bc14cdf529e.

2015-12-10 12:18:04 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstmultisocketsink.c:
	* gst/tcp/gstmultisocketsink.h:
	  multisocketsink: add GstNetworkMessage event
	  Add a property and logic to send a GstNetworkMessage event containing
	  the message that was received from a client. This can be used to
	  implement simply bidirectional communication.

2015-12-10 12:14:37 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstmultisocketsink.c:
	* gst/tcp/gstmultisocketsink.h:
	  multisocketsink: add dispatched event
	  Add a property and logic to send a GstNetworkMessageDispatched
	  event upstream to notify that a buffer has been sent. This can be used
	  to keep track of what client received what buffers.

2015-12-04 11:17:37 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstsocketsrc.c:
	* gst/tcp/gstsocketsrc.h:
	  socketsrc: handle GstNetworkMessage events
	  Add a property to handle GstNetworkMessage events. These events contain
	  a buffer that is sent on the socket to allow for simple bidirectional
	  communication.

2015-12-09 17:16:26 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio-convert: improve converter API
	  Improve the converter API to allow for an max input and output number of
	  samples and return the number of consumed/produced samples.

2015-12-08 11:15:34 +0100  Philippe Normand <philn@igalia.com>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: duration query support based on the size property
	  https://bugzilla.gnome.org/show_bug.cgi?id=759126

2015-12-07 09:08:05 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From b319909 to 86e4663

2015-12-04 12:25:11 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstmultisocketsink.c:
	  multisocketsink: let downstream know we support metadata
	  Let downstream know that we support GstNetControlMessage metadata API.

2015-12-03 16:38:45 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Avoid pushing buffers before segment start
	  In the case where the stream doesn't have a framerate set and the frames
	  don't have a duration set, we still want to use the clipping path to
	  make sure we don't push buffers outside of the segment.
	  The problem was the previous iteration was setting a duration of 2s, which
	  meant that any buffer which was less than 2s before the segment start would
	  end up getting pushed.
	  Instead, use a saner 40ms (25fps single frame duration) to figure out whether
	  the frame could be within the segment or not

2015-12-02 20:19:43 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst-libs/gst/allocators/Makefile.am:
	* gst-libs/gst/app/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/fft/Makefile.am:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/sdp/Makefile.am:
	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/video/Makefile.am:
	  Drop usage of deprecated g-ir-scanner --strip-prefix flag

2015-12-02 18:16:05 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: fix "Attempt to unlock mutex that was not locked"
	  Introduced in commit ee44337f, caused the decodebin
	  test_text_plain_streams unit test to abort.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752651

2015-11-16 14:50:58 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstrawcaps.h:
	  playback: Expose XSUB formats by default
	  This is a workaround, we should remove this once we have a proper
	  decoder

2015-11-16 14:50:30 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Also consider XSUB as a subtitle format

2015-11-16 14:49:55 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/descriptions.c:
	  pbutils: Add description for XSUB subpicture format

2015-11-16 14:49:19 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: 'DXSA' is the same as 'DXSB'
	  Which is subpicture/x-xsub

2015-07-21 09:58:56 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Rename GstStream => GstSyncStream
	  Avoid clashes with future GstStream from core

2015-12-02 09:00:31 -0500  Evan Callaway <evan.callaway@ipconfigure.com>

	* gst-libs/gst/rtsp/gstrtspdefs.c:
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	  rtspconnection: Update capitalization of x-sessioncookie
	  Some servers incorrectly parse header names with strict case-sensitivity.  For
	  compatibility with these systems change X-Sessioncookie to x-sessioncookie.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758921

2015-12-02 16:16:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Update buffering messages when removing an element that had buffering pending
	  Otherwise we'll remove that element while keeping its buffering message in our
	  list, and because of that never ever report buffering 100% as that element
	  will always be at a lower percentage.
	  This fixes e.g. seeking over Period boundaries in DASH and various other
	  issues when buffering happens between group switches.
	  Also use a new mutex for protecting the buffering messages. The object lock is
	  already used by gst_object_has_as_ancestor() and we need to use it now for
	  checking if the buffering message sender has the to-be-removed element as
	  ancestor.

2015-12-02 09:52:19 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstmultisocketsink.c:
	* gst/tcp/gstmultisocketsink.h:
	  multisocketsink: keep on reading when we stop sending
	  When we stop sending because we need more data, still keep a GSource
	  around to receive data from the clients.
	  Also handle read and write in the same go.

