+=== release 1.11.2 ===
+
+2017-02-24 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.11.2
+
+2017-02-24 10:04:21 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Give a name to the thread-pool threads
+ This way they can be distinguished from any other threads in the same
+ process.
+
+2017-02-24 10:02:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Only lock the thread pool mutex when running with more than 1 thread
+ There's no reason to lock anything if only the current thread is ever
+ going to do any work.
+
+2017-02-20 21:38:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ * gst/videoconvert/gstvideoconvert.c:
+ * gst/videoconvert/gstvideoconvert.h:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ video-converter: Implement multi-threaded scaling/conversion
+ This adds a property to select the maximum number of threads to use for
+ conversion and scaling. During processing, each plane is split into
+ an equal number of consecutive lines that are then processed by each
+ thread.
+ During tests, this gave up to 1.8x speedup with 2 threads and up to 3.2x
+ speedup with 4 threads when converting e.g. 1080p to 4k in v210.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778974
+
+2017-02-21 11:59:12 +0100 Georg Lippitsch <glippitsch@toolsonair.com>
+
+ * gst-libs/gst/video/gstvideotimecode.c:
+ * tests/check/libs/videotimecode.c:
+ videotimecode: Validate for drop-frame correctness
+ In gst_video_time_code_is_valid, also check for invalid
+ ranges when using drop-frame TC. Refactor some code which
+ broke after the check was added.
+ https://bugzilla.gnome.org/show_bug.cgi?id=779010
+
+2017-02-15 18:40:21 +0100 Georg Lippitsch <glippitsch@toolsonair.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideotimecode.c:
+ * gst-libs/gst/video/gstvideotimecode.h:
+ * tests/check/libs/videotimecode.c:
+ * win32/common/libgstvideo.def:
+ videotimecode: Init from GDateTime
+ Add a function to init the time code from a GDateTime
+ https://bugzilla.gnome.org/show_bug.cgi?id=778702
+
+2017-02-20 13:44:37 +0200 Jochen Henneberg <jh@henneberg-systemdesign.com>
+
+ * ext/vorbis/gstvorbiscommon.c:
+ vorbis: Fix channel reorder map for 5.1, 6.1 and 7.1
+
+2017-02-15 21:41:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: Fix upscaling if width & height change and we're starting not at y=0
+ It was taking the initial input y-offset from the output value, which
+ only works for y=0 (in which case both are the same). If y > 0, we would
+ always stay behind the requested input offset and never ever read
+ anything from the input.
+
+2017-02-14 22:31:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Fix distcheck
+ Buildbot doesn't like wildcards here for some reason.
+
+2017-02-14 19:44:43 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ meson: dist meson build files
+ Ship meson build files in tarballs, so people who use tarballs
+ in their builds can start playing with meson already.
+
+2017-02-14 19:43:47 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/MANIFEST:
+ win32: dist all .def files
+
+2017-02-10 17:32:29 +0900 Heekyoung Seo <heekyoung.seo@lge.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: prevent unsigned int overflow
+ https://bugzilla.gnome.org/show_bug.cgi?id=778432
+
+2017-02-10 21:28:49 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: ensure the discont flag is correct
+ Previously it happened that reused buffer caused the discont to be on the wrong
+ buffers.
+
+2017-02-08 11:42:45 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Ignore more parser related fields when comparing streams
+ The parser might do some conversion on a stream but the stream keeps
+ being the same, and we need to make sure GstDiscoverer detects it is the
+ case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778298
+
+2017-02-04 14:46:00 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: Also add FLAC to debug strings.
+ Oops, also add FLAC to the debug strings array.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777655
+
+2017-02-04 14:42:33 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: Prevent overflow of debug names array
+ Add new audio types to the list of strings used for debug
+ so we don't index past the end of that array.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777655
+
+2017-02-02 14:56:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/multifdsink.c:
+ multifdsink: Make sure to use a 64 bit integer for the units-max property
+
+2017-01-20 19:49:46 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3-parse.c:
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Fix crash caused by slot double freeing
+ Probe for MultiQueue source pad might receive EOS twice,
+ the first is fake-eos and the other is actual EOS.
+ And the slot can be freed with fake-eos/EOS if the slot has no input.
+ Since slot freeing is async, double free can be possible.
+ So, decodebin3 needs to remove the probe also with slot freeing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777530
+
+2017-01-31 16:47:32 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/examples/decodebin_next/playbin-test.c:
+ examples: Fix leak
+
+2016-12-03 13:38:28 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Fix list leak on handle_stream_switch()
+ Free no more used list variables
+ https://bugzilla.gnome.org/show_bug.cgi?id=775553
+
+2016-12-03 13:22:54 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Change requested_selection to have its own memory for stream-id
+ "requested_selection" list might be generated by select-streams event.
+ And memory of stream-id(s) in select-streams is independent from that of stream-collection.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775553
+
+2016-12-03 12:47:41 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Change return types of stream_in_{list,collection}
+ Change return types of functions to get memory address of stream-id.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775553
+
+2016-12-03 12:43:22 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstplaybin3.c:
+ playback: Fix leak on select_streams
+ Since gst_event_parse_select_streams() returns newly allocated
+ memory for stream-id(s), it should be freed explicitly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775553
+
+2017-01-02 15:12:47 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/encodebin.c:
+ encodebin: fix caps leak in test
+ https://bugzilla.gnome.org/show_bug.cgi?id=776797
+
+2017-01-30 12:35:04 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86-sse41.c:
+ audio-resampler: Fix integer overflow in clamping code
+ https://bugzilla.gnome.org/show_bug.cgi?id=777921
+
+2017-01-25 19:13:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/overlay/Makefile.am:
+ qt: The videooverlay example requires at least C++11
+ ... and clang requires this to be specified on the commandline while gcc
+ nowadays defaults to C++11 or even newer.
+
+2017-01-24 19:20:53 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ meson: Properly use ':' for defining keywords
+
+2017-01-23 19:45:05 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: fix LATENCY query
+ The latency query originally had a fallthrough to the default
+ label at the end as fallback, but that got messed up when the
+ DURATION and POSITION queries were added, so it then fell through
+ to the duration query handler instead. Restore original behaviour.
+ https://bugzilla.gnome.org/show_bug.cgi?id=699077
+
+2017-01-23 19:08:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ * tests/check/elements/videorate.c:
+ videorate: fix duration and position query handling
+ Duration query would return TRUE and duration=-1. This
+ worked in the unit test because the unit test implementation
+ was a bit broken.
+ Both queries need to access rate with a lock.
+ Fix broken duration query test as well. It relied on broken
+ behaviour by the videorate query handler, and also it was
+ implemented as a downstream query rather than an upstream
+ query. And we must return HANDLED from the probe so that the
+ query we intercept actually returns TRUE.
+ https://bugzilla.gnome.org/show_bug.cgi?id=699077
+
+2017-01-23 19:50:09 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.h:
+ audio: add since markers to docs for new enums
+ https://bugzilla.gnome.org/show_bug.cgi?id=777655
+
+2016-11-17 13:04:18 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ * gst-libs/gst/audio/gstaudioringbuffer.h:
+ audio: add FLAC to GstAudioRingBufferFormatType
+ https://bugzilla.gnome.org/show_bug.cgi?id=777655
+
+2017-01-23 18:31:54 +0000 Olivier Crete <olivier.crete@collabora.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ * gst-libs/gst/audio/gstaudioringbuffer.h:
+ audioringbuffer: Also support raw AAC
+ Support raw AAC streams without the ADTS header
+ https://bugzilla.gnome.org/show_bug.cgi?id=777655
+
+2017-01-20 23:28:23 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: update libs section
+ Include documented symbols that were not declared in section file.
+
+2017-01-20 12:41:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: Don't divide block align by zero channels
+ https://bugzilla.gnome.org/show_bug.cgi?id=777525
+
+2017-01-20 08:02:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/subparse/samiparse.c:
+ samiparse: Check that the string has a non-zero length before overwriting the last byte with '\0'
+ https://bugzilla.gnome.org/show_bug.cgi?id=777502
+
+2017-01-15 18:42:34 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: Don't recurse in for nested WAVEFORMATEX
+ There was already a check for that, but it failed because
+ subformat_guid[0] is a guint32 and that is then casted implicitely to a
+ guint16 when recursing... just that we checked the uncasted value.
+ This caused an infinite recursion and thus stack overflow.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777265
+
+2017-01-18 14:59:18 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Don't leak blocked pad references on errors
+ When the decodebin state change fails because of an error
+ message, we might not go through PAUSED->READY. Don't leak
+ a ref to decodebin pads due to pad blocking in that case.
+ This is because we return ASYNC going to PAUSED, and if
+ we fail before reaching PAUSED the only transition we'll
+ see is READY->NULL.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775893
+
+2014-11-27 18:02:49 -0600 Carl Karsten <carl@personnelware.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: Add options to make ball pattern based on system time, and invert each second.
+ This adds some extra options that affect pattern=ball mode, allowing the
+ animation to be synced to running time or wall-time clock for comparing
+ sync across different instances / pipelines / machines.
+ Also added is the ability to invert the rendering colours every second,
+ and some different ball motion patterns.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740557
+
+2017-01-15 18:31:56 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: Check for valid channels/rate before using the values
+ Otherwise we might divide by zero or otherwise create invalid caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777262
+
+2017-01-13 12:38:52 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: bump version
+
+2017-01-12 16:32:42 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.11.1 ===
+
+2017-01-12 15:30:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * 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:
+ Release 1.11.1
+
+2017-01-12 14:37:17 +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
+
+2017-01-12 14:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/da.po:
+ * po/de.po:
+ * po/fr.po:
+ * po/hr.po:
+ * po/id.po:
+ * po/nb.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2017-01-12 22:28:50 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Preserve seqnum on EOS events
+ When converting EOS to/from our custom fake EOS event,
+ preserve any seqnum on the original event.
+
+2017-01-12 10:51:34 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Avoid races when setting up typefind
+ The state of urisourcebin (and all elements contained within) can
+ change at any point in time, including when setting up the typefind
+ element.
+ In order to avoid ending up with typefind starting without being fully
+ connected, lock the state and connect to the 'have-type' signal.
+
+2017-01-11 18:24:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Fix crashes in fast-paths when converting interlaced formats with different vertical subsampling
+ E.g. the following pipelines fail because chroma values after the last
+ line are read (note: 486 % 4 == 2):
+ gst-launch-1.0 videotestsrc ! "video/x-raw,interlace-mode=interleaved,width=720,height=486,format=UYVY" ! videoconvert ! "video/x-raw,format=I420" ! fakesink
+ gst-launch-1.0 videotestsrc ! "video/x-raw,interlace-mode=interleaved,width=720,height=486,format=I420" ! videoconvert ! "video/x-raw,format=UYVY" ! fakesink
+ gst-launch-1.0 videotestsrc ! "video/x-raw,interlace-mode=interleaved,width=720,height=486,format=I420" ! videoconvert ! "video/x-raw,format=AYUV" ! fakesink
+
+2017-01-11 22:48:02 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Drop fake EOS if the pad got relinked
+ If our pad got relinked in since the fake-EOS was sent to the
+ pad, then drop the fake-EOS event.
+ CID: 1398546
+
+2017-01-11 17:26:01 +0530 Arun Raghavan <arun@arunraghavan.net>
+
+ * common:
+ common: Revert unintentional change in previous commit
+
+2017-01-11 17:14:46 +0530 Arun Raghavan <arun@arunraghavan.net>
+
+ * common:
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Drop some dead code
+ The ret == GST_STATE_CHANGE_FAILURE is handled in the previous for loop
+ already.
+ CID: 1398544
+
+2017-01-11 12:35:40 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Clear pad from pending list if it was linked
+ If not, the other slots might try to link the pad again.
+ This can happen when the demuxer has multiple src pads
+ and their caps are identical
+ https://bugzilla.gnome.org/show_bug.cgi?id=777121
+
+2017-01-11 08:22:21 +0100 Edward Hervey <edward@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ win32: update def file
+
+2017-01-10 16:36:08 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * tests/check/libs/videotimecode.c:
+ videotimecode: Added unit test for GstVideoTimeCodeInterval
+ https://bugzilla.gnome.org/show_bug.cgi?id=776447
+
+2016-12-29 14:42:52 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst-libs/gst/video/gstvideotimecode.c:
+ * gst-libs/gst/video/gstvideotimecode.h:
+ videotimecode: New GstVideoTimeCodeInterval type, ability to add to a GstVideoTimeCode
+ Sometimes there is a human-oriented timecode that represents an
+ interval between two other timecodes. It corresponds to the human
+ perception of "add X hours" or "add X seconds" to a specific timecode,
+ taking drop-frame oddities into account. This interval-representing
+ timecode is now a GstVideoTimeCodeInterval. Also added function to add it to
+ a GstVideoTimeCode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776447
+
+2017-01-10 21:52:34 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Configure typefind element for non-streaming uri
+ To ensure configuring adaptivedemux if needed,
+ setup typefind element even if uri is not matched to streaming protocol.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776458
+
+2016-12-24 16:44:26 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Use GList for typefind elements
+ We need typefind elements per source element's srcpad
+ https://bugzilla.gnome.org/show_bug.cgi?id=776458
+
+2016-12-24 16:15:45 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Remove unused signal handler variable
+ Remove never used handler id
+ https://bugzilla.gnome.org/show_bug.cgi?id=776458
+
+2017-01-10 08:57:51 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ pbutils: Fix annotation in gst_encoding_profile_set_preset
+
+2017-01-09 19:45:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ win32: update .def file for new video API
+
+2017-01-09 19:10:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: fix another c99-ism
+ Missed one.
+
+2017-01-09 19:02:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/ximage/ximagesink.c:
+ * tests/check/libs/audiodecoder.c:
+ * tests/check/libs/sdp.c:
+ * tests/check/libs/videodecoder.c:
+ Fix indentation
+
+2017-01-09 18:58:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: fix compiler warnings due to c99-ism
+ audiodecoder.c:160:5: error: ‘for’ loop initial declarations are only allowed in C99 mode
+
+2016-12-30 20:27:48 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * tests/check/libs/videotimecode.c:
+ videotimecode: Add GstValue functions unit test
+ https://bugzilla.gnome.org/show_bug.cgi?id=772764
+
+2016-12-30 19:08:16 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst-libs/gst/video/gstvideotimecode.c:
+ * gst-libs/gst/video/gstvideotimecode.h:
+ videotimecode: Add GstValue functions
+ Add compare, serialization and deserialization functions
+ https://bugzilla.gnome.org/show_bug.cgi?id=772764
+
+2017-01-08 21:53:27 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Clear EOS state with stream-start/flush-stop event
+ The EOS state marker should cleared on stream-start or flush-stop
+ https://bugzilla.gnome.org/show_bug.cgi?id=777009
+
+2017-01-08 21:36:04 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Never push actual EOS event to slot
+ Due to the special nature of adaptivedemux, reconfigure happens
+ frequently with seek/track-change.
+ In very exceptional cases, the following sequence is possible:
+ * EOS event is pushed to queue element and still buffers are queued
+ * During draining remaining buffers, reconfiguration downstream
+ happens due to track switch.
+ * The queue gets a not-linked flow return from downstream
+ * Because the sinkpad is EOS, the queue registers an
+ error on the bus, causing the pipeline to fail.
+ Avoid the sinkpad getting marked EOS in the first place, by using a
+ custom event in place of EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777009
+
+2017-01-09 21:31:37 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/libs/video.c:
+ testsuite: Add some test checks for gst_video_guess_framerate()
+
+2017-01-09 21:25:26 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/video.c:
+ gst_video_guess_framerate: Don't throw away all precision
+ When operating on framerates near 10000fps, at least keep 1
+ digit of precision for calculations
+
+2017-01-06 12:56:00 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * win32/common/libgstpbutils.def:
+ Update win32 def files
+
+2017-01-06 11:39:27 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-target: Properly free temporary list
+
+2017-01-04 14:27:40 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-profile.h:
+ encoding-profile: Add a way to copy an encoding profile
+ It is often usefull to make sure that you get a full copy of a profile.
+ For example you want to let the user modify it in the user interface
+ but still keep an unchanged version for later use.
+ API:
+ gst_encoding_profile_copy
+
+2017-01-04 14:56:36 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * pkgconfig/Makefile.am:
+ * 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:
+ * pkgconfig/meson.build:
+ meson: generate pkg-config -uninstalled pc files
+ Generating those files is useful for users building the GStreamer stack
+ using meson and having to link it to another project which is still
+ using the autotools.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776810
+
+2017-01-04 11:21:51 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: Fix stream_group_free when creating it went bad
+ Avoiding trying to use NULL pointers
+
+2016-12-30 17:55:18 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/playback/gstplaysink.c:
+ playsink: do not link to audio or video filter using padname
+ ... as a sinkpad need not be called "sink", and it is not the case
+ for e.g. timeoverlay (and friends).
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=776623
+
+2017-01-04 13:44:53 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/tcp/gstmultihandlesink.c:
+ multihandlesink: fix some property descriptions
+
+2017-01-03 02:23:43 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstparsebin.c:
+ playback: Fix a small race on decodebin/parsebin shutdown.
+ When shutting down decodebin2 and parsebin, they set their
+ output pads to flushing, and there is a very small window
+ where elements might send a sticky event such as a tag event
+ (which silently fails due to flushing) and then sends a buffer,
+ and the buffer will return GST_FLOW_ERROR because it can't
+ forward sticky events. The element will then send an error
+ message on the bus. This can also happen when elements send EOS
+ just as shutdown is happening. Since we're about to destroy all
+ the elements inside parsebin and decodebin anyway, just discard
+ error messages from them.
+ A nicer but more difficult fix for GStreamer 2.0 is to make
+ all event pushing / handling in core return a GstFlowReturn
+ like buffers do, so we can report a FLUSHING state cleanly.
+
+2017-01-02 12:54:32 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: fix queue property types when setting
+
+2015-03-13 18:04:31 +0800 Song Bing <b06498@freescale.com>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: allow more buffers in output queue for better performance
+ https://bugzilla.gnome.org/show_bug.cgi?id=744191
+
+2017-01-02 17:56:36 +0530 Arun Raghavan <arun@arunraghavan.net>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: Relocate a NULL check before accessing converter
+ CID 1396745
+
+2015-07-02 07:23:23 +0200 Tobias Mueller <muelli@cryptobitch.de>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: fix compiler warning
+ Initialize min and max _get_property() to gets rid of these
+ compiler warnings:
+ gstappsrc.c:741:7: error: 'max' may be used uninitialized in this function
+ g_value_set_int64 (value, max);
+ ^
+ gstappsrc.c:733:7: error: 'min' may be used uninitialized in this function
+ g_value_set_int64 (value, min);
+ ^
+ Which happens because gcc doesn't know that GST_IS_APP_SRC will never
+ fail here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752052
+
+2015-11-25 11:30:42 +0000 Stuart Weaver <stuart.weaver@datapath.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtsp-url: unescape special chars in user/pass part of URL
+ This way special characters such as '@' can be used in
+ usernames or passwords, e.g.
+ rtsp://view:%40dm%4An@<IP-ADDR>/media/camera1
+ will now parse username and password into:
+ User: view
+ Pass: @dm:n
+ https://bugzilla.gnome.org/show_bug.cgi?id=758389
+
+2015-11-18 13:59:30 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Add support to dump dot files
+ Dump graphs during error/warning messages and discover is done
+ https://bugzilla.gnome.org/show_bug.cgi?id=758259
+
+2016-12-24 10:15:24 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/tag.c:
+ tests: tag: add unit test for ID3v2 UTF-16 string list parsing
+ https://bugzilla.gnome.org/show_bug.cgi?id=770355
+
+2016-12-24 14:32:34 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/tag.c:
+ tests: tag: add test for ID3v2 extended header parsing
+ https://bugzilla.gnome.org/show_bug.cgi?id=770355
+
+2016-08-24 11:39:39 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst-libs/gst/tag/id3v2frames.c:
+ id3v2: fix splitting strings in ISO-8859-1 and UTF-16 formats
+ When parsing NUL-terminated strings, do not include the terminating
+ NUL byte(s). Depending on the encoding used, either g_utf8_validate()
+ failed due to this, or worse the call to g_utf16_to_utf8() would
+ return 0 items read on an empty string, causing it to fail parsing
+ certain frames.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770355
+
+2016-08-24 10:33:14 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst-libs/gst/tag/id3v2.c:
+ id3v2: fix handling of tags with extended headers
+ The extended header size value does not include itself.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770355
+
+2016-12-23 18:08:43 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Initialize variables to avoid build failures
+ encoding-profile.c: In function ‘get_profile_format_from_possible_factory_name’:
+ encoding-profile.c:1532:6: error: ‘fact’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ if (fact)
+ ^
+ encoding-profile.c: In function ‘profile_from_string’:
+ encoding-profile.c:1720:6: error: ‘res’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ if (profile)
+ ^
+ cc1: all warnings being treated as errors
+
+2016-12-23 14:23:48 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Allow using factory names in serialization format
+ Instead of enforcing the user to know and understand caps to describe
+ the encoding format, let him use element factory names directly.
+ This also makes it possible to ensure that a specific encodore/muxer
+ is used instead of letting the ranking system do it.
+ It is now possible to describe an encoding format simply specifying:
+ matroskamux:x264enc:vobisenc
+ Factor out functions in the parsing, cleaning up the whole thing.
+ Update documentation.
+
+2016-12-21 19:32:41 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Also take into account preset name when comparing profiles
+
+2016-12-21 13:24:37 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-profile: Handle path to serialized target when deserializing a profile
+ The synthax is path/to/encoding/profile.gep:profilename
+
+2016-12-21 12:13:38 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-target.h:
+ encoding-target: Add 'file-extension' as a known category
+
+2016-12-21 11:05:30 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-target: Allow using name and targets from serialized file
+ We used to only care about the name of the files even if the name
+ is defined in the encoding target serialized file.
+ That commit also allows user to define several names for a single
+ target file (using a ';' between the names) which allows us to have
+ a target for youtube that is called 'youtube;yt' or a target for
+ 'ogg;ogv;oga' file extension.
+
+2016-12-21 11:01:27 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-target: Auto convert loading target name to lowercase
+ We *only* support lowercase encoding target names so we can just
+ handle user to use uper case ones converting them.
+
+2016-12-21 10:02:31 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ pbutils: Add documentation about encoding targets
+
+2016-12-10 11:43:47 +0900 hoonhee.lee <hoonhee.lee@lge.com>
+
+ * tests/examples/decodebin_next/playbin-test.c:
+ playbin-test: Don't use removed playbin3 'auto-select-streams' property
+ https://bugzilla.gnome.org/show_bug.cgi?id=775917
+
+2016-09-02 15:23:18 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: Fix incorrect start of tick waveform
+ Make sure ticks start with an accumulator value of 0 by incrementing it
+ after filling in samples instead of before and by resetting the accumulator
+ every time a tick begins. This prevents it from being discontinuous at the
+ beginning of the tick.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774050
+
+2016-12-22 18:47:19 +0100 Nicolas Dechesne <nicolas.dechesne@linaro.org>
+
+ * tools/gst-play.c:
+ tools: gst-play: set GST_GL_XINITHREADS
+ This ensure that XInitThreads is called and so gl contexts are properly
+ initialized.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776403
+
+2014-06-26 18:01:06 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ * gst-libs/gst/audio/gstaudioringbuffer.h:
+ * win32/common/libgstaudio.def:
+ audioringbuffer: add set_callback_full() for g-i
+ https://bugzilla.gnome.org/show_bug.cgi?id=678301
+
+2016-12-20 12:33:12 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpclientsrc.h:
+ tcpclientsrc: add timeout property
+ https://bugzilla.gnome.org/show_bug.cgi?id=749567
+
+2016-12-21 00:11:06 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstparsebin.c:
+ parsebin: Ignore failure to send sticky events
+ When plugging and then exposing a parser, don't fail
+ if it fails to send sticky events. The most likely
+ reason is that things were flushed due to the app
+ immediately doing a seek, but we can't detect flushing
+ separately to other error conditions without a
+ gst_pad_send_event_full() core function that returns
+ a GstFlowReturn.
+
+2016-12-20 13:00:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: Fix up last commit
+
+2015-03-28 18:16:16 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add ADPCM_G722 support
+ https://bugzilla.gnome.org/show_bug.cgi?id=746574
+
+2016-12-19 15:20:35 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/elements/encodebin.c:
+ tests: Fix build
+
+2016-12-19 15:08:12 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: Fix build initializing sprof
+
+2016-12-16 22:11:41 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/encoding/gstencodebin.c:
+ * tests/check/elements/encodebin.c:
+ encodebin: Fallback to other profile if we fail with one
+ In some case we might have EncodingProfile that will be defined
+ in a way that, for example if a Preset is not present, another
+ profile for that stream should be used.
+ A test is added showing the feature.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776188
+
+2016-12-16 16:27:04 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: Enhance error debug when failing to create an encoder
+
+2016-12-18 12:29:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/tag/id3v2.c:
+ tag: id3v2: turn redundant check into an assert
+ We checked this already earlier, so this is dead code.
+ Leave an assert in place for consistency with the other
+ branch and in case the rest of the code changes.
+ CID 1397350.
+
+2016-12-17 21:58:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Get caps from the element's srcpad if possible
+ The caps put into the stream topology by decodebin are the caps at the
+ moment the pads are exposed on it. This is usually before decoders
+ received any buffers.
+ In discoverer we however wait for pre-roll, which ensures that each
+ decoder handled buffers already. At this point, there might be more
+ information known about the caps already that we could make use of.
+ One example here is extra information stored in the SEI of H264, like
+ the multiview-mode. This will be known if there is a SEI before the
+ first keyframe, but decodebin won't put this into the topology as it
+ only waits for the initial caps of h264parse (which come directly after
+ SPS/PPS).
+ With this change, the multiview-mode is in the caps reported by
+ discoverer in many cases.
+
+2016-12-17 21:35:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Put the correct element srcpad into the topology for the very last element of a chain
+ We were putting the decode pad there, which is the ghostpad linked to
+ the last element. The decode pad is already in the pad field.
+
+2016-12-17 21:34:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Put the correct pad into the stream-topology if a parser/converter is used
+ We have to take the capsfilter into account then as the elements are not
+ linked directly. Previously this caused NULL be set in these cases.
+
+2016-12-16 17:39:59 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-plugins-base.spec.in:
+ Actually delete .spec.in file as well
+ Only removed it from build description.
+
+2016-12-16 11:11:57 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst/encoding/gstencodebin.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ encoding-profile: Fix documentation and port to gtk markdown
+ And remove some trailling whitepsaces
+
+2016-12-16 09:59:25 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * docs/libs/meson.build:
+ * docs/meson.build:
+ * meson.build:
+ * meson_options.txt:
+ meson:doc: Build libraries documentations
+
+2016-12-16 09:58:15 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ base: Actually support using the default encoding target
+
+2016-12-15 16:12:02 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-target: Remove useless check for local presence
+
+2016-12-15 16:10:55 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ pbutils: Add safe guard too encoding profile API
+
+2016-12-15 10:57:14 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/audio/audio-channels.c:
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ audio: Fix introspection annotation
+ In gst_audio_check_valid_channel_positions the mask
+ is an out parameter.
+ And minor conversion from a print to a GST_ERROR.
+
+2016-12-14 18:06:09 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-target: Handle GST_ENCODING_TARGET_PATH in list_all
+ And fix the compare_target function
+
+2016-12-15 16:29:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: For adaptive streaming, ensure to put the buffering multiqueue after a parser or demuxer
+ There are cases when there is no demuxer involved that could do the
+ buffering, e.g. HLS with raw MP3 or AAC. In this case we want to place
+ the buffering multiqueue after the parser.
+ Before this change, we've considered the first element after the
+ adaptive streaming demuxer as a parser. This is not always true, e.g.
+ id3demux. Instead we now wait until we actually have a parser (or
+ decoder).
+ Fixes playback on such HLS streams.
+
+2016-12-14 09:48:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ Remove generated .spec file
+ Likely extremely bitrotten, and we should not ship this anyway.
+
+2016-12-13 22:45:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ * tests/check/libs/rtsp.c:
+ * win32/common/libgstrtsp.def:
+ rtsp: add boxed types for new authentication credential API
+ To make the structs usable in bindings, and fix
+ gstrtspmessage.c:1188: Warning: GstRtsp:
+ gst_rtsp_message_parse_auth_credentials: return value: Invalid
+ non-constant return of bare structure or union; register as
+ boxed type or (skip)
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-12-13 22:26:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * tests/check/elements/videotestsrc.c:
+ gst: Don't declare variables inside the for loop header
+ This is a C99 feature.
+
+2016-12-13 09:44:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: do not require 4 byte multiple for encoded MPEG
+ Bytes per frame doesn't make sense for encoded audio.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776038
+
+2016-12-12 14:50:11 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstrawcaps.h:
+ playback: Add ANY caps features to default text raw caps
+ Raw text caps with any caps features should be also default raw caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=775967
+
+2016-12-09 17:08:20 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * meson.build:
+ meson: Support building without Gst debug
+
+2016-12-09 17:36:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ xmptag: Don't leak the namespace string if there are multiple
+ https://bugzilla.gnome.org/show_bug.cgi?id=775887
+
+2016-12-09 17:59:09 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/tag/id3v2.c:
+ id3v2: Clarify id3v2_add_id3v2_frame_blob_to_taglist()
+ Pass the frame data and size explicitly to
+ id3v2_add_id3v2_frame_blob_to_taglist() and add a
+ comment that it's being deliberately / manually
+ passed the full ID3v2 frame including header.
+
+2016-12-09 17:57:52 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/tag/id3v2.c:
+ id3v2: Add missing overrun check for frame sizes
+ When frames claim to have a footer, ensure they
+ are large enough to contain one to avoid an invalid
+ read overrun.
+ Spotted by Joshua Yabut
+
+2016-11-22 23:08:09 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/ogg/gstogmparse.c:
+ ogg: Fix element factory klass for OGM parsers
+ They're parsers, not decoders, so fix the klass info
+ accordingly.
+
+2016-12-08 23:01:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * configure.ac:
+ * docs/Makefile.am:
+ * docs/design/Makefile.am:
+ * docs/design/draft-hw-acceleration.txt:
+ * docs/design/draft-va.txt:
+ docs: design: remove outdated draft docs (hw-acceleration, va)
+
+2016-12-08 22:59:58 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/design/Makefile.am:
+ * docs/design/design-audiosinks.txt:
+ * docs/design/design-decodebin.txt:
+ * docs/design/design-encoding.txt:
+ * docs/design/design-orc-integration.txt:
+ * docs/design/draft-keyframe-force.txt:
+ * docs/design/draft-subtitle-overlays.txt:
+ * docs/design/part-interlaced-video.txt:
+ * docs/design/part-mediatype-audio-raw.txt:
+ * docs/design/part-mediatype-text-raw.txt:
+ * docs/design/part-mediatype-video-raw.txt:
+ * docs/design/part-playbin.txt:
+ * docs/design/part-stereo-multiview-video.markdown:
+ docs: design: move most design docs to gst-docs module
+
+2016-12-03 23:01:53 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3-parse.c:
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Remove unused variable
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-12-03 22:46:20 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3-parse.c:
+ decodebin3: More cleanup DecodebinOutputStream and MultiQueueSlot
+ When removing DecodebinInputStream, cleanup DecodebinOutputStream and
+ MultiQueueSlot also if they were drained.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-12-03 22:37:55 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Drop duration query during _input_pad_unlink ()
+ Playbin3 takes lock when querying duration and handling
+ stream-collection message. So,to post stream-collection message,
+ duration query should be dropped when input pad is being unlinked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-12-03 22:12:21 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Update stream-collection with _input_pad_unlink()
+ Since parsebin does not post new stream-collection message when
+ it was being removed, decodebin3 should update it itself.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-12-03 22:28:28 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Cleanup no more used DecodebinInput
+ Remove DecodebinInput using gst_element_call_async() API.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-12-03 21:50:47 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Cleanup no more used MultiQueueSlot
+ Since MultiQueueSlot cannot be removed inside of streaming thread,
+ use gst_element_call_async() API.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-12-03 21:42:30 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3-parse.c:
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Send custom-eos event to notify drained state
+ Likewise how urisourcebin is doing, use custom event if other streams
+ are still alive.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-12-03 20:44:21 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstplaybin3.c:
+ playbin3: Reconfigure playsink again with pad-removed
+ If selected streams and actived streams are matched,
+ do reconfigure of playsink again with pad-removed signal
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-10-25 21:06:40 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstplaybin3.c:
+ playback: Remove trailing whitespace
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-10-23 22:10:39 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Try to link output slot before cleanup
+ Before cleaning up output slot, check pending pads first, if available.
+ Then, cleanup it only if linking was failed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-10-22 18:53:17 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Cleanup unused output slot
+ Since urisourcebin cannot cleanup unused output slot
+ in streaming thread, it will be handled in thread pool
+ with gst_element_call_async ().
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-12-06 16:29:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: Fix crash when shutting down element during getrange()
+ Ensure that nothing is in any of the streaming thread functions
+ anymore when going from PAUSED to READY. While the parent's state change
+ function has deactivated all pads, there is nothing preventing
+ downstream from activating our srcpad again and calling the getrange()
+ function. Although we're in READY!
+ https://bugzilla.gnome.org/show_bug.cgi?id=775687
+
+2016-12-03 08:19:15 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From f980fd9 to 39ac2f5
+
+2016-12-02 15:12:12 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add another test to itc typefinder
+ Report certainty after every test passes.
+ Additionally:
+ - Remove self-explanatory comment.
+
+2016-12-01 19:57:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/subparse/gstssaparse.c:
+ ssaparse: Free initialization section before storing the next one
+ If getting multiple caps events.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775480
+
+2016-12-01 15:12:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't end up ignoring caps just because there are no headers for this stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=775459
+
+2016-11-30 10:55:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: Error out if mapping input/output buffer failed
+
+2016-11-30 10:48:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: Don't map the input buffer in in-place mode
+ Input and output buffer are the same, let's not do unnecessary work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775369
+
+2016-11-30 10:43:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ audio-converter: In passthrough, also don't copy if in and out block are the same
+ In and out array are usually different, they are stack allocated arrays.
+ However the blocks inside them still can be the same.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775369
+
+2016-11-30 10:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: Don't call transform_ip() in passthrough mode
+ https://bugzilla.gnome.org/show_bug.cgi?id=775369
+
+2016-11-29 15:30:43 +0100 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst/tcp/gstmultihandlesink.c:
+ multihandlesink: Fix buffers-queued being off by one
+ max_buffer_usage is the index of the oldest buffer in the queue,
+ starting at zero, not the number of buffers queued.
+ find_limits returns the index of the oldest buffer that satisfies the
+ limits in its min_idx parameter, not the number of buffers needed. Fix
+ this use too in order to keep passing the tests that read
+ buffers-queued.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775351
+
+2016-11-29 16:26:22 +0100 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * tests/check/elements/multifdsink.c:
+ multifdsink: Add a test involving a slow client
+ https://bugzilla.gnome.org/show_bug.cgi?id=774908
+
+2016-11-23 14:35:04 +0100 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst/tcp/gstmultihandlesink.c:
+ multihandlesink: Update bufpos in a separate pass
+ If a client gets dropped and the iteration gets restarted, bufpos is
+ incremented again for all clients that preceded the dropped one, causing
+ havoc.
+ Adjust the bufpos for all clients first before trying to drop any.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774908
+
+2016-11-29 16:37:50 +0530 Garima Gaur <garima.g@samsung.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Fix caps memory leak in usage of gst_static_caps_get() API
+ https://bugzilla.gnome.org/show_bug.cgi?id=775310
+
+2016-11-28 20:25:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstaudio.def:
+ win32: update .def file for new audioconverter API
+ Fixes distcheck.
+
+2016-11-28 18:28:24 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: Add headers and libm to has_function checks
+ The functions from math.h may be implemented in libm.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774876
+
+2016-11-28 19:45:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Handle NULL/ANY/EMPTY caps without crashing
+
+2016-11-28 16:54:55 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/elements/videorate.c:
+ check/videorate: Avoid leaking extra buffers
+
+2016-11-28 16:53:10 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/video-info.c:
+ video-info: Properly initialize/set extra fields
+ The flags and field order weren't properly initialized in the
+ gst_video_info_init().
+ Furthermore in gst_video_info_from_caps() we might set unitiliazed
+ values previously, this only sets them if valid.
+
+2016-11-28 16:51:43 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Fix attribute leak
+ We need to free the strdup'd string (to_free) in all cases
+
+2016-11-28 16:51:23 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: Don't leak authorization string
+
+2016-10-19 12:21:37 +0200 Petr Kulhavy <brain@jikos.cz>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-converter.h:
+ * gst/audioconvert/gstaudioconvert.c:
+ audio-converter: optimize endian conversion
+ Optimize LE<->BE conversion by adding a dedicated fast path instead of
+ using the generic converter. Implement transform_ip function in order to do the
+ endian swap in place.
+ This saves buffer allocation for the intermediate format, can be done in place
+ and also performs the conversion in one step instead of unpack-convert-pack.
+ For all bit widths the naive algorithm is implemented, which provides the best
+ performance when compiled with -O3. ORC was considered but eventually removed
+ as it requires a dedicated function for in-place conversion (due to the
+ "restrict" parameters).
+ A more complex algorithm for the 24-bit conversion with unrolled loop and
+ 32-bit processing is implemented in the #if 0 section. It performs better if
+ compiled with -O2. With -O3 however the naive algorithm performs better.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773073
+
+2016-10-21 14:30:31 +0200 Petr Kulhavy <brain@jikos.cz>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ audio-convert: simplify the chain free process
+ It is not needed to store a pointer to every single chain element to free it.
+ Instead walk the channel list backwards and free the chain elements one by one.
+ Rename GstAudioConverter->chain_pack to chain_end.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773073
+
+2016-11-28 17:12:26 +0530 Garima Gaur <garima.g@samsung.com>
+
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Fix caps memory leak when failing to get sinkpad from subtitle renderer
+ https://bugzilla.gnome.org/show_bug.cgi?id=775224
+
+2016-11-28 10:12:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Extract video information from caps manually without GstVideoInfo
+ The caps might not be fixated (which is required by GstVideoInfo) and we
+ would assert otherwise. However the caps often contain useful
+ information in the already-fixed parts that we can use here.
+
+2016-11-28 10:04:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Also stop waiting for subtitles if we get EOS
+ We're not going to get a buffer or GAP event anymore after EOS and would
+ wait forever otherwise.
+
+2016-11-26 13:53:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: Check if caps are NULL before using them for the first time, not afterwards
+ Otherwise we'll get a g_critical() before erroring out cleanly on
+ https://samples.mplayerhq.hu/A-codecs/ATRAC3/SND0.AT3
+
+2016-11-26 11:20:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitmodules:
+ common: use https protocol for common submodule
+ https://bugzilla.gnome.org/show_bug.cgi?id=775110
+
+2016-11-25 10:48:06 +0100 Miguel Paris <mparisparis@gmail.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: Fix ensure_buffers() if whole packet is in one GstMemory
+ When gst_rtp_buffer_add_extension_onebyte_header() is used over a
+ GstRtpBuffer that only contains a memory for the whole packet,
+ ensure_buffers function crashes at the next point:
+ mem = gst_memory_copy (rtp->map[i].memory, offset, rtp->size[i]);
+ when i==2 because the payload is not mapped.
+ In addition the offset is calculated subtracting in the wrong direction.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774959
+
+2016-11-24 15:40:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-info.c:
+ * tests/check/libs/video.c:
+ video-info: Add unit test for overflow checks
+ And also prevent overflows caused by allowing uint width/height in
+ gst_video_info_set_format() but storing them as (signed!) ints.
+
+2016-11-24 15:12:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-info.c:
+ video-info: And change the overflow check to not actually overflow itself
+
+2016-11-23 20:10:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/gstvideopool.c:
+ * gst-libs/gst/video/video-blend.c:
+ * gst-libs/gst/video/video-overlay-composition.c:
+ video: Handle errors in gst_video_info_set_format() / gst_video_info_align()
+ https://bugzilla.gnome.org/show_bug.cgi?id=774588
+
+2016-11-23 20:00:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-info.h:
+ video-info: Sanity check the frame size to prevent overflows
+ https://bugzilla.gnome.org/show_bug.cgi?id=774588
+
+2016-11-23 13:48:06 +0100 Ulf Olsson <ulfo@axis.com>
+
+ * gst-libs/gst/sdp/gstmikey.c:
+ mikey: Generate the correct SRTP policy
+ https://bugzilla.gnome.org/show_bug.cgi?id=774911
+
+2016-11-23 18:26:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ rtspdefs: Assert on the excepted MD5 digest string length
+ CID 1394494.
+
+2016-11-23 21:27:55 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: bounds check windows ico detection
+ Fixes out of bounds read
+ https://bugzilla.gnome.org/show_bug.cgi?id=774902
+
+2016-11-22 21:12:23 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst-libs/gst/tag/mklicensestables.c:
+ tag: fix some warnings in mklicensestables
+ https://bugzilla.gnome.org/show_bug.cgi?id=774878
+
+2016-10-07 15:08:37 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ gstfdmemory: log with GST_INFO instead of GST_ERROR on permission denied
+ For example mmap can fail with EACCES if the the fd has been open
+ with read only mode. And mapping the memory might be the only way
+ to check that. So no need to print out an error.
+ Ex: ioctl(dev, DRM_IOCTL_PRIME_HANDLE_TO_FD, flags & ~DRM_RDWR)
+ https://bugzilla.gnome.org/show_bug.cgi?id=765600
+
+2016-10-18 16:18:19 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add typefinder for Apple/iTunes itc artwork files
+ Avoids audio/mpeg false-positive described at:
+ https://bugzilla.gnome.org/show_bug.cgi?id=773172
+
+2016-11-18 16:51:26 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ * tests/check/libs/rtsp.c:
+ * win32/common/libgstrtsp.def:
+ rtsp: Add gst_rtsp_message_parse_auth_credentials() to parse authentication credentials
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-18 13:20:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * win32/common/libgstrtsp.def:
+ rtsp: Add gst_rtsp_generate_digest_auth_response() to calculate digest auth response
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-20 15:43:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * win32/MANIFEST:
+ * win32/common/_stdint.h:
+ * win32/common/audio-enumtypes.c:
+ * win32/common/audio-enumtypes.h:
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/gstrtsp-enumtypes.h:
+ * win32/common/multichannel-enumtypes.c:
+ * win32/common/multichannel-enumtypes.h:
+ * win32/common/pbutils-enumtypes.c:
+ * win32/common/pbutils-enumtypes.h:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ win32: remove copies of generated headers
+
+2016-11-18 14:51:29 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/allocators/gstdmabuf.h:
+ dmabuf-allocator: Add missing padding in the class
+ This class was made subclassable, though for future growth of the code,
+ it's better if we have some room for add class members. Using the small
+ padding since this is unlikely.
+
+2016-11-17 20:18:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ textoverlay: Mark pad as needing reconfiguration again if it failed
+ And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:46:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Ensure to set the RECONFIGURE flag again if reconfiguration failed
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 16:45:32 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: add_global_arguments -> add_project_arguments
+ https://bugzilla.gnome.org/show_bug.cgi?id=774656
+
+2016-11-17 10:16:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/libs/videotimecode.c:
+ videotimecode: Add test for the calculations of distance from the daily jam
+ https://bugzilla.gnome.org/show_bug.cgi?id=774585
+
+2016-11-16 19:13:14 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst-libs/gst/video/gstvideotimecode.c:
+ videotimecode: Fix incorrect nsec_since_daily_jam calculation
+ For drop-frame timecodes, the nsec_since_daily_jam doesn't necessarily
+ directly correspond to this many hours/minutes/seconds/frames. We have
+ to get the frame count as per frames_since_daily_jam and then convert.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774585
+
+2016-11-16 20:48:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Extract bpp from the strf for vnmc
+ Needed for avdec_vnmc to work.
+
+2016-11-17 00:40:43 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: warn if a custom sink is set that has no 'sink' pad
+
+2016-11-15 09:32:24 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder, audiodecoder: parse format before checking in src_query_default
+ The logic change in these commits misordered the parsing and checking of
+ format in position queries:
+ 2b06e54 videodecoder: Don't answer BYTES queries
+ 1840b02 audio: Don't answer BYTES queries
+ https://bugzilla.gnome.org/show_bug.cgi?id=774484
+
+2016-11-15 18:32:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ appsink: fix g-i warnings and add since markers
+ Rename function parameter and make sure the name in the
+ declaration matches the name in the implementation, to
+ avoid g-i warnings. Also add Since markers for gtk-doc.
+ gstappsink.c:1248: Warning: GstApp: gst_app_sink_set_buffer_list_support:
+ unknown parameter 'buffer_list' in documentation comment, should be 'drop'
+
+2016-11-15 15:12:12 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Do not try to unref the bus if it has not been set yet
+ It might happen if creation of the discoverer failed
+
+2016-07-04 09:32:28 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ * tests/check/elements/appsink.c:
+ * win32/common/libgstapp.def:
+ appsink: add support for buffer lists
+ https://bugzilla.gnome.org/show_bug.cgi?id=752363
+
+2016-11-15 15:23:20 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/playback/gstplaybin3.c:
+ playbin3: remove dead code
+ It never reach into this code path, custom_combiner always not null
+ here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774454
+
+2016-11-15 23:36:41 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Don't answer BYTES queries
+ Refuse to answer BYTES queries ourselves. The only
+ time they make sense is on raw elementary streams,
+ in which case upstream would already have answered.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757631
+
+2016-11-15 23:27:17 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audio: Don't answer BYTES queries
+ Refuse to answer BYTES queries ourselves. The only
+ time they make sense is on raw elementary streams,
+ in which case upstream would already have answered.
+ They especially don't make sense for encoders to answer
+ based on upstream values - although perhaps later
+ we could make it do TIME->BYTES conversion on the source
+ pad based on bitrate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757631
+
+2016-11-14 16:55:36 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: cast away const in call to g_free
+ MSVC warns about the const here. It's safe to cast away.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-11-14 16:48:16 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst-libs/gst/audio/gstaudiometa.c:
+ * gst-libs/gst/video/gstvideoaffinetransformationmeta.c:
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/video-overlay-composition.c:
+ Cast away const from GstMetaInfo in *_get_meta_info() functions
+ MSVC warns about the const in the implicit argument conversion in the
+ calls to g_once_init_{enter,leave}. It's OK so explicitly cast it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-11-13 13:15:38 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3-parse.c:
+ decodebin3: Clear saw_eos flag of DecodebinInputStream by FLUSH event
+ Likewise how GstPad is doing, saw_eos flag of DecodebinInputStream
+ must be cleared by FLUSH event.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774343
+
+2016-10-17 15:38:37 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/playback/gstplaybin3.c:
+ playbin3: Fix deadlock when adding multiple parsebin
+ https://bugzilla.gnome.org/show_bug.cgi?id=773131
+
+2016-11-14 11:39:33 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * ext/vorbis/meson.build:
+ meson: vorbis: Add -DTREMOR to flags for gstivorbisdec
+ Matching the flags set by Makefile.am
+ https://bugzilla.gnome.org/show_bug.cgi?id=774445
+
+2016-11-14 16:28:42 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/audio/meson.build:
+ * gst-libs/gst/video/meson.build:
+ * gst/adder/meson.build:
+ * gst/videotestsrc/meson.build:
+ * gst/volume/meson.build:
+ meson: Fix build when orc is disabled
+ Making sure not to use the orc_dep variable in case
+ orc has been explicitely disabled.
+
+2016-11-11 10:38:58 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst-libs/gst/video/video-info.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaybin3.c:
+ Use intermediate guint when handling GstVideoMultiviewFlags
+ The underlying integer type of the enum GstVideoMultiviewFlags is
+ implementation defined and may not have the same size as guint.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-11-11 10:35:00 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * ext/ogg/gstoggstream.c:
+ * gst-libs/gst/video/gstvideotimecode.c:
+ Remove 'return' from `void` functions
+ https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-10-26 22:37:19 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: don't add_global_arguments when being built as a subproject
+ https://bugzilla.gnome.org/show_bug.cgi?id=773568
+
+2016-11-10 17:05:19 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/meson.build:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/meson.build:
+ rtsp: Include GstSdp-1.0.gir when generating the gir
+ It is actually needed as we need some symbols. We do not link
+ to libgstsdp as the user of the lib should do it (same with
+ autotools build).