2015-12-01 19:57:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiobasesrc.c:
	  audiobasesrc: Post latency message on the bus after set_caps()
	  The latency is only known once the caps are known, and might change
	  whenever the caps are changing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758911

2015-09-25 14:47:48 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst-libs/gst/audio/gstaudiobasesink.c:
	  audiobasesink: Post latency message on the bus after set_caps()
	  Any latency query before this will not get the correct latency so a new
	  latency query should be triggered once the audio sink know its own latency.
	  Without this the initial latency query from the pipeline arrives too early
	  sometimes and the resulting latency is too short.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758911

2015-11-06 14:21:14 +0000  Thomas Bluemel <tbluemel@control4.com>

	* gst/playback/gstdecodebin2.c:
	  [PATCH] Fix a race condition accessing the decode_chain field.
	  Make sure that any access to the GstDecodeBin's decode_chain
	  field is protected using the EXPOSE_LOCK.  Also add a simple
	  reference counter to the GstDecodeChain structure so that when
	  the type_found signal fires it can hold onto the decode chain
	  even while the EXPOSE_LOCK is not held.  This should fix a
	  race condition if the type_found signal fires right in the
	  middle of a state change that messes with the same decode
	  chain.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755260

2015-08-20 17:30:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin: early out on pad-added when the pad is inactive
	  The pad may be recently deactivated if the element is switched
	  back down very quickly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752651

2015-08-20 17:29:36 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin: lock the expose lock around decode_chain use
	  Helps with a crash in decodebin when quickly switching states.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752651

2015-11-28 14:24:55 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: accept wrong version field in OpusHead header
	  Some Opus files found on the wild have 0 in the version field of the
	  OpusHead header, instead of the correct value of 1. The files still
	  play, don't make this error fatal.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758754

2015-11-26 11:33:02 +0000  William Manley <will@williammanley.net>

	* gst-libs/gst/allocators/gstfdmemory.c:
	  allocators: add debug category for fd memory and allocator
	  Debugging can now be viewed by setting GST_DEBUG=fdmemory:9
	  https://bugzilla.gnome.org/show_bug.cgi?id=758744

2015-11-20 20:18:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/libs/tag.c:
	  tests: tags: add unit test for ID3v2 PRIVATE_DATA tag extraction
	  https://bugzilla.gnome.org/show_bug.cgi?id=730926

2014-09-29 14:17:39 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst-libs/gst/tag/gstid3tag.c:
	* gst-libs/gst/tag/id3v2frames.c:
	  id3v2frames: Handle private frames
	  Handle PRIV ID3 tag having owner information (string)
	  and binary data, add to tag messages list.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730926

2015-11-20 19:15:22 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/tag/id3v2.c:
	  tags: id3: make sure to register private-id3v2-frame tag before using it

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2015-11-17 15:23:17 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/opus/gstopusenc.c:
	  Remove unnecessary NULL checks before g_free()
	  g_free() is NULL-safe

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2015-11-17 17:07:37 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* tests/check/libs/rtspconnection.c:
	  rtspconnection: Add support for parsing custom headers
	  https://bugzilla.gnome.org/show_bug.cgi?id=758235

2015-11-15 02:58:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	* gst-libs/gst/pbutils/encoding-target.c:
	* gst-libs/gst/rtsp/gstrtspmessage.c:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	* tests/examples/encoding/encoding.c:
	  Remove unnecessary NULL checks before g_free()
	  g_free() is NULL-safe

2015-11-17 09:06:34 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.c:
	  xvimagesink/ximagesink: Fix structure memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=758204

2015-11-12 14:39:17 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: guint8 can't hold value over 255
	  channels is a guint8, so the max value is 255 and checking if it value is
	  > 256 will never be false.
	  CID 1338687, CID 1338688

2015-11-12 14:18:03 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: remove unneeded check for unsigned < 0
	  Commit ff6d1a2a25b247688f38e117782a6b43d525706a changed sample's type from
	  gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
	  which means it can never be a negative value and the check making sure that
	  in_samples is >= 0 is never going to be false. Removing it.
	  CID 1338689

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2015-11-12 12:21:54 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* ext/opus/gstopusenc.c:
	  opusenc: avoid potential overflow expression
	  The result of the two expressions will be promoted to guint64 anyway,
	  perform all the arithmetic in 64 bits to avoid potential overflows.
	  CID 1338690, CID 1338691

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2015-11-11 14:44:55 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* tests/check/libs/video.c:
	  tests:video: Fix overlay rectangle and buffer leak
	  Created overlay rectangle is not being freed in video tests
	  pix2 buffer is being created and not freed
	  https://bugzilla.gnome.org/show_bug.cgi?id=757927

2015-11-11 14:37:21 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst-libs/gst/pbutils/encoding-target.c:
	  pbutils:encoding-target: Fix string memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=757926

2015-11-11 15:02:39 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst-libs/gst/audio/audio-quantize.c:
	  audio-quantize: Fix dither_buffer memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=757928

2015-11-11 00:59:16 +1100  Jan Schmidt <jan@centricular.com>

	* ext/vorbis/gstvorbisdec.c:
	  vorbisdec: Re-init on new caps
	  If we get new input caps, then reset the decoder
	  ready for new headers and fresh data. Makes
	  chained oggs work when reusing the decoder.