+ This reverts previous commit
+
+2016-11-10 16:36:49 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/rtsp/Makefile.am:
+ libs:rtsp: Remove wrong dependency on Sdp for the gir file
+
+2016-11-10 16:36:49 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/meson.build:
+ libs:rtsp: Remove wrong dependency on Sdp for the gir file
+
+2016-10-20 17:17:27 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/allocators/meson.build:
+ * gst-libs/gst/app/meson.build:
+ * gst-libs/gst/audio/meson.build:
+ * gst-libs/gst/fft/meson.build:
+ * gst-libs/gst/pbutils/meson.build:
+ * gst-libs/gst/riff/meson.build:
+ * gst-libs/gst/rtp/meson.build:
+ * gst-libs/gst/rtsp/meson.build:
+ * gst-libs/gst/sdp/meson.build:
+ * gst-libs/gst/tag/meson.build:
+ * gst-libs/gst/video/meson.build:
+ * meson.build:
+ * meson_options.txt:
+ meson: Generate girs
+ https://bugzilla.gnome.org/show_bug.cgi?id=773944
+
+2016-11-07 12:01:16 +0100 Petr Kulhavy <brain@jikos.cz>
+
+ * gst-libs/gst/audio/audio-channels.c:
+ audio-channels: map buffer read-write only if channels differ
+ gst_audio_buffer_reorder_channels() was always mapping the buffer read-write
+ regardless whether any reordering was needed. If the from and to channel order
+ is identical return immediately without remapping the buffer.
+ Add a small helper function gst_audio_channel_positions_equal() which is used
+ in both gst_audio_reorder_channels() and gst_audio_buffer_reorder_channels().
+ https://bugzilla.gnome.org/show_bug.cgi?id=773833
+
+2013-09-17 17:42:05 +0200 Joris Valette <joris.valette@gmail.com>
+
+ * gst/videorate/gstvideorate.c:
+ * gst/videorate/gstvideorate.h:
+ * tests/check/elements/videorate.c:
+ videorate: Add fixed rate property
+ https://bugzilla.gnome.org/show_bug.cgi?id=699077
+
+2016-11-04 16:41:05 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: fix 120 ms buffers being wrongly emitted
+ Using the max 120 ms buffer size to ensure we have enough space
+ for decoded data meant that Opus could actually return 120 ms'
+ worth of data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771723
+
+2016-11-04 18:55:44 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ win32: Update exports for new API
+
+2016-10-14 15:14:14 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ video: add gst_video_decoder_allocate_output_frame_with_params
+ It adds a third argument to pass GstBufferPoolAcquireParams
+ to gst_buffer_pool_acquire_buffer.
+ If a user subclasses GstBufferPoolAcquireParams, this allows to
+ pass an updated param to the underlying buffer pool at each
+ gst_video_decoder_allocate_output_frame_with_params call.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773165
+
+2016-11-04 16:25:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-info.h:
+ video-info: Fix the docs to say interlace-mode, not interlaced-mode
+
+2016-11-03 21:34:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstallocators.def:
+ win32: add new API to .def file
+ Fixes make check and make distcheck
+
+2015-12-11 17:05:14 +0000 Julien Isorce <j.isorce@samsung.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/allocators/gstdmabuf.h:
+ allocators: define GST_CAPS_FEATURE_MEMORY_DMABUF
+ Adds "memory:DMABuf" caps feature. Since 1.11 tag.
+ Useful when the the dma-buf buffer cannot be mapped to CPU for r/w requests.
+ Example: protected content or platform constraints.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759358
+
+2016-10-24 11:00:07 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ * gst-libs/gst/allocators/gstdmabuf.h:
+ dmabuf: Make the allocator sub-classable
+ This should allos for cleaner code when implement such allocator.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768794
+
+2014-11-27 13:52:52 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosrc: Leave read loop if no longer running
+ In the case a src stops providing data (read calls returns 0). The audio
+ src thread will never leave. Instead, check the condition and leave the
+ loop.
+
+2016-11-03 17:18:05 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/videoscale.c:
+ check: Fix corrupted xml check files
+ By making sure each different videoscale check instance gets logged
+ into different output file
+
+2016-11-02 11:04:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video: Update orc generated files
+
+2016-11-02 11:03:42 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/meson.build:
+ * gst/pbtypes/meson.build:
+ meson: Add pbtypes plugin
+
+2015-05-28 22:50:05 +1000 Jan Schmidt <jan@centricular.com>
+
+ * configure.ac:
+ * gst/Makefile.am:
+ * gst/pbtypes/Makefile.am:
+ * gst/pbtypes/gstpbtypes.c:
+ pbtypes: Add a stub plugin that owns the plugins-base dynamic types
+ https://bugzilla.gnome.org/show_bug.cgi?id=750079
+
+2016-10-07 16:20:24 +0900 Changbok Chea <changbok.chea@gmail.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Fix adaptive demuxer's property checking and buffering setting
+ - Add adaptive demuxer's 'connection-speed' property checking
+ - Set adaptive demuxer q2 buffering property via urisrc use_buffering value
+ https://bugzilla.gnome.org/show_bug.cgi?id=772550
+
+2016-11-01 23:51:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ * gst-libs/gst/video/video-info.c:
+ Revert "video-color: Allow converting incomplete colorimetry to a string"
+ This reverts commit 158eae7e7e3da3545712dd7d6121492c53085fd9.
+ It already *always* allowed to convert incomplete colorimetry to a
+ string.
+
+2016-05-02 09:48:09 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasedepayload: Reject non-TIME segments
+ https://bugzilla.gnome.org/show_bug.cgi?id=765796
+
+2016-11-01 21:09:04 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ Revert "basertpdepayload: create valid segment when given non-time segment"
+ This reverts commit 0f609bc6c67fea294f4556627228fed72a74d0fb.
+
+2016-09-30 15:03:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ * gst-libs/gst/video/video-info.c:
+ video-color: Allow converting incomplete colorimetry to a string
+ This is only a good idea for non-raw caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771376
+
+2016-09-29 14:57:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Proxy colorimetry and chroma-site from input to output caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=771376
+
+2016-09-29 14:48:29 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Proxy field order to the output caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=771376
+
+2016-09-29 14:48:00 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Proxy interlace-mode and field-order fields from the input to the output caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=771376
+
+2016-09-29 14:36:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-info.h:
+ * gst-libs/gst/video/videoorientation.c:
+ * win32/common/libgstvideo.def:
+ video-info: Add optional field-order caps field for interlaced-mode=interleaved
+ Usually this information is static for the whole stream, and various
+ container formats store this information inside the headers for the
+ whole stream.
+ Having it inside the caps for these cases simplifies code and makes it
+ possible to express these requirements more explicitly with the caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771376
+
+2016-11-01 18:08:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+2016-10-22 11:08:18 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Remove trailing whitespace
+ https://bugzilla.gnome.org/show_bug.cgi?id=773341
+
+2016-10-14 15:18:28 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * tests/check/elements/videotestsrc.c:
+ videotestsrc: Make snow deterministic
+ Deterministic generation of snow and smpte is important for tests so
+ that it's not affected by other videotestsrc elements in current or
+ possibly previous tests.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773102
+
+2016-10-14 22:31:41 +0200 Petr Kulhavy <brain@jikos.cz>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: optimize mask calculation
+ find_suitable_mask() had complexity O(n^2) on the number of bits.
+ For common case like 2-channel audio the mask was calculated in about 4k loop
+ cycles.
+ Optimize both n_bits_set() and find_suitable_mask() to O(n) where n is the
+ number of bits set in the mask.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772864
+
+2016-10-13 10:12:10 +0900 hoonhee.lee <hoonhee.lee@lge.com>
+
+ * gst/playback/gstparsebin.c:
+ parsebin: Rename variables include 'decode' to 'parse'
+ https://bugzilla.gnome.org/show_bug.cgi?id=772832
+
+2016-10-31 16:33:41 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Fix GST_TYPE_URI_SOURCE_BIN macro typo
+ https://bugzilla.gnome.org/show_bug.cgi?id=772445
+
+2016-10-03 17:12:29 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: fix to log event pointer
+ https://bugzilla.gnome.org/show_bug.cgi?id=772445
+
+2016-09-28 16:13:46 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/playback/gsturisourcebin.c:
+ urisourcebin: Make use of adaptive demuxer variable
+ https://bugzilla.gnome.org/show_bug.cgi?id=772445
+
+2016-10-06 11:44:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: interpret zero duration as unknown
+ This fixes missing audio when we get buffers with zero
+ duration, denoting unknown duration. When several such
+ buffers are received in a row, they're all at the same
+ timestamp, with zero duration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771723
+
+2016-09-26 10:50:52 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: fix "buffer too small" error
+ Always supply a buffer with max size to the decoder, as we
+ can't really decide how many samples will be in the lost packet
+ based on the timestamps we get.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771723
+
+2016-10-28 08:47:40 +0200 Tomasz Zajac <tomasz.zajac@motorolasolutions.com>
+
+ * tests/check/libs/sdp.c:
+ sdp: Add tests for rtcp-fb parsing
+ https://bugzilla.gnome.org/show_bug.cgi?id=769698
+
+2016-10-28 08:47:01 +0200 Tomasz Zajac <tomasz.zajac@motorolasolutions.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Parse rtcp-fb media attributes
+ https://bugzilla.gnome.org/show_bug.cgi?id=769698
+
+2016-08-10 11:38:58 +0200 Tomasz Zajac <tomasz.zajac@motorolasolutions.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Add rtcp-fb media attributes based on caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=769698
+
+2016-09-07 15:01:13 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-orc.orc:
+ * tests/check/libs/video.c:
+ video: Add VYUY pixel format
+ This format is sometimes the output of JPEG decoders. It is the same as
+ YUY2 and UYVY but with a different component order.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767450
+
+2015-10-15 12:52:27 +0200 Marcin Kolny <marcin.kolny@gmail.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ audioclock: use GstAudioClock* as first argument in GstAudioClock methods
+ All the GstAudioClock method declarations required object of GstClock type
+ as a first argument, but in fact, required GstAudioClock object (runtime
+ check in function body). Instead of checking type in run-time, we can
+ change functions declaration, to accept only GstAudioClock methods. Then,
+ runtime check is not necessary anymore, since always GstAudioClock object
+ is passed to a function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756628
+
+=== release 1.11.0 ===
+
+2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.10.0 ===
+
+2016-11-01 17:53:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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 1.10.0
+
+2016-11-01 17:43:45 +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
+
+2016-10-25 08:52:52 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: Don't depend on gstreamer-check-1.0 on windows
+ https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-24 19:13:22 +0000 Graham Leggett <minfrin@sharp.fm>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: Fix assertion failure when unreffing NULL stream caps
+ GStreamer-CRITICAL **: gst_mini_object_unref: assertion 'mini_object != NULL' failed
+ https://bugzilla.gnome.org/show_bug.cgi?id=773441
+
+2016-10-25 11:46:38 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ * tests/check/meson.build:
+ Revert "meson: move gstreamer-check-1.0 dependency to tests/check"
+ This reverts commit e3c7c17b9b0ff8efb81d23e135178a7be7eaeb1e.
+ Does not actually work. See:
+ https://bugzilla.gnome.org/show_bug.cgi?id=773114#c31
+
+2016-10-24 00:28:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/meson.build:
+ meson: fix build outside of gst-all
+ Unknown variable "apiversion".
+
+2016-10-21 00:32:15 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ * tests/check/meson.build:
+ meson: move gstreamer-check-1.0 dependency to tests/check
+
+2016-10-20 17:17:54 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/audio/meson.build:
+ * meson.build:
+ Revert "meson: Use the new `pic` argument on static libs"
+ This reverts commit e3c22605ae96ee1747020c4f367d49faf6916e14.
+ pic was added after 0.35 and will be present in 0.36 (meson documentation
+ was wrong).
+
+2016-10-20 15:48:34 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/audio/meson.build:
+ * meson.build:
+ meson: Use the new `pic` argument on static libs
+ We depend on 0.35 already
+
+2016-10-14 14:23:38 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/audio/audio.c:
+ audio: don't deref NULL
+ gst_buffer_copy_region() can return NULL when the buffer meta-data is invalid.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=772200
+
+2015-12-04 00:47:38 +1100 Havard Graff <havard.graff@gmail.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: Error-handling for pushing headers
+ https://bugzilla.gnome.org/show_bug.cgi?id=773105
+
+2016-10-13 12:41:29 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * tests/check/libs/videodecoder.c:
+ videodecoder: Default caps sets format I420
+ Also the format must be fixed on the default raw caps. If not
+ gst_video_info_from_caps() will fail and
+ gst_video_decoder_negotiate_default_caps() return FALSE.
+ The test simulates the use case where a gap event is received before
+ the first buffer causing the decoder to fall back to the default caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773103
+
+2016-05-06 16:30:57 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: Plug buffer-leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=773107
+
+2016-10-17 09:46:56 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/audio/audio.c:
+ audio: fix doc string again.
+ There was a second '*' at the start of the line. Reword + reformat to make it
+ obvious.
+
+2016-10-15 22:50:23 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * tests/check/libs/audio.c:
+ tests: add another check for buffer clipping and improve tests
+ Add a test that check that we handle time ranges (a range of time that maps to
+ the same sample).
+ Also update the other tests to use our check api to compare int64 values to get
+ better output on failure.
+
+2016-10-15 21:54:40 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * tests/check/libs/audio.c:
+ tests: clipping in TIME does not use the offset
+ Simplify the test and test only what need to be tested.
+
+2016-10-15 21:30:22 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * tests/check/libs/audio.c:
+ tests: cleanup libs/audio test
+ Split large tests into small tests and name them specifically. Use helpers to
+ avoid repetition. Make sure the order in the file is the same as we add the to
+ the suite.
+
+2016-10-15 22:02:48 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ meson: Don't set c_std to gnu99
+ Use the default for each compiler on every platform instead. This
+ improves our compatibility with compilers that don't have gnu99 as
+ a c_std.
+
+2016-10-15 21:46:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst-libs/gst/audio/meson.build:
+ meson: Add missing audio-enumtypes.h dep in audio-resampler static libs
+ Seen on the Jenkins CI:
+ FAILED: subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o
+ ccache cc '-Isubprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta' '-fdiagnostics-color=always' '-I../subprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/.' '-I../subprojects/gst-plugins-base/.' '-Isubprojects/gst-plugins-base/gst-libs' '-I../subprojects/gst-plugins-base/gst-libs' '-Isubprojects/gstreamer/libs' '-I../subprojects/gstreamer/libs' '-Isubprojects/gstreamer/.' '-I../subprojects/gstreamer/.' '-pipe' '-Wall' '-Winvalid-pch' '-DHAVE_CONFIG_H' '-msse4.1' '-fPIC' '-O0' '-g' '-fPIC' '-I/usr/include/glib-2.0' '-I/usr/lib/glib-2.0/include' '-pthread' '-Isubprojects/gstreamer/gst' '-MMD' '-MQ' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' '-MF' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o.d' -o 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' -c ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c
+ In file included from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h:24:0,
+ from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-private.h:23,
+ from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-macros.h:25,
+ from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.h:23,
+ from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c:24:
+ ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory
+ #include <gst/audio/audio-enumtypes.h>
+ ^
+ compilation terminated.
+
+2016-10-04 17:44:51 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst-libs/gst/tag/meson.build:
+ * meson.build:
+ * tests/check/getpluginsdir:
+ * tests/check/meson.build:
+ meson: Make use of new environment object and set plugin path to builddir
+ Workaround source_root being the root directory of all projects
+ in the subproject case.
+ Remove now unneeded getpluginsdir and define c++ tests in the same loop.
+ Bump meson requirement to 0.35
+
+2016-10-14 14:21:28 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/audio/audio.c:
+ audio: fix typo in doc string
+
+2016-10-13 20:10:09 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/playback/gstdecodebin3-parse.c:
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: More SELECTION_LOCK when linking to slot
+ Since there can be multiple parsebin in a decodebin3,
+ linking parsebin with MultiQueueSlot should be protected also.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772855
+
+2016-10-13 11:42:28 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstplaybin3.c:
+ playback: GstStreamType is a flag
+ Therefor don't use equality
+
+2016-10-11 12:36:00 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstparsebin.c:
+ playback: decodebin3 and parsebin are streams-aware
+ Elements within can add/remove pads at anytime without complying
+ with the fallback system.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772741
+
+2016-10-10 17:08:11 +0900 Wonchul Lee <chul0812@gmail.com>
+
+ * gst/playback/gstparsebin.c:
+ parsebin: re-use existing compare_factories utils func
+ https://bugzilla.gnome.org/show_bug.cgi?id=772676
+
+2016-10-07 12:49:18 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/vorbis/gstvorbisenc.c:
+ vorbisenc: correct codebooks packet identifier from 3 to 5
+ https://bugzilla.gnome.org/show_bug.cgi?id=768763
+
+2016-10-06 16:16:30 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * tests/check/elements/opus.c:
+ opusdec: Fix memory leak in test code
+ gst_caps_to_string function returned allocated memory.
+ So, It should be free using g_free function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772500
+
+2016-10-06 16:24:05 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * tests/check/elements/videorate.c:
+ videorate: Fix memory leakage in test code
+ gst_caps_to_string function returned allocated memory.
+ So, It should be free using g_free function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772501
+
+2016-09-27 09:24:08 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: remove segment stop modification on eos
+ https://bugzilla.gnome.org/show_bug.cgi?id=768763
+
+2016-09-26 16:31:06 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: take audio clip meta into account for buffer duration
+ https://bugzilla.gnome.org/show_bug.cgi?id=768763
+
+2016-09-26 16:25:14 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/vorbis/gstvorbisenc.c:
+ * ext/vorbis/gstvorbisenc.h:
+ vorbisenc: strip after-eos samples from the end of the eos buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=768763
+
+2016-09-30 14:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ video-color: Mark some function arguments as const
+ https://bugzilla.gnome.org/show_bug.cgi?id=771376
+
+2016-10-03 08:56:55 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: require orc 0.4.24 here too
+ Needed for some video stuff. Same requirement as in
+ the autotools build.
+
+2016-10-03 10:59:37 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * config.h.meson:
+ * gst-libs/gst/audio/meson.build:
+ * meson.build:
+ meson: Enable SSE intrinsics in audio-resampler
+ This files need to be built with the specific C flags for the
+ corresponding processor optimisations.
+
+2016-10-03 10:58:09 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * gst-libs/gst/audio/meson.build:
+ * gst-libs/gst/video/meson.build:
+ * gst/adder/meson.build:
+ * gst/videotestsrc/meson.build:
+ * gst/volume/meson.build:
+ * meson.build:
+ meson: Enable Orc in build
+ Top-level meson.build code updated from gst-plugins-good.
+
+2016-09-30 11:35:37 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * hooks/pre-commit.hook:
+ * meson.build:
+ * tests/check/getpluginsdir:
+ meson: Setup pre commit hook and fix getpluginsdir for standalone case
+
+2016-09-30 11:41:10 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+=== release 1.9.90 ===
+
+2016-09-30 13:01:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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 1.9.90
+
+2016-09-30 12:12:12 +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
+
+2016-09-30 11:42:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/de.po:
+ po: Update translations
+
+2016-09-29 19:54:52 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * gst-libs/gst/audio/Makefile.am:
+ audio-resampler: Add a missing header to noinst_HEADERS
+
+2016-09-29 19:45:16 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86-sse41.c:
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ audiorsample: Fix build on 32-bit x86
+ Turns out _mm_cvtsi128_si64() isn't available on 32-bit, so only build
+ SSE 4.1 optimisations on x86-64 for now.
+
+2016-09-28 17:37:38 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * configure.ac:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/audio-resampler-macros.h:
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ * gst-libs/gst/audio/audio-resampler-private.h:
+ * gst-libs/gst/audio/audio-resampler-x86-sse.c:
+ * gst-libs/gst/audio/audio-resampler-x86-sse.h:
+ * gst-libs/gst/audio/audio-resampler-x86-sse2.c:
+ * gst-libs/gst/audio/audio-resampler-x86-sse2.h:
+ * gst-libs/gst/audio/audio-resampler-x86-sse41.c:
+ * gst-libs/gst/audio/audio-resampler-x86-sse41.h:
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audioresample: Separate out CFLAGS used for SSE* code
+ This makes sure that we only build files that need explicit SIMD support
+ with the relevant CFLAGS. This allows the rest of the code to be built
+ without, and specific SSE* code is only called after runtime checks for
+ CPU features.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729276
+
+2016-09-28 19:08:52 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audioresample: Fix some gobject introspection warnings
+
+2016-09-26 10:01:08 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/playback/gstplaybin3.c:
+ playbin3: Remove fallback properties/signals
+ These can all be used via the GstStream API
+ https://bugzilla.gnome.org/show_bug.cgi?id=769079
+
+2016-09-25 22:02:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/meson.build:
+ tests: playbin-complex test needs oggdemux
+
+2016-09-24 21:11:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/videotimecode.c:
+ tests: videotimecode: fix floating point comparisons
+ Comparing floats for equality is not necessarily going to
+ work reliably, so use fail_unless_equals_float() for this.
+ Test would fail on x86 (Intel Atom x5-Z8300).
+
+2016-09-25 16:22:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/adder.c:
+ tests: adder: disable racy flush_start_flush_stop test
+ It's been broken for years, and it's unlikely it will ever
+ be fixed for collectpads/adder now that there's audiomixer
+ which works fine. So let's disable it, since all it does
+ is that it creates noise that distracts from other failures.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708891
+
+2016-09-22 16:15:54 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: take number of bits into account when copying
+ Copy twice the amount of pixels for 16 bits formats.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=747225
+
+2016-09-20 15:12:22 -0400 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Correctly calculate group start times in reverse playback mode
+ We have to calculate from the segment.stop, not the segment.start, as
+ playback goes from stop to start. This fix works around another race
+ condition in streamsynchronizer in my testcase.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=771479
+
+2016-09-20 17:31:55 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/examples/seek/stepping.c:
+ * tests/examples/seek/stepping2.c:
+ examples: seek: fix build with MSVC
+ Use G_PI instead of M_PI. Could also have defined
+ _USE_MATH_DEFINES or included gst/math-compat.h but
+ this seems simplest.
+
+2016-09-19 11:27:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ doc: Add missing map flag to gst_video_frame_map()
+ Add missing map flag, and also add unmap call.
+
+2016-09-17 12:42:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: Fix segment boundary checking for reverse playback
+
+2016-09-14 16:51:30 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: Don't adjust segment time in seek handler
+ basesrc already did that very well for us, adjusting it again on top of
+ that just breaks various non-standard seeks.
+
+2016-09-14 11:29:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Depend on gstreamer 1.9.2.1
+
+2016-09-14 10:14:18 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst-libs/gst/video/video-overlay-composition.c:
+ videooverlaycomposition: document required map flags
+ Fix documentation for gst_video_overlay_composition_blend(). The video frame
+ needs to be mapped with GST_MAP_READWRITE flag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771382
+
+2016-09-12 18:37:21 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/playback/gsturisourcebin.c:
+ playback: Use new gst_bin_set_suppressed_flags() API instead of worrying about the flags in multiple places
+
+2016-09-10 20:50:56 +1000 Jan Schmidt <jan@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b18d820 to f980fd9
+
+2016-09-10 10:05:28 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/videoscale.c:
+ * tests/check/meson.build:
+ tests: videoscale: split test into multiple ones
+ The videoscale test takes eternities to run, that's not
+ great. Split the test into multiple ones. That way they
+ can be run in parallel. Reduces time to run all tests in
+ -base from 29 secs to 12 secs when using meson/ninja.
+
+2016-09-10 09:53:49 +1000 Jan Schmidt <jan@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From f49c55e to b18d820
+
+2016-09-07 17:02:23 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ meson: Raise test timeout to 3 minutes
+ The videoscale testsuite (with 50 tests) last almost 2 minutes here
+
+2016-09-07 14:24:54 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * tests/check/libs/video.c:
+ video/test: Coding style fix
+
+2016-09-05 19:55:58 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/examples/overlay/meson.build:
+ meson: Workaround the qt5 module not letting us now the preprocessor is not avalaible
+ If moc-qt5 is not avalaible, meson breaks:
+ https://github.com/mesonbuild/meson/issues/758
+
+2016-09-05 18:40:19 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/examples/overlay/meson.build:
+ meson: tests: Do not pull qt5 as a hard dependency
+
+2016-09-05 17:43:13 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ * tests/check/getpluginsdir:
+ * tests/check/meson.build:
+ meson: Properly find where GStreamer plugins are when using subprojects
+ And fix building with meson 0.34
+
+2016-09-05 12:22:36 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ meson: Bump version to 1.9.2
+
+2016-08-26 11:30:16 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: safety for failing to determine time length in push mode
+ If we can't find a valid granule near the end of the file, we
+ disable seeking. This guards against the whole file being then
+ read and never going to PLAYING.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770314
+
+2016-08-26 11:27:17 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: increase EOS granpos detection chunk size
+ This can be too small on some files to find a valid granule.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770314
+
+2016-09-04 21:41:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/examples/seek/meson.build:
+ meson: fix joystick header check for jseek example
+
+2016-09-03 11:57:22 +1000 Jonathan Matthew <jonathan@d14n.org>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ pbutils: store missing-plugin structure in current_info->misc again
+ This allows gst_discoverer_info_get_misc to work again, until it
+ finally gets removed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770643
+
+2016-09-04 16:04:00 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: cycle between video tracks without disabling video
+
+2016-09-01 17:56:24 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstrtp.def:
+ win32: Update exports
+
+2016-09-01 22:48:40 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/video-frame.h:
+ video-frame: Expand the range of caps for extended buffer flags
+ The video buffer flags can be applied to encoded video streams,
+ such as video/x-h264 marked up by a demuxer or parser.
+
+2016-09-01 13:07:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybackutils.h:
+ playback: Mark internal functions as G_GNUC_INTERNAL
+
+2016-09-01 14:47:02 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybackutils.c:
+ * gst/playback/gstplaybackutils.h:
+ * gst/playback/gsturidecodebin.c:
+ * gst/playback/gsturisourcebin.c:
+ playbackutils: Move compare_factories_func
+ Move _decode_bin_compare_factories_func function to playbackutils
+ https://bugzilla.gnome.org/show_bug.cgi?id=770692
+
+2016-09-01 09:59:06 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst-libs/gst/video/video-frame.h:
+ video-frame: GST_VIDEO_BUFFER_FLAG are only valid for video/x-raw caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=769771
+
+2016-09-01 09:57:33 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ rtpbuffer: Add buffer flag RETRANSMISSION
+ Useful for elements to know if a buffer is a retransmitted RTP packet.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769771
+
+2016-09-01 12:38:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Require orc >= 0.4.24
+ Needed for being able to compile video.orc
+ https://bugzilla.gnome.org/show_bug.cgi?id=770698
+
+2016-09-01 12:26:40 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.2 ===
+
+2016-09-01 12:26:20 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-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:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ Release 1.9.2
+
+2016-09-01 11:23:10 +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:
+ po: Update translations
+
+2016-09-01 10:53:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/icles/test-colorkey.c:
+ test-colorkey: #define GDK_DISABLE_DEPRECATION_WARNINGS
+ We use gdk_cairo_create() which is deprecated since 3.22.
+
+2016-08-27 11:22:11 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson_options.txt:
+ * tests/examples/app/meson.build:
+ * tests/examples/audio/meson.build:
+ * tests/examples/decodebin_next/meson.build:
+ * tests/examples/dynamic/meson.build:
+ * tests/examples/encoding/meson.build:
+ * tests/examples/fft/meson.build:
+ * tests/examples/gio/meson.build:
+ * tests/examples/meson.build:
+ * tests/examples/overlay/meson.build:
+ * tests/examples/playback/meson.build:
+ * tests/examples/playrec/meson.build:
+ * tests/examples/seek/meson.build:
+ * tests/examples/snapshot/meson.build:
+ * tests/meson.build:
+ meson: build examples
+
+2016-08-27 01:17:25 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/meson.build:
+ meson: enable tests
+ At least on non-Windows platforms.
+
+2016-08-19 11:09:27 -0700 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * ext/ogg/gstoggdemux.c:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ Use the new API to post flow ERROR messages on the bus
+ https://bugzilla.gnome.org/show_bug.cgi?id=770158
+
+2016-08-26 20:48:05 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ build: silence error about pthread for 'make check' in osx
+ Fixes "clang: error: argument unused during compilation: '-pthread'"
+
+2016-08-25 12:19:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/meson.build:
+ meson: update for new files in video lib
+
+2016-08-09 11:39:53 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: forward sticky events on multiqueue
+ When connecting a demuxer through a multiqueue ensure to copy sticky
+ events in order to allow the following factory being properly
+ checked that it is functional.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769580
+
+2016-08-25 11:56:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ win32: Update libgstvideo.def
+
+2016-07-26 19:14:40 +0200 Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video.h:
+ * gst-libs/gst/video/videodirection.c:
+ * gst-libs/gst/video/videodirection.h:
+ * gst-plugins-base.spec.in:
+ * tests/check/libs/gstlibscpp.cc:
+ * tests/check/libs/libsabi.c:
+ * tests/icles/test-header-compile:
+ videodirection: interface for rotation and flip
+ A GstVideoOrientationMethod enumeration is also provided for the
+ admitted property values.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768687
+
+2016-08-17 23:49:02 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/playback/gstparsebin.c:
+ parsebin: do not set global tags to stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=770053
+
+2016-08-12 20:56:31 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * .gitignore:
+ * config.h.meson:
+ * ext/alsa/meson.build:
+ * ext/cdparanoia/meson.build:
+ * ext/libvisual/meson.build:
+ * ext/meson.build:
+ * ext/ogg/meson.build:
+ * ext/opus/meson.build:
+ * ext/pango/meson.build:
+ * ext/theora/meson.build:
+ * ext/vorbis/meson.build:
+ * gst-libs/gst/allocators/meson.build:
+ * gst-libs/gst/app/meson.build:
+ * gst-libs/gst/audio/audio_mkenum.py:
+ * gst-libs/gst/audio/meson.build:
+ * gst-libs/gst/fft/meson.build:
+ * gst-libs/gst/meson.build:
+ * gst-libs/gst/pbutils/meson.build:
+ * gst-libs/gst/pbutils/pbutils_mkenum.py:
+ * gst-libs/gst/riff/meson.build:
+ * gst-libs/gst/rtp/meson.build:
+ * gst-libs/gst/rtp/rtp_mkenum.py:
+ * gst-libs/gst/rtsp/meson.build:
+ * gst-libs/gst/rtsp/rtsp_mkenum.py:
+ * gst-libs/gst/sdp/meson.build:
+ * gst-libs/gst/tag/meson.build:
+ * gst-libs/gst/video/meson.build:
+ * gst-libs/gst/video/video_mkenum.py:
+ * gst-libs/meson.build:
+ * gst/adder/meson.build:
+ * gst/app/meson.build:
+ * gst/audioconvert/meson.build:
+ * gst/audiorate/meson.build:
+ * gst/audioresample/meson.build:
+ * gst/audiotestsrc/meson.build:
+ * gst/encoding/meson.build:
+ * gst/gio/meson.build:
+ * gst/meson.build:
+ * gst/playback/meson.build:
+ * gst/subparse/meson.build:
+ * gst/tcp/meson.build:
+ * gst/typefind/meson.build:
+ * gst/videoconvert/meson.build:
+ * gst/videorate/meson.build:
+ * gst/videoscale/meson.build:
+ * gst/videotestsrc/meson.build:
+ * gst/volume/meson.build:
+ * meson.build:
+ * meson_options.txt:
+ * pkgconfig/meson.build:
+ * sys/meson.build:
+ * sys/ximage/meson.build:
+ * sys/xvimage/meson.build:
+ * tests/check/meson.build:
+ * tests/meson.build:
+ * tools/meson.build:
+ Add support for Meson as alternative/parallel build system
+ https://github.com/mesonbuild/meson
+ With contributions from:
+ Tim-Philipp Müller <tim@centricular.com>
+ Jussi Pakkanen <jpakkane@gmail.com> (original port)
+ Highlights of the features provided are:
+ * Faster builds on Linux (~40-50% faster)
+ * The ability to build with MSVC on Windows
+ * Generate Visual Studio project files
+ * Generate XCode project files
+ * Much faster builds on Windows (on-par with Linux)
+ * Seriously fast configure and building on embedded
+ ... and many more. For more details see:
+ http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
+ http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
+ Building with Meson should work on both Linux and Windows, but may
+ need a few more tweaks on other operating systems.
+
+2016-08-20 11:01:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/.gitignore:
+ tests: ignore new videotimecode test binary
+
+2016-08-19 15:29:13 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst-libs/gst/video/gstvideotimecode.c:
+ videotimecode: Fix false positive coverity issues
+ They are false positive overflows, because coverity doesn't realize that
+ hours <= 24, minutes < 60 and seconds < 60 in all functions. Also casting the
+ number 60 (seconds in minute, minutes in hour) to guint64 for the
+ calculations, in order to avoid overflowing once we allow more than 24-hour
+ timecodes.
+ CIDs #1371459, #1371458
+
+2016-08-18 12:03:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Implement basic support for reverse playback
+ This is enough for making it work in GES, but it's unclear if all the various
+ property combinations are working correctly. It's an improvement over what was
+ there before in any case, which was to just drop all buffers if rate < 0.0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769624
+
+2016-08-12 21:04:03 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst-libs/gst/fft/kiss_fft_s16.h:
+ * gst-libs/gst/fft/kiss_fft_s32.h:
+ gstfft: Use stdint.h instead of _stdint.h
+ _stdint.h is generated by Autotools and we don't really need it.
+ stdint.h is now available on all supported platforms.
+ This really only makes a difference for MSVC, which has it starting from
+ Visual Studio 2015.
+
+2016-08-19 09:27:01 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ * gst-libs/gst/allocators/gstfdmemory.h:
+ fdmemory: add flag to avoid close of the fd
+ Add GST_FD_MEMORY_FLAG_DONT_CLOSE to avoid closing the fd when the
+ memory is freed. When you can guarantee the lifetime of the fd is
+ longer than the memory, this can save a dup() call.
+
+2016-08-17 13:03:43 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst-libs/gst/video/gstvideotimecode.c:
+ videotimecode: Fix various coverity issues
+ Most of them are overflow related and false positives, but coverity can't know
+ that these can't overflow without us giving it more information. Add some
+ assertions for this.
+ One was an actual issue with flags comparison.
+ CIDs #1369051, #1369050, #1369049, #1369048, #1369045
+
+2016-08-08 20:04:11 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/playback/gstplaybin3.c:
+ playbin3: add "element-setup" signal
+ Allows configuration of plugged elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=578933
+
+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
+
+2016-08-04 19:06:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * win32/common/libgstvideo.def:
+ videotimecode: Add to docs and exports list
+
+2016-05-18 19:30:52 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * ext/pango/gsttimeoverlay.c:
+ * ext/pango/gsttimeoverlay.h:
+ timeoverlay: Add support to display timecode
+ Choosing time-mode=time-code will display the time code attached to the
+ buffer, or 00:00:00:00 if no time code is found.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766419
+
+2016-05-14 17:59:20 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/gstvideometa.h:
+ videometa: Added video time code meta
+ It attaches a GstVideoTimeCodeMeta (SMPTE timecode) as metadata to a buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766419
+
+2016-05-14 12:20:38 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/gstvideotimecode.c:
+ * gst-libs/gst/video/gstvideotimecode.h:
+ * gst-libs/gst/video/video.h:
+ * tests/check/Makefile.am:
+ * tests/check/libs/videotimecode.c:
+ videotimecode: Added support for SMPTE time code metadata
+ Can be attached as GstMeta into a video frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766419
+
+2016-07-28 15:04:01 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: don't leak alternate inputs
+ Fix leaks (including parsebin elements) with this pipeline:
+ playbin3
+ uri=http://127.0.0.1:8079/defaults/exMPD_BIP_TC1/exMPD_BIP_TC1.mpd
+ https://bugzilla.gnome.org/show_bug.cgi?id=769270
+
+2016-08-01 16:00:29 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * ext/ogg/gstoggparse.c:
+ ogg: check return values in gst_ogg_parse_new_stream
+ Return NULL in gst_ogg_parse_new_stream when either ogg_stream_pagein() or
+ gst_ogg_stream_setup_map() failed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769299
+
+2016-08-01 15:52:11 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * ext/ogg/gstoggparse.c:
+ ogg: fix memory leak in gst_ogg_parse_new_stream
+ Avoid leaking the stream object
+ https://bugzilla.gnome.org/show_bug.cgi?id=769299
+
+2016-08-01 13:35:16 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: fix output->decoder_{sink,src} leak
+ output->decoder_sink and output->decoder_src are both going to be
+ replaced in the 2 branches of the following 'if'.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769270
+
+2016-08-01 12:37:43 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: fix tag list leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=769270
+
+2016-08-01 12:28:20 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: consume select-streams event
+ https://bugzilla.gnome.org/show_bug.cgi?id=769270
+
+2016-07-28 15:44:27 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/examples/decodebin_next/decodebin3.c:
+ * tests/examples/decodebin_next/playbin-test.c:
+ decodebin_next: fix caps and tags leaks
+ The getters are (transfer full).
+ https://bugzilla.gnome.org/show_bug.cgi?id=769270
+
+2016-07-28 14:46:34 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: fix collection ref handling
+ gst_stream_collection_add_stream() consumes the collection reference
+ passed to it but gst_stream_collection_get_stream() is (transfer none).
+ Fix this pipeline:
+ playbin3
+ uri=http://127.0.0.1:8079/defaults/exMPD_BIP_TC1/exMPD_BIP_TC1.mpd
+ https://bugzilla.gnome.org/show_bug.cgi?id=769270
+
+2016-07-29 11:38:44 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: handle full removal of streams
+ Fix the
+ validate.file.playback.disable_subtitle_track_while_paused.* validate
+ scenarios when using playbin3.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769298
+
+2016-08-02 12:03:18 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Remove sample rate and channel count boundaries in caps
+ WAV is too generic to impose more-or-less arbitrary boundaries on the
+ sample rate and channel count caps. For example, there are 384 kHz WAV
+ files. Another example: it is in theory possible that somebody puts DSD
+ data into a WAV file, which will then have a sample rate of ~2.8 MHz.
+ For this reason, get rid of the rate and channel caps unless they are
+ fixed values. Downstream anyway usually knows the limitations better.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761514
+
+2016-07-29 15:51:35 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst-libs/gst/pbutils/codec-utils.c:
+ pbutils: Add more h264 scalable profiles
+ Adding Scalable Constrained High (G.10.1.2.1) and
+ Scalable High Intra(G.10.1.3) profiles to the profile list
+ https://bugzilla.gnome.org/show_bug.cgi?id=769303
+
+2016-07-26 17:46:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ rtpbuffer: Add some const qualifiers
+ gst_rtp_buffer_add_extension_onebyte_header() and
+ gst_rtp_buffer_add_extension_twobytes_header() can have a const argument for
+ the actual extension data.
+
+2015-12-26 13:19:01 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/playback/gstparsebin.c:
+ parsebin: maintain original order when creating fallback stream collection
+
+2016-03-20 14:37:03 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Send stream-group-done to unblock downstream
+ When processing EOS for a pad, send a stream-group-done
+ for the pad in case downstream is waiting for more
+ data on this stream before it can process related
+ streams from the group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768995
+
+2016-07-22 14:40:25 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstplaybin3.c:
+ playbin3: fix collection leak
+ The collection referenced owned by playbin3 was not released when it was
+ destroyed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769080
+
+2016-07-22 14:35:17 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: fix collection refcounting
+ My collection leak fix 83f30627cd9460157935e7e9603c60a15555967e
+ introduced a crash in this scenario: audiotestsrc ! decodebin3 ! fakesink
+ The reference handling of collection in decodebin3 wasn't very clear and
+ my attempt to fix the leak introduced a regression where we went one
+ reference short in some other scenarios.
+ Fixing this by:
+ - Giving a strong reference to DecodebinInput making things clearer
+ - Fixing get_merged_collection() which was sometimes returning an
+ existing reference and sometimes a new one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769080
+
+2016-07-23 14:42:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/.gitignore:
+ * tests/check/libs/.gitignore:
+ Add more files to .gitignore
+
+2016-07-22 14:42:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/playback/gsturisourcebin.c:
+ docs: urisourcebin: fix typo
+
+2016-07-22 23:21:36 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstparsebin.c:
+ * gst/playback/gstplaybin3.c:
+ * gst/playback/gsturisourcebin.c:
+ playback: Flesh out docs a bit for new elements
+ Add some more text to the docs for urisourcebin,
+ parsebin, decodebin3 and playbin3, including a warning
+ that they are unstable API for now
+
+2016-07-22 12:52:12 +0100 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.signals:
+ * gst/playback/gstparsebin.c:
+ * gst/playback/gstplaybin3.c:
+ docs: add playbin3, decodebin3, parsebin, urisourcebin to docs
+ Docs still need some fleshing out though.
+
+2016-07-13 18:29:52 +0900 Arun Raghavan <arun@arunraghavan.net>
+
+ * ext/vorbis/gstvorbisenc.c:
+ Revert "vorbisenc: push an updated segment stop time when we know it"
+ This reverts commit a16cd5d2a5cbdf084163ead68b59d537d7db99f7.
+ Setting the stop time on the segment breaks reconfiguration, as the
+ encoder signals an EOS, but we reconfigure it an continue to produce
+ buffers.
+ This information should not be required via the segment downstream
+ since we already have the sample count being used to generate buffer
+ durations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768763
+
+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
+
+2016-07-20 12:17:57 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: use GST_STIME_FORMAT for the correct argument
+ GST_STIME_ARGS is used for time_offset not for last_time.
+ This fixes the format string accordingly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768990
+
+2016-07-19 18:20:57 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: after a reset, recalculate the ouput size
+ After we reset the resampler, there is no history anymore in the resampler
+ and the previously calculated output size is no longer valid.
+ Recalculate the new output size after a reset to make sure we don't try
+ to convert too much.
+
+2016-07-19 13:26:06 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: fix some leaks
+ Fixes check-valgrind for subparse test.
+
+2016-07-18 17:26:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/appsink.c:
+ tests: appsink: add minimal test for new pull with timeout functions
+ https://bugzilla.gnome.org/show_bug.cgi?id=768852
+
+2016-07-15 13:20:29 +0200 Joan Pau Beltran <joanpau.beltran@socib.cat>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ * win32/common/libgstapp.def:
+ appsink: add _pull_sample/preroll() variants with timeout
+ The _pull_sample() and _pull_preroll() functions block
+ until a sample is available, EOS happens or the pipeline
+ is shut down (returning NULL in the last two cases).
+ This adds _try_pull_sample() and _try_pull_preroll()
+ functions with a timeout argument to specify the maximum
+ amount of time to wait for a new sample.
+ To avoid code duplication, wait forever if the timeout is
+ GST_CLOCK_TIME_NONE and use that to implement
+ _pull_sample/_pull_preroll with the original behavior.
+ Add also corresponding action signals "try-pull-sample"
+ and "try-pull-preroll".
+ https://bugzilla.gnome.org/show_bug.cgi?id=768852
+
+2016-07-13 14:17:25 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: actually check result of accept caps query
+ We were just checking if the query was handled, not its result.
+ Also fix a leak as gst_pad_query() was not consuming the query.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768811
+
+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
+
+2016-07-18 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
+
+2016-07-07 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
+
+2016-07-14 10:33:38 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstplaybin3.c:
+ playbin3: fix stream leak
+ The stream returned by gst_message_streams_selected_get_stream() is
+ reffed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768811
+
+2016-07-13 16:16:21 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstparsebin.c:
+ decodebin3: fix collection leak
+ The collection owned by GstDecodebin3 has to be unreffed when disposing.
+ gst_event_new_stream_collection() doesn't consume the collection passed
+ to it so no need to give it an extra ref.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768811
+
+2016-07-14 10:34:30 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3-parse.c:
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: fix stream leaks
+ MultiQueueSlot owns a ref on the active stream so it should release it
+ when being freed.
+ DecodebinInputStream owns ref on the active and pending stream so they
+ should be dropped when being freed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768811
+
+2016-07-14 14:24:23 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstparsebin.c:
+ decodebin3: fix event leaks
+ Returning GST_PAD_PROBE_HANDLED means we are taking care of unreffing
+ the probe info.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768811
+
+2016-07-14 16:29:39 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstparsebin.c:
+ decodebin3: fix caps leaks
+ gst_stream_get_caps() returns a reffed caps.
+ The caps passed to gst_query_set_caps_result() are not transfered.
+ The caps in gst_parse_pad_stream_start_event() was either acquired
+ using gst_pad_get_current_caps() which returns a new ref or
+ explicitly reffed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768811
+
+2016-07-15 19:48:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtp: rtpbasedepayload: simplify code
+ Remove unnecessary helper struct for callbacks. The bclass
+ member of the helper struct was not used, so we can just
+ remove it and the GET_CLASS() call and simplify the whole
+ affair by passing the depayloader directly to the callback.
+
+2016-07-13 16:02:25 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstplaybin3.c:
+ * tests/examples/decodebin_next/decodebin3.c:
+ * tests/examples/decodebin_next/playbin-test.c:
+ playbin3: fix leaks of collection returned by message parse API
+ gst_message_parse_stream_collection() and
+ gst_message_parse_streams_selected() actually return a reffed
+ GstStreamCollection.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768776
+
+2016-07-15 22:47:02 +1000 Jan Schmidt <jan@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: Allow disabling audio/video/subtitle tracks
+ When cycling through tracks, add 'disable' to the set
+ of states.
+
+2016-06-24 12:25:30 +1000 Jan Schmidt <jan@centricular.com>
+
+ * ext/alsa/gstalsasink.h:
+ alsasink: Remove unused hwparam/swparam pointers
+ The ALSA params structures aren't kept. The pointers
+ aren't used anywhere, so remove them from the struct.
+
+2016-07-13 15:45:33 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tools/gst-device-monitor.c:
+ tools: fix device leaks in gst-device-monitor
+ gst_message_parse_device_{added,removed} is actually returning a new ref
+ on the device.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768776
+
+2016-07-12 12:03:53 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/videoscale.c:
+ videoscale: fix bus leak in test
+ gst_bus_add_signal_watch() takes a ref on the bus which should be
+ released using gst_bus_remove_signal_watch().
+ https://bugzilla.gnome.org/show_bug.cgi?id=768718
+
+2016-07-11 19:17:41 +0200 Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+ * gst-libs/gst/video/videoorientation.c:
+ videoorientation: Use G_DEFINE_INTERFACE instead of a manually written get_type()
+ https://bugzilla.gnome.org/show_bug.cgi?id=768687
+
+2016-07-12 00:13:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-format.c:
+ video: Fix some compiler warnings for out-of-range enum values
+ https://bugzilla.gnome.org/show_bug.cgi?id=767816
+
+2016-07-11 21:13:37 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f363b32 to f49c55e
+
+2016-07-10 10:28:44 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ * tests/check/elements/appsrc.c:
+ appsrc: Remove trailing whitespace
+ https://bugzilla.gnome.org/show_bug.cgi?id=768510
+
+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
+
+2016-07-08 15:44:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/pbutils-private.h:
+ pbutils: Mark private functions as G_GNUC_INTERNAL
+
+2016-07-07 17:37:51 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: don't reset allowed tags
+ When a discont buffer is processed, the state is re-initialized, which
+ nullifies the allowed_tags.
+ The problem is when a subrip string with tags is processed and allowed_tags is
+ NULL. The function subrip_unescape_formatting() calls g_strjoinv with a
+ str_array as NULL, leading to a GLib-CRITICAL.
+ This patch removes the allowed_tags resetting, in parser_state_init(), but
+ move it into gst_sub_parse_format_autodetect().
+ https://bugzilla.gnome.org/show_bug.cgi?id=768525
+
+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
+
+2016-07-07 17:29:34 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: fix compilation on big-endian
+
+2016-07-07 17:10:17 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: More trickmode fix
+ We need to take into account the input segment flags to know whether
+ we should drain the decoder after a new keyframe in trick mode.
+ Otherwise we would have to wait for the next frame to be outputted (and
+ the segment to be activated) which ... well ... kind of beats the whole
+ point of this draining :)
+
+2016-07-06 21:13:19 +0200 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES
+ https://bugzilla.gnome.org/show_bug.cgi?id=768495
+
+2016-07-07 00:27:00 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Ship audio-resampler-neon.h
+
+2016-07-06 16:14:32 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ * tests/examples/playback/playback-test.c:
+ tests: correctly print guintptr on mac
+
+2016-07-06 13:51:00 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.1 ===
+
+2016-07-06 13:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-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-libs/gst/video/video-orc-dist.c:
+ * 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 1.9.1
+
+2016-07-06 11:42:29 +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
+
+2016-07-06 10:18:00 +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:
+ po: Update translations
+
+2016-06-30 16:36:27 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Take stream lock one time only on drain
+ When the drain is triggered from the chain function the lock is already
+ taken so there is no need to take it one more time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767641
+
+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 11:12:25 +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 11:07:54 +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 11:00:51 +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 10:52:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Use the object lock to protect bytes/time tracking
+
+2016-07-04 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
+
+2016-03-17 00:19:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: If do-timestamp=TRUE, capture the time when the buffer was pushed to the source
+ ... instead of the time when it was pushed further downstream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763630
+
+2016-04-29 00:59:42 -0700 Zaheer Abbas Merali <zaheermerali@gmail.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ basertpdepayload: create valid segment when given non-time segment
+ This will become an error in 1.10.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765796
+
+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
+
+2016-06-30 17:26:56 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/playback/gstparsebin.c:
+ parsebin: Fix authors and description
+
+2016-06-30 17:26:14 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplayback.c:
+ * gst/playback/gstplayback.h:
+ * gst/playback/gsturidecodebin3.c:
+ playback: Remove uridecodebin3
+ This was committed by mistake. The solution forward is to use the
+ appropriate combination of urisourcebin and decodebin3
+
+2016-06-29 18:14:51 +0200 Edward Hervey <edward@centricular.com>
+
+ * configure.ac:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstdecodebin3-parse.c:
+ * gst/playback/gstdecodebin3.c:
+ * gst/playback/gstparsebin.c:
+ * gst/playback/gstplayback.c:
+ * gst/playback/gstplayback.h:
+ * gst/playback/gstplaybin3.c:
+ * gst/playback/gsturidecodebin3.c:
+ * gst/playback/gsturisourcebin.c:
+ * tests/examples/Makefile.am:
+ * tests/examples/decodebin_next/.gitignore:
+ * tests/examples/decodebin_next/Makefile.am:
+ * tests/examples/decodebin_next/decodebin3.c:
+ * tests/examples/decodebin_next/playbin-test.c:
+ playback: New elements
+ With contributions from Jan Schmidt <jan@centricular.com>
+ * decodebin3 and playbin3 have the same purpose as the decodebin and
+ playbin elements, except make usage of more 1.x features and the new
+ GstStream API. This allows them to be more memory/cpu efficient.