2015-11-02 23:12:19 +1100  Matthew Waters <matthew@centricular.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/video/Makefile.am:
	* gst-libs/gst/video/gstvideoaffinetransformationmeta.c:
	* gst-libs/gst/video/gstvideoaffinetransformationmeta.h:
	* win32/common/libgstvideo.def:
	  videometa: add GstVideoAffineTransformationMeta
	  Adds a simple 4x4 affine transformations meta for passing arbitrary
	  transformations on buffers.
	  Based on patch by Matthieu Bouron
	  https://bugzilla.gnome.org/show_bug.cgi?id=731791

2015-11-10 09:52:24 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio-converter: add output size argument
	  Make it possible to have a different number of output samples than input
	  samples when we, for example, want to add resampling later.

2015-11-07 00:43:55 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Check API arguments and assert if needed

2015-11-06 19:31:47 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Properly deactivate ghostpads
	  Just setting the ghostpad as flushing wasn't enough. It needs to be
	  consistent on the internal proxypad also, otherwise you end up in
	  situations where:
	  * a pending buffer on the target pad triggers the sticky event
	  propagation
	  * the default implementation sees that the proxypad is not flushing,
	  so it tries to push it to the other pad (the actual ghostpad)
	  * the ghostpad is flushing, so returns FALSE
	  * the push_event function sees that pushing the event failed...
	  * ... and pending buffer push returns GST_FLOW_ERROR, instead of
	  GST_FLOW_FLUSHING
	  By using gst_pad_set_active(FALSE), we ensure that both the ghostpad
	  and the proxypad are flushing/deactivated. The situation above will
	  no longer occur, and a GST_FLOW_FLUSHING will be returned.

2015-11-06 18:11:41 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioconvert/gstaudioconvertorc-dist.c:
	* gst/audioconvert/gstaudioconvertorc-dist.h:
	* gst/audioconvert/gstaudioconvertorc.orc:
	* gst/audioconvert/plugin.c:
	  audioconvert: fix build
	  Don't include file that is no longer generated, and remove some
	  files that are no longer needed because they have moved into the
	  lib. Fixes distcheck.

2015-11-06 18:00:41 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: require interleaved samples and no resampling
	  We can't yet do resampling or anything other than interleaved audio.

2015-11-06 17:54:21 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/gstaudiopack-dist.c:
	* gst-libs/gst/audio/gstaudiopack-dist.h:
	  audio: update ORC dist files

2015-11-06 17:49:00 +0100  Wim Taymans <wtaymans@redhat.com>

	* docs/plugins/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiopack.orc:
	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/gstaudioconvert.h:
	* tests/check/Makefile.am:
	* win32/common/libgstaudio.def:
	  audio-converter: move audio converter to audio libs
	  Move the audio-converter helper to the audio library.

2015-11-06 17:39:33 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio-channel-mix.c:
	* gst-libs/gst/audio/audio-channel-mix.h:
	* gst-libs/gst/audio/audio.h:
	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* win32/common/libgstaudio.def:
	  audio-channel-mix: move channel mixer to audio libs
	  Move the channel mixer code to the audio library

2015-11-06 17:29:22 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channels.c:
	* gst-libs/gst/audio/audio-info.c:
	* gst-libs/gst/audio/audio.c:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	  audio: add debug categories

2015-11-06 16:42:35 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  channelmix: don't limit channelpositions
	  Don't set a limit on the channel positions, just like the metadata.

2015-11-06 16:03:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  channelmix: simplify API a little
	  Remove the format and layout from the mix_samples function and use the
	  format when creating the channel mixer object. Also use a flag to handle
	  the unlikely case of non-interleaved samples like we do elsewhere.

2015-11-06 15:50:34 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  channelmix: GstChannel -> GstAudioChannel
	  Rename GstChannel to GstAudioChannel

2015-11-06 13:02:19 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-quantize.c:
	* gst-libs/gst/audio/audio-quantize.h:
	  audio-quantize: update docs
	  Update docs
	  Add another flag for the quantizer

2015-11-06 12:46:36 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstaudioconvertorc.orc:
	* gst/audioconvert/gstchannelmix.c:
	  audioconvert: cleanups and add some docs
	  Add docs for the internal audioconvert object before moving it to the
	  audio library.
	  Remove get_sizes and implement the trivial logic in the element.
	  Remove some unused orc functions

2015-11-06 12:46:12 +0100  Wim Taymans <wtaymans@redhat.com>

	* win32/common/libgstaudio.def:
	  defs: update defs

2015-11-06 12:37:14 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/gstaudiopack-dist.c:
	* gst-libs/gst/audio/gstaudiopack-dist.h:
	  audio: update orc files

2015-11-06 12:10:48 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio-quantize.c:
	* gst-libs/gst/audio/audio-quantize.h:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiopack.orc:
	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstaudioconvert.h:
	* gst/audioconvert/gstfastrandom.h:
	  audioconvert: move audio quantize code to libs
	  Move the audio quantize code from audioconvert to the audio library.
	  work on making an audio converter helper function similar to the video
	  converter.
	  Fold fastrandom directly into the quantizer, add some ORC code to
	  optimize this later.