+ * parsebin is a new element that demuxers/depayloads/parses an incoming
+ stream and exposes elementary streams. It is used by decodebin3.
+ It also automatically creates GstStream and GstStreamCollection for
+ elements that don't natively create them and sends the corresponding
+ events and messages
+ * Any application using playbin can use playbin3 by setting the env
+ variable USE_PLAYBIN3=1 without reconfiguration/recompilation.
+
+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 20:53:37 +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.
+
+2016-06-27 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.
+
+2016-06-24 01:56:11 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * configure.ac:
+ configure: Need to add -DGST_STATIC_COMPILATION when building only statically
+ https://bugzilla.gnome.org/show_bug.cgi?id=767463
+
+2016-06-23 10:22:35 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: demote an expected error to debug
+ Dropping a buffer because we have a seek pending is normal,
+ and will now happen when we trigger a seek while going through
+ the packets in a page. So this should not be an error.
+
+2016-06-22 16:02:37 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-resampler.c:
+ * gst-libs/gst/video/video-resampler.h:
+ * gst-libs/gst/video/video-scaler.c:
+ video-converter: fix interlaced scaling some more
+ Fix problem with the line cache where it would forget the first line in
+ the cache in some cases.
+ Keep as much backlog as we have taps. This generally works better and we
+ could do even better by calculating the overlap in all taps.
+ Allocated enough lines for the line cache.
+ Use only half the number of taps for the interlaced lines because we
+ only have half the number of lines.
+ The pixel shift should be relative to the new output pixel size so scale
+ it.
+ Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=767921
+
+2016-06-21 14:53:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ plugin-doc: Minor re-order
+
+2016-06-21 14:40:17 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ Automatic update of plugins doc files
+
+2016-06-21 18:04:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/discoverer.c:
+ tests: discoverer: handle missing ogg/codec plugins gracefully
+ https://bugzilla.gnome.org/show_bug.cgi?id=767859
+
+2016-06-21 11:45:49 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From ac2f647 to f363b32
+
+2016-06-20 12:42:28 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: handle missing buffers with no duration
+ If buffer duration is missing, it is parsed from the packet data.
+ This is not foolproof, since Opus can change durations on the
+ fly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767826
+
+2016-06-17 15:11:20 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: preserve duration when skipping a tag at the beginning of a buffer
+ gst_buffer_copy_region() does not copy the duration if it doesn't start
+ with the first byte. We just skip the tag here, so the duration is still
+ valid.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767791
+
+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
+
+2016-04-14 14:02:27 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix audio glitches with low bitrate vorbis
+ A low bitrate stream which can pack more than 2 seconds of audio
+ in a page would cause the stream's position to be updated not
+ often enough, and would trigger a spurious "jump" via a GAP
+ event. Instead, we update the stream position after calculating
+ the new overall segment position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764966
+
+2016-06-16 10:55:52 +0100 Mikhail Fludkov <misha@pexip.com>
+
+ * tests/check/elements/opus.c:
+ opusdec: test for PLC timestamp when FEC is enabled.
+
+2016-04-05 12:41:45 +0200 Mikhail Fludkov <misha@pexip.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * tests/check/libs/audiodecoder.c:
+ audiodecoder: fix invalid timestamps when PLC and delay
+ Elements inherited from GstAudioDecoder, supporting PLC and introducing
+ delay produce invalid timestamps. Good example is opusdec with in-band FEC
+ enabled. After receiving GAP event it delays the audio concealment until
+ the next buffer arrives. The next buffer will have DISCONT flag set which
+ will make GstAudioDecoder to reset it's internal state, thus forgetting
+ the timestamp of GAP event. As a result the concealed audio will have the
+ timestamp of the next buffer (with DISCONT flag) but not the timestamp
+ from the event.
+
+2016-06-11 17:11:30 +0200 Paulo Neves <pneves@airborneprojects.com>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ exiftag: Increase serialized geo precision
+ The serialization of double typed geographical
+ coordinates to DMS system supported by the exif
+ standards was previously truncated without need.
+ The previous code truncated the seconds part of
+ the coordinate to a fraction with denominator
+ equal to 1 causing a bug on the deserialization
+ when the test for the coordinate to be serialized
+ was more precise.
+ This patch applies a 10E6 multiplier to the numerator
+ equal to the denominator of the rational number.
+ Eg. Latitude = 89.5688643 Serialization
+ DMS Old code = 89/1 deg, 34/1 min, 7/1 sec
+ DMS New code = 89/1 deg, 34/1 min, 79114800UL/10000000UL
+ Deserialization
+ DMS Old code = 89.5686111111
+ DMS New code = 89.5688643
+ The new test tries to serialize a higher precision
+ coordinate.
+ The types of the coordinates are also guint32 instead
+ of gint like previously. guint32 is the type of the
+ fraction components in the exif.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767537
+
+2016-06-10 22:36:32 -0400 Thomas Jones <thomas.jones@utoronto.ca>
+
+ * gst-libs/gst/pbutils/gstaudiovisualizer.c:
+ audiovisualizer: Fix calculations for bytes<->samples conversions
+ Use bpf instead of channels * sizeof(gint16).
+ https://bugzilla.gnome.org/show_bug.cgi?id=767505
+
+2016-06-10 14:04:36 -0400 Thomas Jones <thomas.jones@utoronto.ca>
+
+ * gst-libs/gst/pbutils/gstaudiovisualizer.c:
+ audiovisualizer: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
+ https://bugzilla.gnome.org/show_bug.cgi?id=767506
+
+2016-06-10 22:50:41 -0400 Thomas Jones <thomas.jones@utoronto.ca>
+
+ * gst-libs/gst/pbutils/gstaudiovisualizer.c:
+ audiovisualizer: fix timestamp calculation for audio channels > 1
+ We have to use bps*channels instead of just bps, which is exactly what bpf is for.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767507
+
+2015-04-09 19:09:17 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: handle buffer's flags at offset
+ For reverse playback it is important to handle correctly the frame sync
+ points, which is set when the input buffer doesn't have the DELTA_UNIT flag.
+ This is handled correctly when decoder is packetized, but when it is not the
+ frame's sync point is not copied, and the reverse playback never decodes frame
+ batches.
+ The current patch adds the buffer's flags to the Timestamp list, where the
+ timestamp and duration of the input buffers are hold.
+
+2015-04-09 19:18:58 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: squash two message logs into one
+ There were two consecutive log messages in gst_video_decoder_decode_frame().
+ Given the information they provide, it is more efficient to squash them into a
+ single one.
+
+2015-04-09 19:16:10 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: playback rate is in input_segment
+ The playback rate is hold in the input_segment member variable, not in the
+ output_segment, and the parse_gather list was never filled because of that.
+ This patch changes the comparison with input_segment.
+
+2016-06-09 19:02:49 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Use input segment rate instead of output segment rate to decide whether the drain on keyframes
+ The output segment is only set up after data is output, which might be far in
+ the future for reverse playback. Also we are here interested in the state at
+ the current *input* frame (which is the keyframe), not any possible output.
+
+2016-06-09 18:53:54 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Only drain in KEY_UNITS trick mode after a keyframe in forwards playback mode
+ For reverse playback the same behaviour was already implemented in
+ flush_parse().
+ For reverse playback, chain_forward() is only used to gather frames and not
+ for decoding, and it is actually called by the draining logic, causing an
+ infinite recursion.
+
+2016-06-07 09:48:35 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Don't push late frames
+ While it's a bit tricky to discard frames *before* decoding (because
+ we might not be sure which data is needed or not by the decoder), we
+ can discard them after decoding if they are too late anyway.
+ Any following basetransform based element or similar would drop the frame too.
+
+2016-06-07 10:31:59 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Avoid recursive drain/flush calls
+ _chain_forward() can also be called with reverse playback. Blindly
+ calling drain_out() on DISCONT buffers would end up in a recursive
+ call.
+
+2016-06-04 09:51:17 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Drain out keyframes in TRICK_MODE_KEY_UNITS
+ When asked to just decode keyframe, if we got a keyframe drain out
+ the decoder straight away.
+ This avoids having to wait for the next frame and reduces delay even
+ more.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767232
+
+2016-06-04 09:49:00 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Drain decoder on DISCONT buffers
+ This ensures the decoder is properly drained out when receiving a
+ DISCONT buffer. The optimal way of doing this would have been to
+ receive a GAP event before hand but it is not always possible.
+ Fixes big delays with some decoders (ex gst-libav) that will not
+ drain out data when only decoding keyframes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767232
+
+2016-06-01 11:02:12 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: preserve timestamp when skipping a tag at the beginning of a buffer
+ gst_buffer_copy_region() does not copy the timestamp if it doesn't start
+ with the first byte. We just skip the tag here, so the timestamp is still
+ valid.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767173
+
+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
+
+2016-06-02 13:07:01 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * gst/playback/gstsubtitleoverlay.c:
+ opusenc, subtitleoverlay: use MAY_BE_LEAKED flag
+ Flag caps that are cached locally and will never be freed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767155
+
+2016-06-01 16:56:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Create a new decode element with the parser/convert capsfilter if there is a multiqueue after the parser
+ https://bugzilla.gnome.org/show_bug.cgi?id=767102
+
+2016-05-23 15:11:53 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Make sure the DISCONT flag is set on the outgoing buffer
+ The base class was setting the DISCONT flag before checking whether the buffer
+ would be in segment or not.
+ Fix issues with DISCONT flags not being properly propagated downstream when
+ decoders buffers were out of segment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766800
+
+2016-06-01 15:31:52 +0200 Joan Pau Beltran <joanpau.beltran@socib.cat>
+
+ * docs/design/part-mediatype-video-raw.txt:
+ docs: design: add IYU2 raw video format description
+ https://bugzilla.gnome.org/show_bug.cgi?id=763026
+
+2016-06-01 12:36:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ textoverlay: enable shaded background drawing for new IYU2 format
+
+2016-05-30 16:40:26 +0200 Joan Pau Beltran <joanpau.beltran@socib.cat>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-scaler.c:
+ * tests/check/libs/video.c:
+ video: add IYU2 format
+ This existed in 0.10 and is needed by dc1394src.
+ IYU2 format is a YUV fully-sampled packed format similar to v308
+ but with different component order (U-Y-V instead of Y-U-V).
+ http://www.fourcc.org/yuv.php#IYU2
+ https://bugzilla.gnome.org/show_bug.cgi?id=763026#c5
+
+2016-03-17 23:47:48 +0530 Nirbheek Chauhan <nirbheek.chauhan@gmail.com>
+
+ * ext/libvisual/visual.c:
+ libvisual: Factor out endian-order RGB formats
+ MSVC seems to ignore preprocessor conditionals inside static
+ pad templates. Also remove unnecessary quotes inside caps strings.
+
+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.
+
+2016-05-23 19:28:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opus: use default error messages in some more cases
+
+2016-05-23 15:35:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: use default error message strings in more cases
+ Details should go into the debug message. We should probably
+ make up new codes for encoder/decoder lib init failures too.
+
+2016-05-19 12:26:05 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opus: Post error message on GST_FLOW_ERROR
+ https://bugzilla.gnome.org/show_bug.cgi?id=766265
+
+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
+
+2016-05-16 09:52:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/playbin.c:
+ tests: playbin: add test for new "element-setup" signal
+ https://bugzilla.gnome.org/show_bug.cgi?id=578933
+
+2016-05-14 11:28:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: add "element-setup" signal
+ Allows configuration of plugged elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=578933
+
+2016-05-15 14:43:11 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * gst-libs/gst/app/.gitignore:
+ * gst-libs/gst/app/gstapp-marshal.list:
+ app: remove marshaller files from git
+
+2016-05-15 14:37:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ app: use generic marshallers
+
+2016-05-15 12:01:17 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Reset keyframe_granule when needed
+ This avoids ending up with bogus values when doing flushing seeks
+ in push-mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766467
+
+2016-05-15 13:31:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * 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:
+ docs: Update for git master
+
+2016-05-14 15:43:24 +0300 Matthew Waters <matthew@centricular.com>
+
+ * gst-libs/gst/video/gstvideoaffinetransformationmeta.h:
+ video/affinetransformationmeta: define the coordinate space used
+ Based on the expected output from the already existing usage by androidmedia
+ and the opengl plugins.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764667
+
+2015-12-17 19:38:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for WebVTT
+
+2015-09-30 17:55:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/playsink.c:
+ tests: playsink: add minimal test for playsink element
+ Attempt to reproduce leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755867
+
+2016-05-10 12:17:34 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/vorbistag.c:
+ vorbistag: fix buffer leaks in tests
+ It internally uses gst_check_chain_func() so we
+ should call gst_check_drop_buffers() when tearing down tests to free
+ the buffers which have been exchanged through the pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766226
+
+2016-05-10 12:17:34 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/appsrc.c:
+ appsrc: fix buffer leaks in tests
+ It internally uses gst_check_chain_func() so we
+ should call gst_check_drop_buffers() when tearing down tests to free
+ the buffers which have been exchanged through the pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766226
+
+2016-05-10 12:17:34 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/audiorate.c:
+ audiorate: fix buffer leaks in tests
+ It internally uses gst_check_chain_func() so we
+ should call gst_check_drop_buffers() when tearing down tests to free
+ the buffers which have been exchanged through the pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766226
+
+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
+
+2016-05-10 16:44:04 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ * win32/common/libgstapp.def:
+ appsrc: Add duration property for providing a duration in TIME format
+ https://bugzilla.gnome.org/show_bug.cgi?id=766229
+
+2016-05-10 10:01:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * gst-libs/gst/video/gstvideoencoder.h:
+ videodecoder/encoder: Correct GST_IS_*CODER_CLASS macros
+ They are currently not used, but would result in a compiler error due to wrong
+ variable name usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766203
+
+2016-05-05 13:16:57 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/tcp/gstmultihandlesink.c:
+ multihandlesink: Warn if trying to change the state from the streaming thread
+ Instead of silently returning GST_STATE_CHANGE_FAILURE.
+
+2016-05-04 11:33:50 +1000 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: an element can negotiate before we block it
+ When we initialize an element in decodebin, we 1) set it to PAUSED and
+ push sticky events on its sinkpad to trigger negotiation 2) block its
+ src pad(s) to detect CAPS events. We can't block before 1) as that
+ would lead to a deadlock.
+ It's possible (and common) tho that an element configures its srcpad
+ during 1) and before 2). Therefore before this change we would
+ typically block and expose an element's pad only once the element
+ output its first buffer, triggering sticky events to be resent. One
+ consequence of this behaviour is that it sometimes broke
+ renegotiation.
+ With this change now we consider a pad ready to be exposed when it's
+ ->blocked or has fixed caps (which were set before we could block it).
+ https://bugzilla.gnome.org/show_bug.cgi?id=765456
+
+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
+
+2016-04-29 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 11:21:47 +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 11:18:23 +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 11:15:53 +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
+
+2015-08-21 10:40:33 +0200 Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ exiftag: handle GST_TAG_CAPTURING_FOCAL_LENGTH_35_MM tag
+ This tag match the EXIF_TAG_FOCAL_LENGTH_IN_35_MM_FILM exif tag and is
+ stored on a short. Hence there is a precision loss compared to the
+ GstTag which is a double value.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753930
+
+2015-08-21 10:39:36 +0200 Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ tag: add GST_TAG_CAPTURING_FOCAL_LENGTH_35_MM tag
+ It is the 35 mm equivalent focal length of the lens, mainly used in
+ photography. Tag value is stored in a double value to be consistent with
+ GST_TAG_CAPTURING_FOCAL_LENGTH.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753930
+
+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
+
+2016-04-27 18:08:46 +0900 Kipp Cannon <kipp.cannon@ligo.org>
+
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/audio.h:
+ audio: Add const to segment parameter of gst_audio_buffer_clip()
+ e.g., allows this to be used with the reference retrieved by
+ gst_event_parse_segment().
+ https://bugzilla.gnome.org/show_bug.cgi?id=765663
+
+2016-04-21 08:45:40 +0200 Jakub Adam <jakub.adam@ktknet.cz>
+
+ * sys/ximage/ximagesink.c:
+ ximagesink: generate reconfigure on window handle change
+ When ximagesink is given a new window handle, it should check
+ its geometry and if the size of the new window differs from
+ the previous one, create reconfigure event in order to get
+ a chance to negotiate a more suitable image resolution with
+ the upstream elements.
+ We can't rely on receiving Expose or ConfigureNotify from
+ the X server for the newly assigned window, which would also
+ generate reconfigure.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765424
+
+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 16:48:36 +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 16:47:00 +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 16:06:39 +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.
+
+2016-04-22 15:07:10 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: add more fastpaths for I420 -> RGB
+ Use the I420->BGRA and a new I420->ARGB to speed up any I420 to RGB
+ operation.
+
+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.
+
+2016-04-17 16:21:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/pipelines/vorbisenc.c:
+ tests: vorbisenc: fix with CK_FORK=no
+
+2016-04-12 16:32:20 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Always add a multiqueue in single-stream use-buffering pipelines
+ If we are configured to use buffering and there is no demuxer in the chain, we
+ still want a multiqueue, otherwise we will ignore the use-buffering property.
+ In that case, we will insert a multiqueue after the parser or decoder - not
+ elsewhere, otherwise we won't have timestamps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764948
+
+2016-04-18 17:39:02 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tools/gst-play.c:
+ gst-play: call gst_deinit()
+ So we can use gst-play to track memory leaks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765216
+
+2016-04-15 17:48:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstsdp.def:
+ win32: update .def for new API
+
+2016-04-16 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
+
+2016-04-15 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
+
+2016-04-15 12:54:32 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: fix build with GCC 4.6.3
+ gstsubparse.c: In function ‘parse_subrip’:
+ gstsubparse.c:988:7: error: ignoring return value of ‘strtol’, declared with attribute warn_unused_result [-Werror=unused-result]
+ cc1: all warnings being treated as errors
+ https://bugzilla.gnome.org/show_bug.cgi?id=765042
+
+2016-04-15 13:08:38 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/icles/.gitignore:
+ .gitignore: add test-resample binary
+
+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
+
+2016-04-14 10:00:06 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 6f2d209 to ac2f647
+
+2016-04-13 10:07:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/video-multiview.c:
+ * gst-libs/gst/video/video-overlay-composition.c:
+ videometa: Initialize all fields of all metas with default values
+ The metas are not allocated with all fields initialized to zeroes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764902
+
+2016-04-11 15:28:00 +0000 Arjen Veenhuizen <arjen.veenhuizen@tno.nl>
+
+ * gst-libs/gst/video/gstvideometa.c:
+ videometa: Explicitly initialize GstVideoCropMeta on init
+ It is not allocated with all fields initialized to 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764902
+
+2016-03-21 16:34:37 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/alsa/gstalsa.c:
+ alsa: properly convert position-less channels from ALSA
+ The only way for ALSA to expose a position-less multi channels is to
+ return an array full of SND_CHMAP_MONO. Converting this to a
+ GST_AUDIO_CHANNEL_POSITION_MONO array would be invalid as
+ GST_AUDIO_CHANNEL_POSITION_MONO is meant to be used only with one
+ channel.
+ Fix this by using GST_AUDIO_CHANNEL_POSITION_NONE which is meant to be
+ used for position-less channels.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763799
+
+2016-03-21 16:29:39 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: don't attempt to reorder position-less channels
+ As said in its doc GST_AUDIO_CHANNEL_POSITION_NONE is meant to be used
+ for "position-less channels, e.g. from a sound card that records 1024
+ channels; mutually exclusive with any other channel position".
+ But at the moment using such positions would raise a
+ 'g_return_if_reached' warning as gst_audio_get_channel_reorder_map()
+ would reject it.
+ Fix this by preventing any attempt to reorder in such case as that's not
+ what we want anyway.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763799
+
+2016-03-21 07:26:50 -0400 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audio: add debug output if channels mapping does not match
+ https://bugzilla.gnome.org/show_bug.cgi?id=763985
+
+2016-03-21 11:58:13 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/alsa/gstalsa.c:
+ alsa: add some debugging output to alsa_detect_channels_mapping()
+ https://bugzilla.gnome.org/show_bug.cgi?id=763985
+
+2016-03-21 11:46:45 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/audio-channels.c:
+ * gst-libs/gst/audio/audio-channels.h:
+ * win32/common/libgstaudio.def:
+ gst-audio: add gst_audio_channel_positions_to_string()
+ We currently don't log much about channel positions making debugging
+ harder as it should be. This is the first step in my attempt to improve
+ this.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763985
+
+2016-03-21 05:09:10 -0400 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/alsa/gstalsa.c:
+ * ext/alsa/gstalsa.h:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsa: factor out alsa_detect_channels_mapping()
+ This code was duplicated in alsasrc and alsasink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763985
+
+2016-03-21 05:06:18 -0400 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/alsa/gstalsa.h:
+ alsa: coding style fix
+ Was using tabs instead of spaces.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763985
+
+2016-04-12 16:34:00 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ fdmemory, rtpbasedepayload: Ran gst-indent
+ https://bugzilla.gnome.org/show_bug.cgi?id=764948
+
+2016-04-12 16:25:12 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Rename misleading variable is_parser_converter into is_parser
+ In that place, the variable isn't checking whether the element is a
+ converter, only if it is a parser.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764948
+
+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
+
+2016-04-06 17:57:28 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/gstlibscpp.cc:
+ tests: libscpp: test RTP/RTCP buffer init macros with C++ compiler
+
+2016-04-06 21:03:19 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Don't complain when stream-start is the first event.
+ When blocking the subtitle pad, it's expected that stream-start
+ is the first event, and that it can precede caps arriving on the
+ peer pad - in fact the caps can only have arrived on the peer
+ pad when it was pre-primed with sticky events previously.
+ Instead, just pass the stream-start and don't block, because
+ stream-start is sticky anyway.
+
+2016-04-06 21:00:10 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: WebVTT Cue identifiers are optional
+ Don't require a cue identifier preceding the time range line
+ when parsing WebVTT. We could also store the CueID, but it's
+ not using anywhere, so just ignore it for now.
+
+2016-04-05 14:26:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstaudio.def:
+ win32: Add new libgstaudio symbols
+
+2016-04-01 12:25:14 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.h:
+ libs: audio: split allocation query caps and pad caps
+ Since the allocation query caps contains memory size and the pad's caps
+ contains the display size, an audio encoder or decoder might need to allocate
+ a different buffer size than the size negotiated in the caps.
+ This patch splits this logic distinction for audiodecoder and audioencoder.
+ Thus the user, if needs a different allocation caps, should set it through
+ gst_audio_{encoder,decoder}_set_allocation_cap() before calling the negotiate()
+ vmethod. Otherwise the allocation_caps will be the same as the caps in the
+ src pad.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764421
+
+2016-03-31 15:31:31 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/gstvideoutils.c:
+ * gst-libs/gst/video/gstvideoutils.h:
+ libs: video: split allocation query caos and pad caps
+ Since the allocation query caps contains memory size and the pad's caps
+ contains the display size, a video encoder or decoder might need to allocate
+ a different frame size than the size negotiated in the caps.
+ This patch splits this logic distinction for videodecoder and videoencoder.