2015-11-05 12:42:56 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channels.c:
	* gst-libs/gst/audio/audio-channels.h:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst/audioconvert/gstaudioconvert.c:
	* win32/common/libgstaudio.def:
	  audio-channels: rename get_default_mask
	  Rename _get_default_mask() to _get_fallback_mask() to make it more
	  clear that the function only provides a fallback if nothing else can be
	  done. Also clarify this in the documentation.
	  API: gst_audio_channel_get_fallback_mask()

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2015-11-05 12:11:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusdec.c:
	  opusdec: Update sink pad templates
	  We always require the channel-mapping-field. If it's 0 we require nothing
	  else, otherwise we need channels, stream-count and coupled count to be
	  available.

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2015-11-05 11:34:07 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/volume/gstvolume.c:
	  volume: Do not try to get binding value array if we are not processing any sample
	  In some conditions we might process empty buffers, calling
	  gst_control_binding_get_value_array in that case will lead
	  to the assertion:
	  (lt-ges-launch-1.0:18859): GStreamer-CRITICAL **: gst_control_binding_get_value_array: assertion 'values' failed

2015-11-05 10:40:18 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channels.c:
	* gst-libs/gst/audio/audio-channels.h:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst/audioconvert/gstaudioconvert.c:
	* win32/common/libgstaudio.def:
	  audio-channels: make method to get default channel-mask
	  Add a new method to get the default channel-mask.
	  Use the new method on audiodecoder and audioconvert.
	  API: gst_audio_channel_get_default_mask()

2014-11-10 11:11:37 +0100  Andreas Frisch <fraxinas@opendreambox.org>

	* tests/check/libs/video.c:
	  tests: Add a test for video blending over transparent frames
	  And fix the test_overlay_blend test where we blend over a
	  transparent frame and where expecting wrong results
	  https://bugzilla.gnome.org/show_bug.cgi?id=681447

2013-11-30 01:59:55 +0100  Arnaud Vrac <avrac@freebox.fr>

	* gst-libs/gst/video/video-blend.c:
	  video: blend using OVER operation
	  Also support all premultiplied/non-premultiplied source/destination
	  configurations
	  https://bugzilla.gnome.org/show_bug.cgi?id=681447

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2015-11-04 00:12:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/opus.c:
	  opus: Remove invalid unit test
	  Opus headers should never be in-band, so don't test for correct
	  handling of that.

2015-11-04 00:12:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	  opusenc: Create an empty taglist if there is none
	  There always have to be 2 buffers in the streamheaders, even if
	  the comment buffer is basically empty.

2015-11-03 14:50:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/Makefile.am:
	* ext/opus/gstopusdec.c:
	* ext/opus/gstopusdec.h:
	* ext/opus/gstopusenc.c:
	* ext/opus/gstopusheader.c:
	* ext/opus/gstopusheader.h:
	  opus: Add proper support for multichannel audio
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-02 17:33:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusdec.c:
	  opusdec: Handle GstAudioClippingMeta instead of the pre-skip field in the OpusHead
	  oggdemux is outputting the meta now, and only outputs if it should really
	  apply to the current buffer. Previously we would skip N samples also if we
	  started the decoder in the middle of the stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-02 16:52:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	  opusenc: Add GstAudioClippingMeta to buffers that need to be clipped
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-02 10:30:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	  opusenc: Disable granule position calculations by the base class
	  It is doing the wrong thing because of the Opus pre-skip: while the timestamps
	  are shifted by the pre-skip, the granule positions are not shifted.
	  oggmux is doing the right thing here already.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-10-31 15:02:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	  opusenc: Add some FIXME comments about calculating padding with LPC
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-10-30 20:57:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	* ext/opus/gstopusenc.h:
	  opusenc: Encode exactly the amount of samples we got as input and put correct timestamps on it
	  The first frame has lookahead less samples, the last frame might have some
	  padding or we might have to encode another frame of silence to get all our
	  input into the encoded data.
	  This is because of a) the lookahead at the beginning of the encoding, which
	  shifts all data by that amount of samples and b) the padding needed to fill
	  the very last frame completely.
	  Ideally we would use LPC to calculate something better than silence for the
	  padding to make the encoding as smooth as possible.
	  With this we get exactly the same amount of samples again in an
	  opusenc ! opusdec pipeline.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-10-30 20:47:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	* ext/opus/gstopusheader.c:
	* ext/opus/gstopusheader.h:
	  opusenc: Put lookahead/pre-skip into the OpusHead header
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

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2015-11-03 16:51:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/ogg/gstoggstream.c:
	  oggdemux: Create full Opus caps with all fields
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-03 18:30:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/pbutils/codec-utils.c:
	* gst-libs/gst/pbutils/codec-utils.h:
	* win32/common/libgstpbutils.def:
	  codec-utils: Add utilities for Opus caps and the OpusHead header
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-03 11:11:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/ogg/gstoggmux.c:
	  oggmux: Use GstAudioClippingMeta for Opus for accurate end clipping
	  ... instead of relying on the segment. For the clipping at the start we assume
	  a proper value in the OpusHead, as generated by opusparse or opusenc.
	  Transmuxing in general is not guaranteed to produce the correct values, or
	  even have a OpusHead (e.g. when having RTP input).
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-03 10:58:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/ogg/Makefile.am:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstoggstream.h:
	  oggdemux: Add GstAudioClippingMeta for Opus for accurate start/end clipping
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-02 16:19:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiometa.c:
	* gst-libs/gst/audio/gstaudiometa.h:
	* win32/common/libgstaudio.def:
	  audio: Add GstAudioClippingMeta for specifying clipping on encoded audio buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-02 11:19:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstoggstream.h:
	  oggdemux: Allow start clipping for Opus
	  The granulepos does not have the pre-skip subtracted while timestamps do,
	  and the last granulepos will be shorter by the number of samples that should
	  be dropped because of padding in the end.
	  As such, extrapolating the granule of the beginning of the first frame will
	  lead to a negative value, which is not a problem but intentional.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-03 16:38:09 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/audio/gstaudiopack-dist.c:
	* gst-libs/gst/audio/gstaudiopack-dist.h:
	  audio: update disted orc backup files

2015-11-03 14:08:25 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/audio/gstaudioclock.c:
	  audioclock: use GST_STIME_FORMAT for GstClockTimeDiff
	  GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
	  handle negative values better.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-03 13:44:39 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Print GstClockTimeDiff as a signed integer in debug logs

2015-11-03 11:59:09 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-format.c:
	* gst-libs/gst/audio/audio-format.h:
	* gst-libs/gst/audio/gstaudiopack.orc:
	* gst/audioconvert/audioconvert.c:
	  audio-format: add TRUNCATE_RANGE flag
	  Add a TRUNCATE_RANGE flag for unpack functions to fill the least
	  significate bits with 0 (as did the old code). Also add functions
	  that don't truncate. Use the TRUNC flag in audioconvert for
	  backwards compatibility for now.

2015-11-03 11:57:32 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/gstaudiopack.orc:
	  audiopack: improve pack functions
	  Avoid shifts by using convh functions.

2015-11-03 11:44:54 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstaudioconvertorc.orc:
	* tests/check/elements/audioconvert.c:
	  audioconvert: change multiplier for int<->float conversion
	  Use (1 << 31) as the multiplier for int<->float conversions. This makes
	  sure that int->float conversions always end up with floats between
	  [-1.0, 1.0].
	  For the conversion from float to int, this multiplier will give the complete
	  int range after we perform clipping.
	  Change the unit test to take this into consideration.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755301

2015-11-02 17:32:55 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/audio/gstaudiobasesink.c:
	  audiobasesink: use GST_STIME_ARGS for GstClockTimeDiff
	  No need to use G_GINT64_FORMAT for potentially negative values of
	  GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
	  Plus it creates more readable values in the logs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-02 16:36:35 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* ext/ogg/gstoggmux.c:
	  oggmux: Print GstClockTimeDiff as a signed integer in debug logs

2015-11-02 16:09:52 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: Use GstClockTimeDiff and print signed integer in debug logs
	  Use GstClockTimeDiff and Clock macros to print signed integer time
	  differences in the debug logs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-02 14:06:39 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* tests/examples/seek/scrubby.c:
	  examples: use GST_STIME_FORMAT for GstClockTimeDiff
	  GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
	  handle negative values better.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-02 17:14:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiometa.h:
	  audio: Fix parameters to gst_buffer_get_audio_downmix_meta() in macro

2015-11-02 15:54:19 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	  audiotestsrc: increase freq limit
	  Raise the frequency limit and try to negotiate to a samplerate of 4*freq
	  when larger then the default samplerate.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=754450

2015-11-02 15:46:22 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	  audiotestsrc: add support for unlimited number of channels
	  Raise the channel limit and set the channel-mask for > 2 channels.

2015-11-02 13:19:09 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/audiotestsrc/gstaudiotestsrc.h:
	  audiotestsrc: add support for all formats
	  Use the pack functions to also support the other audio formats we
	  have.

2015-11-02 12:09:42 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: subtract time difference with GST_CLOCK_DIFF
	  To ensure the subtraction of two GstClockTime values (which are guint64)
	  can be negative. Use GST_CLOCK_DIFF which returns a gint64.
	  CID 1338049

2015-11-02 11:34:56 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Do not force user to provide an encoding profile name
	  And use the profile called `default` if none provided.