+ The user if needs a different allocation caps, should set the allocation_caps
+ in the GstVideoCodecState before calling negotiate() vmethod. Otherwise the
+ allocation_caps will be the same as the caps set in the src pad.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764421
+
+2016-04-04 16:39:21 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: fix gtk-doc comment format
+
+2016-04-02 10:37:55 +0200 Mikhail Fludkov <misha@pexip.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ * tests/check/libs/rtpbasedepayload.c:
+ rtpbasedepayload: look at ssrc before sequence numbers
+ Doing so prevents us dropping buffers in the rare, but possible, situations,
+ when the stream changes SSRC and new sequence numbers does not differ
+ much from the last sequence number from previous SSRC. For example:
+ ssrc - 0xaaaa 101,102,103,104 ssrc - 0xbbbb 102, 103, 104, 105...
+ In the scenario above we don't want to drop the first 3 packets of
+ 0xbbbb stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764459
+
+2016-04-03 11:40:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Don't fill up the segment with duplicate buffers if drop_only==TRUE
+
+2016-04-03 11:38:28 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Remove dead code
+ We never get into this code path at all if drop_only==TRUE.
+
+2016-03-29 17:19:41 +0200 Frédéric Bertolus <frederic.bertolus@parrot.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: avoid useless buffer copy in drop-only mode
+ Make writable the buffer before pushing it lead to a buffer copy. It's
+ because a reference is keep for the previous buffer.
+ The previous buffer reference is only need to duplicate the buffer. In
+ drop-only mode, the previous buffer is release just after pushing the
+ buffer so a copy is done but it's useless.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764319
+
+2016-04-02 15:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ video: fix example code in gst_video_frame_map() docs
+ GST_VIDEO_FRAME_PLANE_PSTRIDE() does not exist.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764414
+
+2016-04-02 10:09:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ discoverer: copy over result and seekable fields when copying a discoverer info
+ The function gst_discoverer_info_copy doesn't copy the data members seekable
+ and result of the source GstDiscovererInfo.
+ In the case of copying a GstDiscovererInfo for later use, the seekbale will be
+ undefined, which in practice usually will be false, even though the seekable of
+ the original GstDiscovererInfo is true.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762710
+
+2016-03-31 13:32:32 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/video-format.h:
+ video-format: Fix macro documentation
+ The parameter type was wrongly documenting that a GstVideoInfo structure
+ pointer was needed, while it needs a GstVideoFormatInfo structure
+ pointer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764414
+
+2016-03-26 20:53:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/subparse.c:
+ * tests/check/libs/rtp.c:
+ test: fix indentation
+
+2016-03-26 20:52:16 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtp: rtcpbuffer: fix indentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=761944
+
+2016-03-26 20:50:31 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtp: rtpcbuffer: fix Since markers
+ https://bugzilla.gnome.org/show_bug.cgi?id=761944
+
+2016-03-30 11:16:49 +1100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: disable neon on arm64
+ Fix the build on arm64 by using HAVE_ARM_NEON instead of __ARM_NEON__.
+
+2016-03-29 22:16:38 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: Add more parsing guards
+ Insert extra checks for the validity of the incoming
+ data when parsing subrip/webvtt content and debug log
+ output for invalid content.
+ Should fix Coverity warnings.
+
+2016-03-29 10:23:08 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: add missing break between formats
+ A break is missing at the end of case GST_SUB_PARSE_FORMAT_LRC or it will
+ fallthrough to WebVTT. This fixes commit fd2a14144a7a.
+
+2016-03-29 12:11:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ audio-resampler: Use _mm_set_epi64x(0, x) instead of _mm_cvtsi64_si128(x) in more places
+
+2016-03-29 11:25:15 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * win32/common/video-enumtypes.c:
+ win32: Update exports for new video formats
+ Update win32 exports for P010_10BE and P010_10LE
+ video formats.
+
+2016-03-29 11:16:42 +0300 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ video: add P010 format support
+ P010 is a YUV420 format with an interleaved U-V plane and 2-bytes per
+ component with the the color value stored in the 10 most significant
+ bits.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761607
+ ---
+ Changes since v2:
+ - Set bits=16 in DPTH10_10_10_HI
+ Changes since v1:
+ - Fixed x-offset calculation in uv.
+ - Added 6-bit shifts to FormatInfo.
+
+2016-03-29 10:15:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ resampler: Use _mm_set_epi64x(0, x) instead of _mm_cvtsi64_si128(x)
+ The latter is only available on x86-64 for some reason.
+
+2016-03-29 08:21:54 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Fix distcheck
+ Don't forget to dist the needed files (which don't need to be installed)
+
+2016-03-28 15:37:36 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: estimate memory usage in auto mode
+ Estimate the memory usage and use this to decide between full or
+ interpolated filter.
+
+2016-03-28 12:51:26 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/README:
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: remove last ORC remains
+
+2016-03-16 12:55:56 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: small optimizations
+
+2016-03-04 17:15:44 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ audio-resampler: improve non-interleaved flags
+ Make it possible to have different interleaving on input and output
+ because we can quite trivially do that.
+
+2016-03-02 11:40:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: unroll some more loops
+ Unroll some loops.
+
+2016-03-01 16:31:18 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ audio-resampler: keep precision
+ Transpose and add before applying the cubic interpolation to avoid
+ overflows when using full precision.
+
+2016-03-01 16:26:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: small cleanups
+
+2016-02-25 15:38:46 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: optimize no resampling
+ Switch to the faster nearest resample method when are doing no rate
+ conversion.
+
+2016-02-25 14:09:44 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ audio-resampler: add VARIABLE_RATE flag
+ Add a VARIABLE rate flag that selects an interpolating filter.
+ Move some function setup code in the _new function.
+
+2016-02-23 04:46:55 -0500 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ audio-resampler: more neon optimizations
+
+2016-02-24 12:57:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ audio-resampler: avoid overflow in cubic interpolation
+ Shift out an extra bit to have some more headroom when doing cubic
+ interpolation.
+
+2016-02-24 12:56:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: overread only 8 taps
+ We only need 8 taps of zeroes as headroom for the SIMD optimized
+ functions.
+
+2016-02-24 12:55:28 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ audio-converter: use helper to check intermediate format
+
+2016-02-23 15:37:37 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: fix phase
+
+2016-02-22 11:16:28 -0500 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ audio-resampler: fix neon assembler
+
+2016-02-22 13:19:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: avoid some format conversion
+ Store the filter in the desired sample format so that we can simply do a
+ linear or cubic interpolation to get the new filter instead of having to
+ go through gdouble and then convert.
+
+2016-02-22 03:28:21 -0500 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ audio-resampler: fix neon linear float interpolation
+
+2016-02-19 16:39:43 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: reorder filter coefficients for more speed
+ Reorder the filter coefficients to make it easier to use SIMD for
+ interpolation.
+ Fix orc flags a little.
+ Add specialized nearest resampling function.
+
+2016-02-19 10:40:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: remove stereo optimizations
+ The stereo optimizations don't give enough benefit.
+ Rename none to full to make it clear that we use a full filter instead
+ of an interpolated one
+
+2016-02-18 12:48:45 -0500 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ audio-resample: remove neon double stubs
+ NEON does not have double types.
+
+2016-02-18 12:38:49 -0500 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ audio-resampler: add more neon optimizations
+
+2016-02-18 11:05:18 -0500 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ audio-resampler: add more neon optimizations
+
+2016-02-17 11:20:06 -0500 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-neon.h:
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: add neon optimizations
+ Unroll some more loops in the fallback code that seems to work fine
+ for ARM.
+ Add some simple ARM optimizations taken from speex.
+
+2016-02-17 13:12:31 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: give better hints about the precision
+ Give better hints to the compiler about the precision we expect from
+ the multiplications.
+
+2016-02-17 12:05:58 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resample: small optimizations
+ Remove some inline functions that are called in the slow path.
+ Unroll C fallback functions a little.
+
+2016-02-16 09:18:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: Use n_phases when calculating taps offset
+ Tweak linear interpolation oversampling.
+ Clear filter cache on rate changes when using a full filter.
+
+2016-02-15 18:06:19 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ audio-resampler: improve filter construction
+ Remove some unused variables from the inner product functions.
+ Make filter coefficients by interpolating if required.
+ Rename some fields.
+ Try hard to not recalculate filters when just chaging the rate.
+ Add more proprties to audioresample.
+
+2016-02-12 10:00:22 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: avoid overflow in fraction calculation
+
+2016-02-11 19:42:31 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: increase precision
+
+2016-02-11 17:40:56 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ audio-resampler: add more optimizations
+
+2016-02-11 13:23:07 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resample: fix taps conversion
+ We do taps conversion in place so make sure we don't overwrite the
+ input with temporary data.
+ Optimize some more gint16 functions.
+
+2016-02-11 11:57:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: Improve taps memory layout
+ Rearrange the oversampled taps in memory to make it easier to use
+ SIMD instructions on them. this simplifies some sse code.
+ Add some more optimizations
+
+2016-02-10 17:28:24 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: add cubic interpolation
+
+2016-02-10 13:31:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * win32/common/libgstaudio.def:
+ audio-resampler: add more functions
+ Use some macros to generate more functions
+
+2016-02-10 12:04:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ audio-resampler: add linear interpolation method
+ Make more functions into macros.
+ Add linear interpolation of filter coefficients.
+
+2016-02-04 15:22:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/icles/Makefile.am:
+ * tests/icles/test-resample.c:
+ tests: add resample test
+
+2016-02-04 15:21:40 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ audio-resampler: add max-phase-error config
+
+2016-02-04 15:19:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: improve tap calculation
+ Return the taps from make_taps, this makes it possible to not actually
+ have to cache the taps when we want to.
+ Fix overflow in phase calculation.
+
+2016-02-02 12:06:44 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ audio-resampler: fix guint -> gint
+
+2016-02-02 11:48:16 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: improve phase error
+ Accept a phase error of maximum 10%, which turns out to be inaudible.
+
+2016-02-01 17:18:32 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: improve phase calculation
+ Also calculate the GCD with the current phase so that we can accurately
+ represent the current phase with the new resample rates.
+
+2016-01-26 22:53:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: fix history after buffer resize
+ When we resize the temp buffer, move the history in its new place.
+
+2016-01-26 16:42:16 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ * gst/audioresample/gstaudioresample.c:
+ * win32/common/libgstaudio.def:
+ audio-resampler: add reset function
+ Add a function to reset the audio-resampler.
+ Use new function in audio-converter
+ Use the new functions in gstaudioresample and fixup drain functions.
+
+2016-01-26 16:40:57 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: Small fixes
+ Fix the phase.
+ Reset the new sample buffer with 0.
+ Move samples around when we change the filter size.
+
+2016-01-26 16:38:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: Rework make_taps
+ Make it return a pointer to the generated taps. That way we can later
+ decide to actually cache it or not.
+
+2016-01-26 09:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst/audioresample/gstaudioresample.c:
+ audio-resampler: handle filter length changes
+ Update the buffer with history samples when the filter length changes
+ because of an update of the parameters or sample rates.
+
+2016-01-22 17:34:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: fix samples_avail
+ We only know the taps after we calculate them.
+
+2016-01-22 16:45:28 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: work on dynamically changing the samplerate
+ Calculate the new phase for the new sample rate.
+ Fix some docs.
+
+2016-01-22 10:28:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: small cleanups
+
+2016-01-21 10:38:17 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: add fallback to mono function
+ Remove stereo implementations. Implement fall back to mono functions
+ when the stereo function is missing.
+
+2016-01-18 12:52:41 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: add float stereo SSE function
+
+2016-01-15 12:45:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * configure.ac:
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ audio-resampler: Fix compilation of intrinsics
+ Only compile intrinsics when we are building for the selected
+ architecture.
+ Add sse4.1 optimized int32 resampler code.
+
+2016-01-15 11:43:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ audioconvert: only resample on supported formats
+
+2016-01-15 11:20:29 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst/audioresample/gstaudioresample.c:
+ audio-converter: make some optimized functions
+ Make an optimized function that just calls the resampler when possible.
+ Optimize the resampler transform_size function a little.
+
+2016-01-15 10:26:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: remove mirror function
+ We don't need to mirror the input, just assume 0 samples.
+ Always move the processed samples to the start of the buffer.
+ Add some G_LIKELY
+
+2016-01-13 17:50:38 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ audio-resampler: also enable sse when sse2 is available
+
+2016-01-13 17:44:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: optimizations
+ Improve int16 resampling by using pmaddwd
+ Use intrinsics to scale and pack int16 samples
+ Align the coefficients so that we can use aligned loads
+ Add padding to taps and samples so that we don't have to use partial
+ loads for the remainder of the loops.
+ Remove copy_n, we can reuse the plain copy function with some new
+ parameters.
+ Align and pad the sample array.
+
+2016-01-12 18:55:19 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-x86.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: make pluggable optimized functions
+ Add support for x86 specialized functions and select them at runtime.
+
+2016-01-12 10:23:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-resampler-core.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ audio-resampler: combine functions
+
+2016-01-11 16:25:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * win32/common/libgstaudio.def:
+ defs: update
+
+2016-01-05 16:06:22 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-converter.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ * gst/audioresample/gstaudioresample.c:
+ audio-converter: simplify API
+ Remove the consumed/produced output fields from the resampler and
+ converter. Let the caler specify the right number of input/output
+ samples so we can be more optimal.
+ Use just one function to update the converter configuration.
+ Simplify some things internally.
+ Make it possible to use writable input as temp space in audioconvert.
+
+2016-01-04 18:28:38 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/audio-converter.c:
+ * gst-libs/gst/audio/audio-converter.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ audio-converter: more work on resampling
+ - Fix the resampler in the audio converter
+ - fix memory leaks
+
+2015-11-13 15:32:29 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * 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-resampler-core.h:
+ * gst-libs/gst/audio/audio-resampler.c:
+ * gst-libs/gst/audio/audio-resampler.h:
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/audio/dbesi0.c:
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/arch.h:
+ * gst/audioresample/fixed_arm4.h:
+ * gst/audioresample/fixed_arm5e.h:
+ * gst/audioresample/fixed_bfin.h:
+ * gst/audioresample/fixed_debug.h:
+ * gst/audioresample/fixed_generic.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/resample_neon.h:
+ * gst/audioresample/resample_sse.h:
+ * gst/audioresample/speex_resampler.h:
+ * gst/audioresample/speex_resampler_double.c:
+ * gst/audioresample/speex_resampler_float.c:
+ * gst/audioresample/speex_resampler_int.c:
+ * gst/audioresample/speex_resampler_wrapper.h:
+ audio-converter: add resampler
+ Add a resampler to the processing chain when needed.
+ port the audio resampler to the new audioconverter library
+
+2016-03-25 01:13:54 +1100 Jan Schmidt <jan@centricular.com>
+
+ * win32/common/libgstpbutils.def:
+ * win32/common/libgstrtp.def:
+ win32: update win32 exports for new API
+
+2016-03-07 23:29:43 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ * tests/check/elements/subparse.c:
+ subparse: WebVTT parsing support
+ WebVTT is a new subtitle format for HTML5 video. In this first
+ version of the parser the cue settings are parsed but only stored in
+ the internal parser state structure. Later on these settings could be
+ part of the GstBuffer metadata.
+ https://bugzilla.gnome.org/show_bug.cgi?id=629764
+
+2016-02-26 02:58:26 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add a typefinder for WebVTT files
+
+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.
+
+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
+
+2016-03-08 19:22:18 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: avoid unnecessary gst_pad_has_current_caps() checks
+ No need to do this for each input buffer, we have the input caps
+ stored somewhere already.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763337
+
+2016-03-22 11:25:49 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/codec-utils.c:
+ * gst-libs/gst/pbutils/codec-utils.h:
+ * win32/common/libgstpbutils.def:
+ codec-utils: Add utilities for AAC and the AACHead header
+ Add utilities about the channels and sample rate for AAC.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749110
+
+2016-03-21 16:06:20 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Modify result of seekable in check_upstream_seekable function
+ In check_upstream_seekable function, it returns FALSE value even though
+ we already declare about the seekable variable. So, This patch return
+ result of seekable in check_upstream_seekable function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763975
+
+2016-03-03 16:46:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/alsa/gstalsamidisrc.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/libvisual/visual.c:
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstogmparse.c:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisenc.c:
+ * ext/vorbis/gstvorbisparse.c:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/audio/gstaudiocdsrc.c:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/encoding/gstencodebin.c:
+ * gst/encoding/gstsmartencoder.c:
+ * gst/encoding/gststreamcombiner.c:
+ * gst/encoding/gststreamsplitter.c:
+ * gst/gio/gstgiobasesink.c:
+ * gst/gio/gstgiobasesrc.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysinkconvertbin.c:
+ * gst/playback/gststreamsynchronizer.c:
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/subparse/gstssaparse.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/tcp/gstmultihandlesink.c:
+ * gst/tcp/gstsocketsrc.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/videoconvert/gstvideoconvert.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ * tests/check/elements/audiorate.c:
+ * tests/check/elements/decodebin.c:
+ * tests/check/elements/playbin-complex.c:
+ * tests/check/elements/playbin.c:
+ * tests/check/elements/videoscale.c:
+ * tests/check/libs/audiodecoder.c:
+ * tests/check/libs/audioencoder.c:
+ * tests/check/libs/baseaudiovisualizer.c:
+ * tests/check/libs/rtpbasedepayload.c:
+ * tests/check/libs/rtpbasepayload.c:
+ * tests/check/libs/videodecoder.c:
+ * tests/check/libs/videoencoder.c:
+ base: use new gst_element_class_add_static_pad_template()
+ https://bugzilla.gnome.org/show_bug.cgi?id=763075
+
+2015-10-06 17:02:03 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ * tests/check/libs/rtp.c:
+ rtcpbuffer: Add API for APP packets
+ https://bugzilla.gnome.org/show_bug.cgi?id=761944
+
+2014-07-29 15:37:12 +0200 Haakon Sporsheim <haakon@pexip.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ * tests/check/libs/rtp.c:
+ * win32/common/libgstrtp.def:
+ rtcpbuffer: Add profile-specific extension API.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761950
+
+2016-03-24 13:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.8.0 ===
+
+2016-03-24 12:19:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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
+
+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
+
+=== release 1.7.91 ===
+
+2016-03-15 12:02:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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
+
+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
+
+=== release 1.7.90 ===
+
+2016-03-01 18:14:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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
+
+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
+
+=== release 1.7.2 ===
+
+2016-02-19 11:48:30 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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
+
+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
+
+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
+
+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
+
+=== release 1.7.1 ===
+
+2015-12-24 13:59:15 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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
+
+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
+
+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
+
+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
+
+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
+
+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
+
+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
+
+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()
+
+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.
+
+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
+
+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
+
+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
+
+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.
+
+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.org/show_bug.cgi?id=754054
+
+2015-08-20 15:59:15 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst-libs/gst/tag/mklicensestables.c:
+ * tests/examples/encoding/encoding.c:
+ * tests/examples/playback/playback-test.c:
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/icles/stress-playbin.c:
+ * tests/icles/test-effect-switch.c:
+ * tools/gst-device-monitor.c:
+ * tools/gst-discoverer.c:
+ * tools/gst-play.c:
+ gstreamer: base: Fix memory leaks when context parse fails.
+ When g_option_context_parse fails, context and error variables are not getting free'd
+ which results in memory leaks. Free'ing the same.
+ And replacing g_error_free with g_clear_error, which checks if the error being passed
+ is not NULL and sets the variable to NULL on free'ing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753852
+
+2015-06-24 23:55:35 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: Fix special case
+ Allows to run such a command line :
+ gst-launch-1.0 uridecodebin uri=file:///home/meh/Music/sthg.mp4 ! \
+ encodebin profile-string="audio/x-wav|1" ! filesink location=sthg.wav
+ Previously the code failed because wavenc is considered as a muxer.
+ We still want encodebin to audio/x-wav as an AudioEncodingProfile,
+ so this simple fix allows that.
+ Ability to mux raw streams in containers such as matroskamux
+ is a different issue.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751470
+
+2015-09-29 10:12:28 +0530 Rajat Verma <rajat.verma@st.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: free hidden groups at time of switching groups
+ hidden groups should be freed at time of switching groups to avoid memory use
+ from balloning up.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755770
+
+2015-10-02 10:07:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstpbutils.def:
+ win32: Update exports for new audiovisualizer symbols
+
+2015-10-02 15:04:34 +1000 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/baseaudiovisualizer.c:
+ tests: Add baseaudiovisualizer test, moved from -bad
+
+2015-10-02 15:05:26 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Don't fixate framerate if downstream didn't provide one
+ intersection with a downstream that accepts any video/x-raw caps
+ with no further detail won't create a framerate field. If it's
+ not in the caps, don't fixate it, just set it to 30/1
+
+2015-10-01 21:53:20 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * 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/inspect/plugin-alsa.xml:
+ docs: add alsamidisrc to docs
+
+2015-10-01 21:43:21 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * ext/alsa/Makefile.am:
+ * ext/alsa/gstalsamidisrc.c:
+ * ext/alsa/gstalsamidisrc.h:
+ * ext/alsa/gstalsaplugin.c:
+ midi: add an ALSA MIDI sequencer source
+ The alsamidisrc element allows to get input event from ALSA MIDI
+ sequencer devices, and possibly convert them to sound using some
+ downstream element like fluiddec.
+ Fixes #738687
+
+2015-10-01 15:27:55 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst-libs/gst/pbutils/gstaudiovisualizer.c:
+ visual: make private all variable subclasses don't need
+ Subclasses don't need access to all variables. Making them private.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-10-01 11:55:59 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * ext/libvisual/Makefile.am:
+ * ext/libvisual/visual.h:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/gstaudiovisualizer.c:
+ * gst-libs/gst/pbutils/gstaudiovisualizer.h:
+ visual: merge audiovisalizer base classes
+ Move the audiovisualizer base class to pbutils, so it can be used by plugins
+ from other modules
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-10-01 12:48:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: minor clean-up
+ Remove unnecessary brackets from IS_MPEGTS_HEADER macro.
+
+2015-10-01 12:32:33 +0100 Pankaj Darak <pankajdarak@gmail.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: mpeg-ts detection improvement
+ Allow AFC to be 0 for null pid packets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=726117
+
+2015-09-30 18:18:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/subparse.c:
+ tests: subparse: add unit test for closing tag detection
+ </ i> should be handled like </i>
+ https://bugzilla.gnome.org/show_bug.cgi?id=755875
+
+2015-09-30 18:17:13 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: detect closing tags even if there's a space after the slash
+ </ i> should be handled like </i>
+ https://bugzilla.gnome.org/show_bug.cgi?id=755875
+
+2015-09-23 11:59:22 -0400 Perry Hung <perry@leaflabs.com>
+
+ * gst-libs/gst/app/Makefile.am:
+ app: pass PKG_CONFIG_PATH for gir files for libgstapp as well
+ gir include search directories should respect PKG_CONFIG_PATH,
+ just like we do everywhere else. Makes g-i pick up the right
+ paths when using ./configure --with-pkg-config-path=
+ https://bugzilla.gnome.org/show_bug.cgi?id=755494
+
+2015-09-25 23:51:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.6.0 ===
+
+2015-09-25 23:15:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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-libs/gst/video/video-orc-dist.c:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 1.6.0
+
+2015-09-25 22:50:51 +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
+
+2015-09-24 18:06:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ video-dither: Use saturated add when adding ordered dither for > 8 bit targets
+ Otherwise our 16 bit integers are going to overflow in intermediate
+ calculations, causing video to become mostly black.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755392
+
+2015-09-24 11:33:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: Fix gst_video_frame_copy() for formats with pstride==0
+ v210, UYVP and IYU1 are complex formats for which pixel stride does not really
+ have a meaning. If we copy width*pstride bytes per line, it's not going to do
+ the right thing. As a fallback, copy stride bytes per line. This might copy
+ uninitialized bytes at the end of each line, but at least copies the frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755392
+
+2015-09-10 15:08:35 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ fdmemory: remove 'allow-none' annotation in gst_fd_allocator_alloc() doc
+ gst_fd_allocator_alloc() ensure that passed allocator is a fd memory
+ allocator, so that we can't pass NULL allocator.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754833
+
+2015-09-10 15:08:35 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ dmabuf: remove 'allow-none' annotation in gst_dmabuf_allocator_alloc() doc
+ gst_dmabuf_allocator_alloc() ensure that passed allocator is a DMABuf
+ allocator, so that we can't pass NULL allocator.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754833
+
+=== release 1.5.91 ===
+
+2015-09-18 19:20:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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/config.h:
+ Release 1.5.91
+
+2015-09-18 19:19:16 +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
+
+2015-09-18 17:48:49 +0200 Christophe Fergeau <cfergeau@redhat.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ appsink: Fix 'steaming' typo in API doc
+ There are several occurrences of 'steaming' where 'streaming' was meant.