2015-11-02 11:30:07 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-target.c:
	  encoding-target: Do not unconditionally break when searching for a target
	  Otherwise the loop is useless!
	  Fixes CID 1338051

2015-10-24 20:08:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioresample/gstaudioresample.c:
	  audioresample: Clip input buffers to the segment before handling them
	  https://bugzilla.gnome.org/show_bug.cgi?id=757068

2015-10-24 20:05:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: Clip input buffers to the segment before handling them
	  https://bugzilla.gnome.org/show_bug.cgi?id=757068

2015-10-24 20:02:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiofilter.c:
	  audiofilter: Clip input buffers to the segment before handling them
	  https://bugzilla.gnome.org/show_bug.cgi?id=757068

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2015-11-01 23:34:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusdec.c:
	  opusdec: Assume 48kHz if no sample rate is given in the header

2015-10-30 20:59:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	  opusenc: Place 48kHz first in the caps
	  For all the other sample rates the encoder will have to resample internally.

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2015-11-01 23:05:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioconvert/gstaudioconvertorc-dist.c:
	* gst/audioconvert/gstaudioconvertorc-dist.h:
	  audioconvert: update orc backup code to fix build without orc

2015-10-26 21:32:41 +0100  Csaba Toth <tocsanti@gmail.com>

	* gst/tcp/gstmultisocketsink.c:
	  multisocketsink: fix "client-removed" signal on 64-bit platforms and with bindings
	  The client-removed signal used G_INT_TYPE instead of G_SOCKET_TYPE
	  in its definition leading to problems on platforms where the size
	  of a pointer is larger than the size of an integer, It would also
	  not work at all with dynamic language bindings.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757155

2015-10-28 18:36:41 +0100  Joan Pau Beltran <joanpau.beltran@socib.cat>

	* gst/videotestsrc/gstvideotestsrc.c:
	  videotestsrc: fix handling of Bayer format 'gbrg'
	  Due to a typo, videotestsrc did not handle the Bayer
	  format 'gbrg' properly and reported it as invalid,
	  causing negotiation errors.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757264

2015-10-30 17:36:48 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvertorc.orc:
	* gst/audioconvert/gstaudioquantize.c:
	* gst/audioconvert/gstaudioquantize.h:
	  audioconvert: rework audioconvert
	  Rewrite audioconvert to try to make it more clear what steps are
	  executed during conversion.
	  Add passthrough step that just does a memcpy when possible.
	  Add ORC optimized dither and quantization functions.
	  Implement noise-shaping on S32 samples only and allow for arbitrary
	  noise shaping coefficients if we want this later.

2015-10-30 17:33:32 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  channelmix: fix up API a little
	  don't use gpointer * for something that should be gpointer.

2015-10-28 11:40:42 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstaudioquantize.c:
	  audioquantize: make helper for add with saturation

2015-10-29 16:52:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Print another time difference as a signed integer instead of a huge unsigned one

2015-10-29 16:01:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Print GstClockTimeDiff as a signed integer in debug logs

2015-10-29 00:01:01 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  tools: gst-device-monitor: fix two memory leaks
	  The removed GList link needs to be freed too, and
	  the G_OPTION_REMAINING arguments need to be freed.

2015-10-28 15:50:44 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-target.c:
	  encoding-target: Add a GST_ENCODING_TARGET_PATH envvar to find target files

2015-10-28 15:47:00 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-target.c:
	  encoding-target: Allow having encoding target without a category set
	  There was already some code to handle that, but the support was not
	  complete in those code paths.

2015-10-27 12:56:48 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-target.c:
	  encoding-target: Create directory before trying to save encoding targets

2015-10-27 12:50:26 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Allow specifying the target category in the serialized encoding target

2015-10-27 17:28:06 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstaudioconvert.h:
	* gst/audioconvert/gstaudioquantize.c:
	* gst/audioconvert/gstaudioquantize.h:
	  audioconvert: make the quantizer a reusable object
	  Turn the quantizer into a reusable object.

2015-10-27 13:24:31 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  audioconvert: make the channel mixer a separate reusable object
	  A first attempt at making the channel mixer a separate object.

2015-10-28 11:32:57 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstaudioquantize.c:
	  audioquantize: fix 8-pole noise shaping
	  Fix the 8-pole noise shaping error update. We were mixing errors from
	  different channels.

2015-10-27 15:44:06 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Send SEEK events directly to adaptive streaming demuxers
	  This makes sure that they will always get SEEK events, even if we're currently
	  in the middle of a group switch (i.e. switching to another
	  representation/bitrate/etc).
	  https://bugzilla.gnome.org/show_bug.cgi?id=606382

2015-10-06 15:20:51 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin: fix event leak
	  As stated in GST_PAD_PROBE_HANDLED's documentation, we are
	  supposed to unref the event before returning.
	  Fixes an event leak in the validate.hls.playback.play_15s.hls_bibbop
	  validate scenario.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754459

2015-10-23 19:13:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioconvert/gstaudioconvertorc-dist.c:
	* gst/audioconvert/gstaudioconvertorc-dist.h:
	  audioconvert: Update disted orc files

2015-10-23 16:58:17 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvertorc.orc:
	* gst/audioconvert/gstaudioquantize.c:
	* gst/audioconvert/gstchannelmix.c:
	  audioconvert: use pack/unpack functions
	  Rework the converter to use the pack/unpack functions
	  Because the unpack functions can only unpack to 1 format, add a separate
	  conversion step for doubles when the unpack function produces int.
	  Do conversion to S32 in the quantize function directly.
	  Tweak the conversion factor for doing float->int conversion slightly to
	  get the full range of negative samples, use clamp to make sure we don't
	  exceed our int range on the positive axis (see also #755301)

2015-10-23 12:02:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaybin2.c:
	  playbin: Send upstream events directly to playsink
	  Send event directly to playsink instead of letting GstBin iterate
	  over all sink elements. The latter might send the event multiple times
	  in case the SEEK causes a reconfiguration of the pipeline, as can easily
	  happen with adaptive streaming demuxers.
	  What would then happen is that the iterator would be reset, we send the
	  event again, and on the second time it will fail in the majority of cases
	  because the pipeline is still being reconfigured

2015-10-23 17:25:50 +0900  Eunhae Choi <eunhae1.choi@samsung.com>

	* tests/check/gst/typefindfunctions.c:
	  tests: typefindfunctions: fix error leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=757008

2015-09-23 18:47:52 +0200  Thibault Saunier <tsaunier@gnome.org>

	* gst/videotestsrc/gstvideotestsrc.c:
	  videotestsrc: Force alpha downstream if foreground color contains alpha
	  Otherwise the foreground color won't be fully represented in the
	  outputted frames.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755482

2015-10-22 12:07:44 +0800  Pavel Bludov <pbludov@gmail.com>

	* gst-libs/gst/video/video-overlay-composition.h:
	  video: overlay-composition: fix rectangle and composition cast macros
	  Closing parenthesis was missing in two cases.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756893

2015-10-21 14:34:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* common:
	  Automatic update of common submodule
	  From b99800a to b319909

2015-10-20 17:29:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Use new GST_ENABLE_EXTRA_CHECKS #define
	  https://bugzilla.gnome.org/show_bug.cgi?id=756870

2015-10-21 14:25:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* README:
	* common:
	  Automatic update of common submodule
	  From 9aed1d7 to b99800a

2015-10-20 12:08:23 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbuffer.h:
	  rtp: GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is Since 1.6.1

2015-10-20 03:58:26 +1100  Matthew Waters <matthew@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: track the exposable pads through connect_pad
	  The logic introduced by
	  [d50b713: decodebin: set the decode pad target before setting elements to PAUSED]
	  to expose pads would only ever be able to possibly expose one (the last) pad per element.
	  Make it so that any exposable pads are able to be exposed rather than just the
	  last pad returned by connect_element.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742924

2015-10-20 03:52:24 +1100  Matthew Waters <matthew@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: return the possibly new chain in analyze_new_pad
	  In the case of analyzing a demuxer chain, analyze_new_pad may create
	  a new GstDecodeChain.  This was not propagated to the calling function which as
	  of [d50b713f decodebin: set the decode pad target before setting elements to PAUSED]
	  is now required to be able to expose the correct pad.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742924

2015-10-19 15:32:19 +0530  Rajat Verma <rajat.verma@st.com>

	* gst/playback/gstplaysink.c:
	  playsink: relink text_pad in case of reconfiguration
	  In case of reconfiguration, text_pad should be re-connected with
	  stream synchronizer sink pad. Otherwise we'll leave an unlinked pad around if
	  there always was a streamsynchronizer text pad.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756804

2015-09-14 15:25:11 +0900  eunhae choi <eunhae1.choi@samsung.com>

	* gst-libs/gst/audio/gstaudiobasesink.c:
	  audiobasesink: fix issue about eos handling during flushing
	  If the flush-start is arrived during _eos_wait() in basesink,
	  the 'eos' flag is overwritten to TRUE after exiting the _eos_wait().
	  To resolve the overwritten issue,
	  the subclass doing the _eos_wait() call should return the right value.
	  If the eos flag is set to TRUE again, it will cause error(enter the eos flow)
	  of the following state changing from PAUSED to PLAYING in basesink.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754980

2015-10-17 22:25:22 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gstplaysink.c:
	* gst/playback/gstsubtitleoverlay.c:
	  decodebin/playbin/playsink/subtitleoverlay: Post async-done on state change failures
	  https://bugzilla.gnome.org/show_bug.cgi?id=756611

2015-10-17 22:20:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Immediately error out if state change fails
	  Otherwise we chain up to the parent class' change_state function and might
	  override the failure with SUCCESS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756611

2015-10-17 21:47:07 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaybin2.c:
	* gst/playback/gsturidecodebin.c:
	  playbin/uridecodebin: Always post async-done immediately if we're a live pipeline
	  Not only if the base class told us, but also if one of our own elements did.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756611