+
+2015-09-18 11:49:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-09-18 10:59:46 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tests/examples/playback/playback-test.c:
+ playback-test: avoid critical on exit
+ Only free vis_entries array when not null on exit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755201
+
+2015-09-18 09:48:18 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: fix typo in debug log message
+ https://bugzilla.gnome.org/show_bug.cgi?id=755198
+
+2015-09-17 14:27:33 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst/tcp/gstmultisocketsink.c:
+ audiosink, multisocketsink: Fix error leak during failures
+ https://bugzilla.gnome.org/show_bug.cgi?id=755143
+
+2015-09-16 19:53:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Use the correct caps name for MS Smooth Streaming manifests
+ Thanks to John Chang <r97922153@gmail.com> for reporting.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755098
+
+2015-09-15 15:39:11 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: remove check for number of channels
+ opus decoder can convert from different number of channels, no
+ need to check, just let it negotiate and create a new decoder if
+ needed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755059
+
+2015-09-15 15:26:44 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: minor docs fix
+
+2015-09-11 23:36:47 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/gstvideopool.c:
+ videopool: ensure allocation alignment is consistent with video alignment requirements
+ Make sure GstAllocationParams alignment is not less than
+ any alignment requirement specified via GstVideoAlignment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754120
+
+2015-09-14 09:36:20 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix error leak when context creation fails
+ When context creation fails, error is getting leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754973
+
+2015-09-11 11:22:35 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: improve deprecated properties docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=754819
+
+2015-09-11 11:11:09 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: do not throw g_warning when getting deprecated properties
+ https://bugzilla.gnome.org/show_bug.cgi?id=754819
+
+2015-09-11 23:28:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playback: Add POINTER_TO_ULONG() macro for consistency
+
+2015-09-11 23:48:05 +0900 Kouhei Sutou <kou@clear-code.com>
+
+ * gst/playback/gstplaybin2.c:
+ playback: fix build error for 64bit Windows build by MinGW
+ Casting to gpointer from gulong generates the following warning with
+ 64bit Windows target MinGW:
+ gstplaybin2.c: In function 'pad_added_cb':
+ gstplaybin2.c:3476:7: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
+ (gpointer) group_id_probe_handler);
+ ^
+ cc1: all warnings being treated as errors
+ We should cast to guintptr from gulong before we cast to gpointer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754755
+
+2015-09-09 19:00:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: add missing alpha flag for some formats
+ Some formats didn't have the alpha flag marked, use the correct
+ macro so they get it right.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754808
+
+2015-09-09 12:33:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Always take the mutex before flushing the queue
+ Otherwise the application might push new buffers into the queue while we're
+ flushing, potentially causing the GQueue data structure to become inconsistent
+ and causing crashes soon after.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754597
+
+2015-09-08 01:35:19 +0530 Vikram Fugro <vikram.fugro@gmail.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ * tests/check/elements/appsrc.c:
+ appsrc: retain the latest caps in queue when flushing
+ - Retain the latest caps in the internal queue, when
+ flushing.
+ - Add a unit test case for the same.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754597
+
+2015-09-07 00:19:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-format.h:
+ video: improve GstVideoFormatUnpack docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=754650
+
+2015-09-06 18:17:15 +0900 Kouhei Sutou <kou@clear-code.com>
+
+ * gst-libs/gst/video/video-dither.c:
+ libs: Fix build error on MinGW where "%ll" is not available
+ "ll" isn't available on MinGW. We can use G_GINT64_MODIFIER for portable
+ 64bit size data modifier.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754630
+
+2015-08-31 10:46:43 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst-libs/gst/Makefile.am:
+ libs: build rtp after audio
+ The dependency setup does not seem to work for all systems,
+ causing the build to fail with:
+ gstrtpbaseaudiopayload.c:65:0:
+ fatal error: gst/audio/audio-enumtypes.h: No such file or directory
+ My setup:
+ gcc (Ubuntu 4.8.4-2ubuntu1~14.04) 4.8.4
+ autoconf (GNU Autoconf) 2.69
+ automake (GNU automake) 1.14.1
+ libtool (GNU libtool) 2.4.2
+ https://bugzilla.gnome.org/show_bug.cgi?id=754344
+
+2015-08-31 10:49:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/Makefile.am:
+ libs: rtp is no longer an independent subdir
+ https://bugzilla.gnome.org/show_bug.cgi?id=754344
+
+2015-09-03 17:55:10 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: fix gamma-decode option
+ We need to use the enum to configure the option now.
+
+2015-09-03 08:58:08 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: FIX crash if padding greater than video size
+ Skipping rendering of textimage if overlay is completely
+ outside video frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754429
+
+2015-08-31 11:09:09 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: remove duplicate get_size vfunc assignment
+
+2015-08-29 21:38:52 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst-libs/gst/allocators/allocators.h:
+ allocators: include gstfdmemory.h in the main library header, allocators.h
+
+2015-08-29 10:44:28 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: fix misleading error message debug detail
+ https://bugzilla.gnome.org/show_bug.cgi?id=754260
+
+2015-08-28 09:36:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: implement accept-caps handler
+ The default one will just go through the internal elements which might
+ just be identity when it is in passthrough which will lead to the query
+ being handled by the downstream sink, ignoring all that playsinkconvertbin
+ could actually handle and convert.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754235
+
+2015-08-27 23:08:51 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstaudiobasesink.h:
+ audiobasesink: Fix incorrect/missing custom slaving method documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=754199
+
+2015-08-19 21:19:05 +0900 Eunhae Choi <eunhae1.choi@samsung.com>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: use g_clear_error instead of g_error_free
+ To avoid invalid pointer accees the err pointer should be set to NULL.
+ By using g_clear_error() it calls free and clear the pointer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753817
+
+=== release 1.5.90 ===
+
+2015-08-19 13:10:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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-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:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ Release 1.5.90
+
+2015-08-19 12:39:17 +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
+
+2015-08-19 11:23:09 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/hu.po:
+ * po/nb.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/uk.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-08-19 08:37:46 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tools/gst-discoverer.c:
+ tools: discoverer: When info is NULL just print error and return
+ In case discover_uri returns NULL info, passing the info to discoverer APIs
+ result in critical assertion errors. Hence instead of passing NULL info along,
+ print the error and return.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753701
+
+2015-08-18 18:47:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ Revert "decodebin: Handle the preroll multi-queue size"
+ This reverts commit 5c8ef0ea05123506dfc35c70c8b165bca7435dad.
+
+2015-08-18 18:47:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ Revert "decodebin: Store extra_buffer_required per group, not globally"
+ This reverts commit 1ea81114ea6bd48b581f19002018680933aa7a12.
+
+2015-08-18 18:47:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ Revert "decodebin: If extra buffers are going to be required, we're still prerolling"
+ This reverts commit a3b24f0241bd55a005a072ba8ddcd53e0fdbf827.
+
+2015-08-18 16:28:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ video(en|de)coder: Return TRUE when we consumed a tag event without creating a new event
+ Fixes spurious flow errors that especially break gst-validate.
+
+2015-08-18 16:01:28 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: If there are no tags, don't try to do event handling on a NULL event
+ Fixes some crashes.
+
+2015-08-18 15:58:57 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: If there are no tags, don't try to do event handling on a NULL event
+ Fixes some crashes.
+
+2015-08-18 13:50:17 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * tools/gst-play.c:
+ tools: gst-play: Use g_build_filename instead of g_strconcat
+ When running gst-play against a directory name, and suffix the path with a
+ directory separator (e.g. tab completion), gst-play was printing two directory
+ separators in a row. g_build_filename fixes this, and additionally allows for
+ both '/' and '\' as separators on Windows.
+
+2015-08-18 15:16:25 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: If extra buffers are going to be required, we're still prerolling
+
+2015-08-18 15:01:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Store extra_buffer_required per group, not globally
+ It's only relevant for each group, and by storing it in the group
+ we have locking and everything else like for the other buffering-related
+ variables. Locking looks a bit fishy still, but it was like that for a long
+ time already so shouldn't be worse than before.
+
+2015-07-30 10:33:25 +0900 Myoungsun Lee <ohmygod0327@gmail.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Handle the preroll multi-queue size
+ Overview:
+ There are some of interleaved streams which has long-term location of audio data.
+ It mean the audio data is located far away more than multiqueue size.
+ In this case, because of multiqueue overrun, the pipeline is stopped.
+ To prevent hanging-like state, the decodebin needs to handle the queue size.
+ Caused:
+ The multiqueue size is not enough, the pipeline will stay being stalled status
+ and decodebin cannot complete to build decode chain.
+ In this issue file, decodebin did not receive no_more_pads signal or audio data yet.
+ Steps to Reproduce:
+ play the high-resolution(4K file) files or some streaming media(push mode).
+ Actual Results:
+ There is no audio or subtitle.
+ We can see only video or infinite loading.
+ Resolution:
+ Decodebin detect this problem, and add extra buffer size to multiqueue.
+ The multiqueue is larger than before, the next data can be pushed the downstream element.
+ Additional Information:
+ The max-preroll extra buffer size is set 8MB.
+ We can use total pre-roll buffer 10MB.
+ Only first overrun callback can handle multiqueue size.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733235
+
+2015-08-18 12:29:29 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: fix tag handling
+ Merge upstream tags with encoder tags and update whenever
+ any of those changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-18 11:45:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: fix tag handling
+ Merge upstream tags with encoder tags and update whenever
+ any of those changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-18 12:56:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add typefinder for TTML+XML
+ Used in DASH among other things, as SMPTE Timed Text.
+
+2015-08-18 09:06:39 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ pbutils: discoverer: Set GError when NULL info is being returned.
+ When discovering the URI, if info is NULL, then instead of just returning NULL,
+ set the GError, so the error can be printed and notified.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753701
+
+2015-08-16 07:18:34 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/opus/gstopusenc.c:
+ audioencoders: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+ Elements: faac, gsmenc, opusenc, sbcenc, voamrwbenc, adpcmenc, sirenenc
+
+2015-08-17 11:18:25 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tools/gst-discoverer.c:
+ discoverer: free context and error during failures
+ When g_option_context_parse or gst_discoverer_new fails, then there will
+ be memory leaks for ctx and err variables. Free'ing the same.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753701
+
+2015-08-16 18:28:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: try harder to avoid sending unnecessary tag updates
+
+2015-08-16 17:55:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: fix tag handling
+ Before we just merged everything in pretty much random ways
+ ad-hoc instead of keeping state properly. In 0.10 that was
+ how it worked, but in 1.x the tag events sent should always
+ reflect the latest state and replace any previous tags.
+ So save the upstream (stream) tags, and save the tags set
+ by the decoder subclass with merge mode, and then update
+ the merged tags whenever either of those two changes.
+ This slightly changes the behaviour of gst_video_decoder_merge_tags()
+ in case it is called multiple times, since now any call replaces
+ the previously-set tags. However, it leads to much more predictable
+ outcomes, and also we are not aware of any subclass which sets this
+ multiple times and expects all the tags set to be merged.
+ If more complex tag merging scenarios are required, we'll have
+ to add a new vfunc for that or the subclass has to intercept
+ the upstream tags itself and send merged tags itself.
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-14 17:59:29 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: add unit test for tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-14 17:44:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: fix tag handling
+ Before we just merged everything in pretty much random ways
+ ad-hoc instead of keeping state properly. In 0.10 that was
+ how it worked, but in 1.x the tag events sent should always
+ reflect the latest state and replace any previous tags.
+ So save the upstream (stream) tags, and save the tags set
+ by the decoder subclass with merge mode, and then update
+ the merged tags whenever either of those two changes.
+ This slightly changes the behaviour of gst_audio_decoder_merge_tags()
+ in case it is called multiple times, since now any call replaces
+ the previously-set tags. However, it leads to much more predictable
+ outcomes, and also we are not aware of any subclass which sets this
+ multiple times and expects all the tags set to be merged.
+ If more complex tag merging scenarios are required, we'll have
+ to add a new vfunc for that or the subclass has to intercept
+ the upstream tags itself and send merged tags itself.
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-15 22:23:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vorbis/gstvorbisenc.c:
+ vorbisenc: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+
+2015-08-16 12:20:51 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vorbis/gstvorbisenc.c:
+ vorbisenc: use more accurate sink pad template caps
+ Removes the need for custom caps query handling and makes it more
+ correct from the beginning on the template. It is a bit uglier
+ to read because there is 1 entry per channel but makes code easier
+ to maintain.
+
+2015-08-15 22:22:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+
+2015-08-16 08:12:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.h:
+ audioencoder: add src and sink query methods
+ Allows subclasses to do their own handling of GstQuery and still
+ chain up to the parent class to handle the ones that they don't want
+ to handle
+
+2015-08-16 12:53:02 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Fix list iteration
+ We were using the wrong variable ...
+ CID #1316477
+
+2015-08-15 12:58:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/opus/gstopusdec.c:
+ audiodecoders: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+ Elements: dtsdec, faad, gsmdec, mpg123audiodec, opusdec,
+ sbcdec, adpcmdec, sirendec
+
+2015-05-04 11:19:28 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Handle flushing with multiple decode groups
+ When an upstream element wants to flush downstream, we need to take
+ all chains/groups into consideration.
+ To that effect, when a FLUSH_START event is seen, after having it
+ sent downstream we mark all those chains/groups as "drained" (as if
+ they had seen a EOS event on the endpads).
+ When a FLUSH_STOP event is received, we check if we need to switch groups.
+ This is done by checking if there are next groups. If so, we will switch
+ over to the latest next_group. The actual switch will be done when
+ that group is blocked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=606382
+
+2015-04-29 15:56:39 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Forward event/queries for unlinked groups
+ When upstream events/queries reach sinkpads of unlinked groups (i.e.
+ no longer linked to the upstream demuxer), this patch attempts to find
+ the linked group and forward it upstream of that group.
+ This is done by adding upstream event/query probes on new group sinkpads
+ and then:
+ * Checking if the pad is linked or not (has a peer or not)
+ * If there is a peer, just let the event/query follow through normally
+ * If there is no peer, we find a pad to which to proxy it and return
+ GST_PROBE_HANDLED if it succeeded (allowing the event/query to be properly
+ returned to the initial called)
+ Note that this is definitely not thread-safe for the time being
+ https://bugzilla.gnome.org/show_bug.cgi?id=606382
+
+2015-08-15 08:18:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ * win32/common/libgstaudio.def:
+ Revert "audiodecoder: expose default query handling function"
+ Apparently I forgot how gobject works, there is no need to expose
+ it directly as one can call it from the parent_class pointer
+ This reverts commit 8a64592481dab985ca520a5b1cb394a609275c60.
+
+2015-08-15 08:14:00 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * win32/common/libgstvideo.def:
+ Revert "videodecoder: expose default query handling function"
+ Apparently I forgot how gobject works, there is no need to expose
+ it directly as one can call it from the parent_class pointer
+ This reverts commit ea9b6a7e3c4eea512650adf530b7f1acb0eccd84.
+
+2015-08-15 07:41:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vorbis/gstvorbisdec.c:
+ vorbisdec: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+
+2015-08-15 07:40:55 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/theora/gsttheoradec.c:
+ theoradec: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+
+2015-08-15 07:31:54 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ * win32/common/libgstaudio.def:
+ audiodecoder: add option to use default pad accept-caps handling
+ Add gst_audio_decoder_set_use_default_pad_acceptcaps() to allow
+ subclasses to make videodecoder use the default pad acceptcaps
+ handling instead of resorting to the caps query that is, usually,
+ less efficient and unecessary
+ API: gst_audio_decoder_set_use_default_pad_acceptcaps
+
+2015-08-15 07:20:25 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * win32/common/libgstvideo.def:
+ videodecoder: add option to use default pad accept-caps handling
+ Add gst_video_decoder_set_use_default_pad_acceptcaps() to allow
+ subclasses to make videodecoder use the default pad acceptcaps
+ handling instead of resorting to the caps query that is, usually,
+ less efficient and unecessary
+ API: gst_video_decoder_set_use_default_pad_acceptcaps
+
+2015-08-15 23:33:14 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasedepayload: Make stats creation threadsafe, fix a CRITICAL
+ Use the object lock to protect the internal segment when updating
+ against access from getting the stats property.
+ Fix a critical in gst-inspect or when retrieving the stats
+ before any segment has arrived by checking whether the
+ segment has been initted..
+
+2015-08-12 03:00:15 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Make the H.264 typefind a tiny bit more lenient.
+ When we see prefix NALs before a Subset SPS has been spotted,
+ it might just be because the stream was truncated at the
+ start, so don't count those as either 'bad' or 'good' packets.
+
+2015-08-14 18:43:03 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: unref the preroll buffer and cleanup the segments on stop()
+ Just for consistency. No need to keep data around.
+
+2015-08-14 18:35:22 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: do not update preroll_caps unless the sink is prerolling
+ Just for consistency with the preroll_segment
+
+2015-08-14 18:06:03 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * tests/check/elements/appsink.c:
+ tests/appsink: add test to ensure that the segment returned by pull-preroll/sample is correct
+ https://bugzilla.gnome.org/show_bug.cgi?id=751147
+
+2015-06-18 12:30:24 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: put the correct segment in the preroll sample
+ last_segment is only being updated in dequeue_buffer(),
+ which is only called from _pull_sample(). _pull_preroll()
+ simply re-uses an old or dummy segment while the actual
+ one sits and waits in the queue.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751147
+
+2015-08-14 08:59:51 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * win32/common/libgstvideo.def:
+ videodecoder: expose default query handling function
+ Subclasses can use it to select what queries they want to handle
+ and forward the rest to the default handling function.
+ API: gst_video_decoder_sink_query_default
+ https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 08:58:58 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ * win32/common/libgstaudio.def:
+ audiodecoder: expose default query handling function
+ Subclasses can use it to select what queries they want to handle
+ and forward the rest to the default handling function.
+ API: gst_audio_decoder_sink_query_default
+ https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 11:11:10 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/generic/states.c:
+ check: Rename states unit test
+ Makes it easier to differentiate from other modules states unit test
+
+2015-08-14 05:48:31 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: remove accept-caps handling
+ Just let the internal element of the bin do it instead of forcing a
+ caps query to do it.
+
+2015-08-13 13:52:17 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: fixate the pixel-aspect-ratio
+ If the pixel-aspect-ratio is not fixed, try to get it as close
+ to 1/1 as possible
+ https://bugzilla.gnome.org/show_bug.cgi?id=748635
+
+2015-08-11 15:09:10 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: mention videorate is often needed in docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=748877
+
+2015-08-11 14:10:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/Makefile.am:
+ rtp: Depend on the audio library
+
+2015-07-01 16:25:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c:
+ rtpbaseaudiopayload: Copy metadata in the (de)payloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without
+ tags or with only the audio tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-10 22:03:48 +0200 Joan Pau Beltran <joanpau.beltran@socib.cat>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: add support for bayer formats
+ Since the videorate element just duplicates or drops frames
+ to achieve the desired framerate, it can accept video/x-bayer media
+ (in any format), which are not present in the current caps.
+ Just add "video/x-bayer(ANY);" to the caps of the static pad template
+ (fixing line style to pass the indent commit hook).
+ https://bugzilla.gnome.org/show_bug.cgi?id=753483
+
+2015-08-05 15:32:54 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ basedepayloader: Don't re-timestamp with running-time
+ There was a confusion, six depayloaders where passing through the
+ timestamp while the base class was re-timestamping to running
+ time. This inconstancy has been unnoticed has in most use cases
+ the incoming segment is [0, inifnity] in which case timestamps are
+ the same as running time. With DTS/PTS shifting added (to avoid
+ negative values) and pcapparse sending a different segment this
+ started being an issue.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753037
+
+2015-08-10 09:49:19 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ videoencoder: remove empty line to make g-i-scanner happy
+ gstvideoencoder.h:228: Warning: GstVideo: "@transform_meta"
+ parameter unexpected at this location:
+ * @transform_meta: Optional. Transform the metadata on ...
+
+2015-08-10 08:17:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: documentation cleanup
+ Remove some whitespace and break lines longer than 80 columns
+
+2015-08-10 00:21:42 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: add test to make sure gap is pushed before segment
+ https://bugzilla.gnome.org/show_bug.cgi?id=753360
+
+2015-08-09 23:23:05 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * tests/check/libs/videodecoder.c:
+ videodecoder: push pending events before gap
+ Push all pending events before pushing the gap. This ensures the
+ segment is pushed before the gap so it can be properly translated
+ to the running time
+ Includes unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753360
+
+2015-07-30 16:39:03 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Set chain pointers to NULL
+ Otherwise, they will refer to freed memory
+ https://bugzilla.gnome.org/show_bug.cgi?id=753078
+
+2015-07-31 13:31:56 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: fix deadend_details string leak
+ deadend_details need not be returned when the pad is not a deadend.
+ Hence checking if res value is TRUE and clearing the string instead of
+ passing it on
+ https://bugzilla.gnome.org/show_bug.cgi?id=753088
+
+2015-08-04 14:41:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Don't set DTS on buffer
+ DTS is for encoded data and have no meaning for raw. It better to not
+ set it, as it's confusing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752791
+
+2015-07-30 18:43:19 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Return FLUSHING if pad if flushing
+ If the initial seek fails because the pad is
+ flushing, then return GST_FLOW_FLUSHING instead
+ of an error.
+
+2015-07-30 15:16:57 +0100 Brian Peters <brianfpeters@gmail.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: avoid accessing NULL buffer even more
+ Previous commit was incompletely applied.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753001
+
+2015-07-30 14:30:44 +0100 Brian Peters <brianfpeters@gmail.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: buffer: don't access NULL buffer pointer
+ unmap will set rtpbuffer->buffer to NULL, so we need to
+ save the pointer to access it while the RTP buffer is
+ unmapped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753001
+
+2015-07-30 12:50:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasedepayload: fix leaks in error code paths
+ This was introduced when reshuffling the buffer unmaps
+ in commit bc14cdf529e21356ea7b2c8f34614958a91f7260
+ rtp: rtpbasedepayload: add process_rtp_packet() vfunc
+ Fixes make check-valgrind.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750235
+
+2015-07-28 13:57:20 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ textoverlay: Query downstream caps for checking if caps features are supported, not just accept-caps
+ accept-caps is not recursive and might stop at the next downstream element,
+ while caps queries are generally recursive. The next element might accept any
+ capsfeatures we want, but that doesn't mean that further downstream it will
+ also work.
+ Additionally for the future:
+ We should probably check if downstream *prefers* the
+ overlay meta, and only enforce usage of it if we can't handle
+ the format ourselves and thus would have to drop the overlays.
+ Otherwise we should prefer what downstream wants here.
+
+2015-07-27 18:39:13 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/opus/gstopuscommon.c:
+ opuscommon: Use GString instead of snprintf for concating
+ Safer, easier to understand, and more portable. Also, skip
+ all this if the log level is too low.
+
+2015-07-23 15:28:42 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/pango/gstbasetextoverlay.h:
+ basetextoverlay: Use the extents rectangle for positioning
+ the extents rectangle is what you need to know to properly position
+ a buffer that has been rendered in a surface of the ink rectangle
+ size. This patch make the placement on par with the placement we had
+ before without having to over allocate.
+ This patch also enable placement for vertical rendering. Note that
+ the halginement, valighment and line-alignment default are set to
+ the previous default when this property is set. This is for backward
+ compatibility, you can change the value after setting vertical render.
+ https://bugzilla.gnome.org/show_bug.cgi?id=728636
+
+2015-07-23 15:19:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Fix clipping issues
+ This patch uses the ink rectangle in order to compute the size
+ of the surface require to render. It also correctly compute the
+ transformation matrix as the ink_rect position might not be at
+ 0, 0. Additionally, shadow_offset and outline_offset (which is
+ in fact the diameter of a dot, not a really an offset) is now
+ taken into account. Redundant matrix operation has been removed
+ for the vertical rendering.
+ Take note that the matrix operation in cairo are excuted in
+ reverse order.
+ https://bugzilla.gnome.org/show_bug.cgi?id=728636
+
+2015-07-24 10:15:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: seek at least in steps of a second
+ In case of very short files we might end up seeking in
+ steps of a fraction of a second, which is silly and gives
+ the impression that seeking doesn't actually work. Make
+ minimum seek step a second instead.
+
+2015-07-22 16:19:48 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Improve further the negotiation function
+ * Only send the caps event once if the query had support for the
+ overlay composition meta.
+ * Only do the allocation query if it is supported through caps.