2015-10-16 03:40:43 +1100  Matthew Waters <matthew@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: set the decode pad target before setting elements to PAUSED
	  Otherwise caps and context queries will disappear into nothing and therefore
	  fail.  With autoplug-query now actually working, users (such as playbin) can
	  proxy these queries to the selected video sink and be able to select an
	  more appropriate configuration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=731204

2015-10-17 20:36:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/video.c:
	  video: Add out annotations to the out parameters of gst_video_calculate_display_ratio()
	  https://bugzilla.gnome.org/show_bug.cgi?id=754567

2015-10-16 10:48:50 +1100  Matthew Waters <matthew@centricular.com>

	* win32/common/libgstrtp.def:
	  win32 update exports for new rtp symbols

2015-07-22 11:31:05 +0200  Stian Selnes <stian@pexip.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	* tests/check/libs/rtp.c:
	  rtpbuffer: Add map flag to skip padding
	  Encrypted RTP buffers may contain encrypted padding, hence it's
	  necessary to have an option to relax the validation in order to
	  successfully map the buffer.
	  When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
	  gst_rtp_buffer_map() will map the buffer like if padding is not
	  present.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752705

2015-10-15 22:40:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  Revert "rtpbuffer: increase logging level when map fails"
	  This reverts commit e3c8a820176ba39dfae85944fa9c6ae202ec681d.
	  It causes too much noise in the logs.

2015-10-15 15:32:58 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  rtpbuffer: increase logging level when map fails
	  https://bugzilla.gnome.org/show_bug.cgi?id=756641

2015-10-15 10:01:38 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/playback/gstplaysink.c:
	  playsink: Fix volume element leak
	  In case sink implements a streamvolume interface, volume element is being got
	  from the sink. But this is transfer full. So the memory should be freed before
	  setting it to NULL. This was resulting in major memory leaks
	  https://bugzilla.gnome.org/show_bug.cgi?id=755867

2015-10-14 00:32:11 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasrc.c:
	  alsa: Use 8 bit pointer type for byte-based pointer arithmetic
	  Usually these loops only run once, so there's no problem here. But sometimes
	  they run twice, and by adding the number of bytes to a 16 bit pointer type we
	  would advance twice as much as we should.
	  Also use snd_pcm_frames_to_bytes() in alsasrc to calculate
	  the number of bytes to skip, same as we do in alsasink.
	  Thanks to Lucio A. Hernandez <lucio.a.hernandez@gmail.com> for reporting.

2015-10-12 14:02:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudioencoder.c:
	* tests/check/libs/audioencoder.c:
	  Revert "audioencoder: timestamp headers same as first buffer and use duration 0"
	  This reverts commit dd4d6d9ed54c2a63a7e45661519d9965417707c5.
	  It breaks ogg muxing and the vorbisenc unit test.

2015-08-28 11:44:19 +0200  Havard Graff <havard.graff@gmail.com>

	* gst-libs/gst/audio/gstaudioencoder.c:
	* tests/check/libs/audioencoder.c:
	  audioencoder: timestamp headers same as first buffer and use duration 0
	  https://bugzilla.gnome.org/show_bug.cgi?id=754224

2015-08-28 11:25:22 +0200  Havard Graff <havard.graff@gmail.com>

	* tests/check/libs/audioencoder.c:
	  audioencoder-tests: port to use GstHarness
	  https://bugzilla.gnome.org/show_bug.cgi?id=754223

2015-08-27 17:28:30 +0200  Havard Graff <havard.graff@gmail.com>

	* tests/check/libs/audiodecoder.c:
	  audiodecoder-test: port to using GstHarness
	  https://bugzilla.gnome.org/show_bug.cgi?id=754196

2015-10-04 18:36:00 +0100  Sebastian Dröge <sebastian@centricular.com>

	* sys/xvimage/xvimagepool.c:
	  xvimagesink: Put error message into debug output instead of just throwing it away

2015-10-02 22:19:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  Update GLib dependency to 2.40.0

2014-03-15 17:35:56 +0100  Sebastian Rasmussen <sebras@hotmail.com>

	* gst-libs/gst/rtp/gstrtpbasepayload.c:
	* tests/check/libs/rtpbasepayload.c:
	  rtpbasepayload: Implement video SDP attributes
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726472

2015-09-25 15:17:53 +0300  Vivia Nikolaidou <vivia@toolsonair.com>

	* tools/gst-play.c:
	  gst-play: Removed erroneous comment
	  The "fall through" comment was wrong. Removed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755440

2015-09-22 23:12:10 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* tools/gst-play.c:
	  gst-play: Add keyboard shortcut '0' to seek to beginning
	  https://bugzilla.gnome.org/show_bug.cgi?id=755440

2015-08-25 16:24:12 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/videorate/gstvideorate.c:
	  videorate: remove unnecessary break statement
	  Trivial patch to remove unncessary break statement used after
	  goto statement.
	  https://bugzilla.gnome.o