+ * Send overlay_caps before doing allocation query rather then normal
+ caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-07-22 20:50:10 +0200 Rico Tzschichholz <ricotz@ubuntu.com>
+
+ * ext/pango/Makefile.am:
+ basetextoverlay: Add missing linking against -lm
+
+2015-07-21 18:40:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/pango/gstbasetextoverlay.h:
+ basetextoverlay: Ensure meta coordinate are in stream scale
+ The GstVideoOverlayComposition meta coordinates should always be
+ in stream scale, regardless of the window size downstream. This
+ way the sink can always scale the composition if the window size
+ have changed after a buffer (with his meta) was rendered before.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-07-21 14:12:41 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/pango/gstbasetextoverlay.h:
+ basetextoverlay: Reorder and cleanup class attribute
+ Also add a minimum amount of comment so we can understand what
+ is doing what.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-07-15 21:56:17 +0300 Ville Skyttä <ville.skytta@iki.fi>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Treat *.umx (Unreal Music Package) as audio/x-mod
+ https://bugzilla.gnome.org//show_bug.cgi?id=752436
+
+2015-07-20 16:25:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Fix upstream composition handling
+ We need to update the render when upstream composition changes
+ or if it was removed.
+ http://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-07-20 16:20:24 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Clear reconfigure flags before negotation
+ This avoids negotiating twice. Current the _setcaps() patch does
+ not clear the initial reconfigure flags, which lead to systematic
+ double renegotiation.
+ http://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-07-20 15:55:07 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetestoverlay: Always query window dimension
+ Remove the optimization to skip allocation query so we can
+ always have the latest window size information. Also, correctly
+ deal with the case where there is no window size information.
+ http://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-07-20 15:11:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Send caps before doing allocation query
+ This is currently a limitation of BaseTransform base class. Which means
+ pretty much every filters out there.
+ http://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-06-18 06:31:00 +0200 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Log GstVideoOverlayComposition negotiation
+ https://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-03-25 14:10:10 +0100 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/pango/gstbasetextoverlay.h:
+ basetextoverlay: Receive window size event and adjust rendering
+ * cache window size event and update handle ratio
+ * init width with 1, don't use 0
+ * don't update overlay when receiving same window size
+ * receive window size from allocation query
+ https://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-03-19 17:59:16 +0100 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/pango/gstbasetextoverlay.h:
+ basetestoverlay: Pass down meta buffers from upstream that supports GstVideoOverlayComposition
+ This makes pipelines with multiple textoverlay elements possible.
+ The meta data is collected from the upstream textoverlay element,
+ merged into a new GstVideoOverlayComposition and passed down downstream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751157
+
+2015-04-20 15:04:56 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: Fix PLC frame size calculations
+ Previously, PLC frames always had a length of 120ms, which caused audio
+ quality degradation and synchronization errors. Fix this by calculating an
+ appropriate length for the PLC frame.
+ The length must be a multiple of 2.5ms. Calculate a multiple of 2.5ms that
+ is nearest to the current PLC length. Any leftover PLC length that didn't
+ make it into this frame is accumulated for the next PLC frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=725167
+
+2015-07-10 12:49:01 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ depayloader: Use input segment start
+ When there is no clock_base provided, the start position is
+ set to 0 instead of the original segment start value. This
+ would break synchronization if start was not 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752228
+
+2015-07-16 21:26:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: add DASH MPD typefinder
+ Moved from dashdemux plugin in -bad.
+
+2015-07-16 10:07:45 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * tests/examples/seek/jsseek.c:
+ jsseek: fix memory leaks
+ ctx, list and visual_entries are not being freed
+ resulting in memory leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=752454
+
+2015-07-16 17:15:33 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/ogg/gstogmparse.c:
+ * ext/pango/gsttextrender.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/videoconvert/gstvideoconvert.c:
+ Update mailing list address from sourceforge to freedesktop
+
+2015-07-16 10:54:29 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-device-monitor.c:
+ tools: gst-device-monitor: fix props leak
+ CID 1311942
+
+2015-07-15 18:22:28 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tools/gst-device-monitor.c:
+ device-monitor: print device properties
+
+2015-07-15 12:45:10 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/gstvideopool.c:
+ * gst-libs/gst/video/video-chroma.c:
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-resampler.c:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/videooverlay.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/videotestsrc.c:
+ video: improve logging
+ Add logging categories for most video objects.
+ Remove some useless debug lines in video-info and videotestsrc.
+ Add a performance debug line in the video scaler.
+
+2015-07-15 12:46:07 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tests/examples/seek/jsseek.c:
+ jsseek: fix tag list leak
+ tags are being leaked while updating the streams in jsseek
+ https://bugzilla.gnome.org/show_bug.cgi?id=752400
+
+2015-07-15 10:50:46 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tests/examples/playback/playback-test.c:
+ playback-test: fix tag list leak
+ tags are being leaked while updating the streams in playback-test
+ https://bugzilla.gnome.org/show_bug.cgi?id=752397
+
+2015-07-14 17:17:34 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: Include generated enum types in gstrtsptransport.h
+ GST_TYPE_RTSP_LOWER_TRANS used to be defined in there, not
+ including the generated file makes older gst-p-good fail to build,
+ so it constitues an API break.
+
+2015-07-14 15:58:43 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/tcp/gstsocketsrc.c:
+ * gst/tcp/gstsocketsrc.h:
+ socketsrc: add caps property
+ Add caps property that allows the src to easily negotiate a format.
+
+2015-07-14 13:00:03 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * tests/examples/playback/playback-test.c:
+ playback-test: fix memory leak
+ context during main and filter list during init
+ visualization are not being freed resulting in memory leak
+ and app->vis_entries
+ https://bugzilla.gnome.org/show_bug.cgi?id=752359
+
+2015-07-14 00:03:10 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: only try to expose complete groups
+ When switching to a new chain it might be that this new chain
+ is not yet ready to be exposed so check it before exposing.
+ Can happen with mpegts that might delay adding pads or pushing data
+ until it has found the PMT/PAT/PCR and that may take a while depending
+ on the stream.
+ It happened frequently with HLS:
+ http://vevoplaylist-live.hls.adaptive.level3.net/vevo/ch1/appleman.m3u8
+
+2015-07-14 00:02:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: fix typo
+ Hided -> hid
+
+2015-05-27 18:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ * gst-libs/gst/rtp/gstrtpbasedepayload.h:
+ rtp: rtpbasedepayload: add process_rtp_packet() vfunc
+ Add process_rtp_packet() vfunc that works just like the
+ existing process() vfunc only that it takes the GstRTPBuffer
+ that the base class has already mapped (with MAP_READ),
+ which means that the subclass doesn't have to map it again,
+ which allows more performant processing of input buffers
+ for most RTP depayloaders.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750235
+
+2015-07-10 11:53:24 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Require the streamvolume interface on the sink when using the sink's volume/mute properties
+ If the sink has properties named volume and mute, we have no idea about their
+ meaning. The streamvolume interface standardizes the meaning.
+ In the case of osxaudiosink for example, the current volume property has a
+ range of 0.0 to 1.0, but we need 0.0 to 10.0 or similar. Also osxaudiosink
+ has no mute property. As such, the volume element should be used here instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752156
+
+2015-07-09 10:47:20 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst-libs/gst/video/video-frame.h:
+ doc/build: Fix doc typos
+ This minor update should workaround a build system bug. While the
+ makefile has been updated to generate more enum type, there is nothing
+ that updates the header and would lead to the generated code to be
+ produced again. This minor doc fix should ensure no one get a build with
+ missing symbols.
+
+2015-07-09 17:20:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ Revert "win32 def: Remove video flags symbol that don't exist"
+ This reverts commit b20cc6a02a007521eabceeceb60356e5a252f38a.
+ They are actually there in the autogenerated enum header/source file.
+
+2015-07-09 10:15:11 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * win32/common/libgstvideo.def:
+ win32 def: Remove video flags symbol that don't exist
+ There has been a some refactoring and these symbols don't exist anynmore.
+ So remove it from the win32 def. This should fix distcheck.
+
+2015-07-07 19:56:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasedepayload: fix typo in comment
+
+2015-07-07 15:05:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasepayload: fix possible segment event leak
+ Need to clear it when shutting down, not when starting up.
+ Fixes leak in rtp-payloading unit test.
+
+2015-07-07 22:23:57 +0900 Hyunjun Ko <zzoonis@gmail.com>
+
+ * gst-libs/gst/audio/gstaudiometa.c:
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/video-overlay-composition.c:
+ video/audio meta: transform_func: return FALSE if not supported or failed
+ https://bugzilla.gnome.org/show_bug.cgi?id=751778
+
+2015-07-07 19:55:44 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: refactor to use gst_pad_push_event
+ Right now navigation events are being sent via gst_pad_send_event
+ after getting the peer pad of the sinkpad.
+ But the same functionality can be done using gst_pad_push_event
+ without need of getting peer pad in xvimagesink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752059
+
+2015-07-07 14:32:25 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * win32/common/libgstvideo.def:
+ video: Add some more GTypes for enums
+
+2015-07-02 07:36:12 +0200 Tobias Mueller <muelli@cryptobitch.de>
+
+ * gst-libs/gst/video/video-scaler.c:
+ GstVideoScaler: Initialised scaling functions to get rid of compiler messages
+ E.g.
+ video-scaler.c: In function 'gst_video_scaler_horizontal':
+ video-scaler.c:1332:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ func (scale, src, dest, dest_offset, width, n_elems);
+ ^
+ video-scaler.c: In function 'gst_video_scaler_vertical':
+ video-scaler.c:1373:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ func (scale, src_lines, dest, dest_offset, width, n_elems);
+ ^
+ GCC's analyses seem to be correct, for the simple fact that if you pass
+ get_functions a known format, but no hscale or vscale, it'll return
+ True without having done anything.
+ Some callers check for the scale values to be not NULL, but then
+ hscale->resampler.max_taps could return 0.
+ A different approach to the one presented in this patch is to check
+ for those max_taps, too, before calling get_functions.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=752051
+
+2015-07-07 19:45:43 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * sys/ximage/ximagesink.c:
+ ximagesink: Post navigation events as message on the bus
+ post unhandled events to bus, so that
+ application can utilise the same if needed
+ https://bugzilla.gnome.org/show_bug.cgi?id=752043
+
+2015-07-07 19:35:40 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * sys/ximage/ximagesink.c:
+ ximagesink: fix navigation event leak
+ Create event only when pad is created
+ and send the event to pad.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752041
+
+2015-07-07 09:31:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix pad memory leak
+ pad is not being freed when xwindow is not created
+ https://bugzilla.gnome.org/show_bug.cgi?id=752042
+
+2015-07-07 08:53:09 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tools/gst-play.c:
+ gst-play: fix memory leak
+ In gst-play, for GST_MESSAGE_ELEMENT bus message,
+ event is being allocated through
+ gst_navigation_message_parse_event, but not freed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752040
+
+2015-07-03 21:48:52 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * sys/ximage/ximage.c:
+ * sys/ximage/ximagepool.c:
+ * sys/ximage/ximagepool.h:
+ * sys/ximage/ximagesink.c:
+ * sys/ximage/ximagesink.h:
+ * sys/xvimage/xvcontext.c:
+ * sys/xvimage/xvimage.c:
+ * sys/xvimage/xvimagepool.c:
+ * sys/xvimage/xvimagesink.c:
+ * sys/xvimage/xvimagesink.h:
+ x/xv_image_sink: rename for consitency
+ Insert '_' to match the CamelCase. This is needed so that the plugin docs can
+ guess the names from the type name.
+
+2015-07-03 21:35:32 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ docs: update master doc for plugins
+
+2015-07-06 10:05:53 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: also check moof to recognize video/quicktime
+ Helps recognizing fragmented files with the right type
+
+2015-07-06 15:36:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * win32/common/libgstvideo.def:
+ docs: Add new symbols to the docs and .def files
+
+2015-07-06 12:53:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-info.h:
+ * gst-libs/gst/video/video-info.h:
+ {audio,video}info: Add GST_TYPE_{AUDIO,VIDEO}_INFO macros
+
+2015-07-06 11:36:58 +0200 Marcin Kolny <marcin.kolny@flytronic.pl>
+
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-info.h:
+ video-info: implement GstVideoInfo as boxed type
+ GstVideoInfo usually is created on the stack, but boxed type can be useful
+ for bindings.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752011
+
+2015-07-02 20:50:00 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * tests/check/libs/rtp.c:
+ rtcpbuffer: Fix validation of packets with padding
+ The padding (if any) is included in the length of the last packet, see
+ RFC 3550.
+ Section 6.4.1:
+ padding (P): 1 bit
+ If the padding bit is set, this individual RTCP packet contains
+ some additional padding octets at the end which are not part of
+ the control information but are included in the length field. The
+ last octet of the padding is a count of how many padding octets
+ should be ignored, including itself (it will be a multiple of
+ four).
+ Section A.2:
+ * The padding bit (P) should be zero for the first packet of a
+ compound RTCP packet because padding should only be applied, if it
+ is needed, to the last packet.
+ * The length fields of the individual RTCP packets must add up to
+ the overall length of the compound RTCP packet as received.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751883
+
+2015-07-01 17:09:35 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Fix setting default pixel-aspect-ratio
+ It's needed to check if pixel-aspect-ratio exists before fixating.
+ It does not exist if input caps is not set yet and allowed caps
+ does not contain pixel-aspect-ratio (e.g. when using GST_VIDEO_CAPS_MAKE)
+ https://bugzilla.gnome.org/show_bug.cgi?id=751932
+
+2015-07-03 21:58:04 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f74b2df to 9aed1d7
+
+2015-07-03 21:16:27 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ * gst/adder/gstadder.h:
+ * gst/tcp/gstmultisocketsink.h:
+ docs: order and canonicalize the -sections.txt file
+ Have all sections in alphabetical order. Also make the macro order consistent.
+ This is a preparation for generating the file. Remove GET_CLASS macro for
+ some elements, since it is not used and the header is not installed.
+
+2015-07-03 21:09:29 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ cdparanoiasrc: remove unused defines
+
+2015-07-03 21:08:03 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: fix debug categories
+ Use a local category for the default category and fix the import for the
+ performance category.
+
+2015-07-02 10:47:45 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Fix bug with unused upstream_has_meta
+ The intention was to skip the allocation query if upstream has decided
+ to use the overlay meta feature in the caps. We can safely assume that
+ upstream have done that query already before making this decision. This
+ is an optimization since doing allocation queries is relatively
+ expensive.
+ CID #1308943
+
+2015-07-02 10:27:39 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ Revert "basetextoverlay: remove dead code"
+ This reverts commit e863e5f8a98ceec0ec0bd24274bbae8795e0ab75.
+
+2015-07-02 14:52:47 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: remove dead code
+ upstream_has_meta is set to FALSE and never changed. The two checks for if
+ upstream_has_meta will never go to the true branch. Removing the boolean
+ and the true branches of these checks.
+ CID #1308943
+
+2015-07-02 13:15:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: Don't try to get buffers from an empty adapter
+
+2015-07-01 10:58:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ {audio,video}{en,de}oder: Also copy POOL metas and make sure to copy over metas when creating subbuffers
+ POOL meta just means that this specific instance of the meta is related to a
+ pool, a copy should be made when reasonable and the flag should just not be
+ set in the copy.
+
+2015-06-29 18:00:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ audiodecoder: Add transform_meta() vfunc with default implementation
+ The default implementation copies all metadata without tags, and metadata
+ with only the audio tag. Same behaviour as in GstAudioFilter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742385
+
+2015-06-29 17:38:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.h:
+ audioencoder: Add transform_meta() vfunc with default implementation
+ The default implementation copies all metadata without tags, and metadata
+ with only the audio tag. Same behaviour as in GstAudioFilter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742385
+
+2015-06-29 15:58:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ videodecoder: Add transform_meta() vfunc with default implementation
+ The default implementation copies all metadata without tags, and metadata
+ with only the video tag. Same behaviour as in GstVideoFilter.
+ This currently does not work if the ::parse() vfunc is implemented as all
+ metas are getting lost inside GstAdapter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742385
+
+2015-06-29 13:59:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/gstvideoencoder.h:
+ videoencoder: Add transform_meta() vfunc with default implementation
+ The default implementation copies all metadata without tags, and metadata
+ with only the video tag. Same behaviour as in GstVideoFilter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742385
+
+2015-06-30 10:37:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c:
+ rtpbaseaudiopayload: Don't copy memory if not needed, just append payload to the RTP buffer
+
+2015-06-30 07:26:00 +0900 danny song <danny.song.ga@gmail.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: remove unnecessary break
+ https://bugzilla.gnome.org/show_bug.cgi?id=751690
+
+2015-06-29 16:16:06 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ videoscaler: remove check for below zero for unsigned value
+ CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
+ number since it is a division of an unsigned integer (i). Removing that check
+ and only checking if it is bigger than max and setting it appropriately.
+ CID #1308950
+
+2015-06-29 13:06:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Also copy metas if their API has no tags attached to it
+ This is the default basetransform behaviour, being more strict than that
+ is not really useful.
+
+2015-06-29 13:06:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: Also copy metas if their API has no tags attached to it
+ This is the default basetransform behaviour, being more strict than that
+ is not really useful.
+
+2015-06-29 13:06:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: Also copy metas if their API has no tags attached to it
+ This is the default basetransform behaviour, being more strict than that
+ is not really useful.
+
+2015-06-29 13:05:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideofilter.c:
+ videofilter: Also copy metas if their API has no tags attached to it
+ This is the default basetransform behaviour, being more strict than that
+ is not really useful.
+
+2015-06-25 00:04:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.2 ===
+
+2015-06-24 23:24:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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-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:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ Release 1.5.2
+
+2015-06-24 22:49:29 +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
+
+2015-06-24 11:14:21 +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
+
+2015-06-17 18:03:09 +0800 Song Bing <b06498@freescale.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Unblock EOS wait when track switching.
+ sink_event () will blocked on EOS event. which will cause can't
+ send event when switch EOS track to non-EOS one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750761
+
+2015-06-22 20:54:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Don't wait for sparse streams when doing stream switches
+ Their stream-start event might come a bit later, like just before the first
+ buffer... and queues might run full before that happens.
+
+2015-06-22 20:29:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Add some more debug output
+
+2015-06-22 20:17:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Reset group start time when flushing
+ We reset the group start time to the running time of the start of the other
+ streams that are not flushed. This fixes seeking in gapless mode after the
+ first track has played.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750013
+
+2015-06-22 19:51:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Only drop everything after the ; of a session header in requests
+ For responses it is actually allowed and used to signal the timeout to the
+ client!
+ https://bugzilla.gnome.org/show_bug.cgi?id=736267
+
+2015-06-18 17:38:09 +0800 Lyon Wang <lyon.wang@freescale.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: Fix alaw/mulaw channel positions
+ For alaw/mulaw we should also try to initialize the channel positions in the
+ ringbuffer's audio info. This allow pulsesink to directly use the channel
+ positions instead of using the default zero-initialized ones, which doesn't
+ work well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751144
+
+2015-06-22 16:53:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/libs/libsabi.c:
+ tests: fix cpp directives
+
+2015-06-22 15:59:42 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/libsabi.c:
+ * tests/check/libs/struct_ppc64.h:
+ tests: add PPC64 abi struct sizes
+
+2015-06-22 14:51:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Reset suburi also when receiving an error message from the sub uridecodebin
+ http://bugzilla.gnome.org/show_bug.cgi?id=751118
+
+2015-06-17 10:20:54 -0500 Brijesh Singh <brijesh.ksingh@gmail.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: free group->suburi on failure
+ If suburidecodebin is failed to negotiate (e.g file does not exist)
+ then free internal suburi variable so that 'current-suburi' property
+ returns correct status.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751118
+
+2015-06-15 16:08:10 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: set building_chain to NULL when deactivating chain
+ The chain is about to be invalidated so we shouldn't keep it around.
+ Prevent a double free crash when the demuxer is being finalized.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751000
+
+2015-06-15 13:43:53 +0200 Mersad Jelacic <mersad@axis.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: Add bitrate to the tags
+ https://bugzilla.gnome.org/show_bug.cgi?id=750992
+
+2015-06-19 19:51:25 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: fix seeking issue
+ For positive seeking segment.stop value will be -1,
+ when we change rate to -1, then the stop value will be udpated
+ with the current position. And then again if we change rate to 1,
+ the segment.stop value does not get updated and remains as position
+ where we last changed rate to -1. Hence playback stops at that point.
+ In case of positive rates, call gst_element_new_seek with correct values
+ https://bugzilla.gnome.org/show_bug.cgi?id=751213
+
+2015-06-18 21:02:48 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtphdrext.c:
+ doc: Fix gsttrtphdrext section name
+
+2015-06-18 18:23:45 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * win32/common/libgstvideo.def:
+ video: Add missing new symbol to win32 def file
+ Fixes make distcheck
+
+2015-06-19 02:19:12 +1000 Jan Schmidt <jan@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Add gst_video_multiview_guess_half_aspect() to the docs
+
+2015-06-15 16:04:55 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/video-multiview.c:
+ * gst-libs/gst/video/video-multiview.h:
+ multiview: Add gst_video_multiview_guess_half_aspect()
+ Add a utility function that, given a video size and a
+ packed stereoscopic mode, attempts to guess if the video
+ is packed at half resolution per view or not, since
+ very few videos provide the information.
+
+2015-06-17 17:09:46 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: make sure we draw enough border for YUY2 formats
+ Round width up to 2 so that we draw all border pixels for YUY2 formats
+
+2015-06-17 16:43:03 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix scaling of odd width for YUY2 formats
+ We need to scale groups of 4 bytes for YUY2 formats so round up to 4.
+ It's possible that there is no Y byte for the last pixel so make sure
+ we clamp correctly.
+
+2015-06-17 10:02:08 +0200 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ discoverer: Fix a wrong naming in the documentation
+ gst_discoverer_stream_get_missing_elements_installer_details does not
+ exist, one should use gst_discoverer_info_get_missing_elements_installer_details
+
+2015-06-16 18:04:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/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/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ gi: Use INTROSPECTION_INIT for --add-init-section
+ This new define was added to common. The new init section fixed
+ compilation warning found in the init line that was spread across
+ all files.
+
+2015-06-16 17:47:24 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6015d26 to f74b2df
+
+2015-06-16 22:32:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: error out instead of crashing if there's no playbin element
+
+2015-06-16 16:08:39 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-dither.c:
+ * gst-libs/gst/video/video-scaler.c:
+ gi: Skip Scaler, Chroma, Conveter, Dither constructor
+ Please box these types before removing the skip mark.
+
+2015-06-16 16:07:27 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/video-overlay-composition.c:
+ gi: Add (transfer none) for various video meta
+ These method chains gst_buffer_add_meta() which is also transfer
+ none.
+
+2015-06-16 15:50:13 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ gi: Fix warnings in GstRtsp
+ * The custom GSource is not boxed (skip for now)
+ * The comment block has wrong name for _read_socket()
+
+2015-06-16 15:16:33 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/riff/Makefile.am:
+ gi: Don't produce gir and typlib for GstRiff
+ The API does not follow the type naming convention. Re-enable
+ only if one take the time to box and rename (see (rename-to SYMBOL)
+ annotation) all types.
+
+2015-06-16 14:36:44 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/fft/gstfftf32.c:
+ * gst-libs/gst/fft/gstfftf64.c:
+ * gst-libs/gst/fft/gstffts16.c:
+ * gst-libs/gst/fft/gstffts32.c:
+ gi: Skip fft constructor for now
+ These types have never been boxed, hence cannot be used
+ safely in interpreted languages. This fixes warnings.
+
+2015-06-16 14:34:04 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/audio/audio-info.c:
+ * gst-libs/gst/audio/gstaudiobasesink.h:
+ * gst-libs/gst/audio/gstaudiometa.c:
+ gi: Fix warnings in libgstaudio
+ * Duplicate section
+ * Miss-named parameter
+ * Missing transfer none annotation for meta
+
+2015-05-27 12:20:19 +0300 Lazar Claudiu <lazar.claudiu.florin@gmail.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/pango/gstbasetextoverlay.h:
+ basetextoverlay: add "draw-shadow" and "draw-outline" properties
+ https://bugzilla.gnome.org/show_bug.cgi?id=749823
+
+2015-06-13 13:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: fix gtk-doc chunk for new function
+
+2015-05-12 14:12:52 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Rewrite negotiation method
+ This cleanup the negotiation function by properly splitting the probe
+ and the decisions. This allow handling correctly pipeline where upstream
+ caps have special memory type. An example pipeline is:
+ gltestsrc ! textoverlay text=bla ! fakesink
+ The upstream caps will be memory:GLMemory, which isn't supported by the
+ blitter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749243
+
+2015-06-05 14:30:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: Don't share internal pool
+ Sharing the internal pool results in situation where the pool may have
+ two upstream owners. This creates a race upon deactivation. Instead,
+ always offer a new pool, and keep the internal pool internal in case
+ we absolutely need it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748344
+
+2015-06-05 14:28:41 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/ximage/ximagesink.c:
+ ximagesink: Don't share internal pool
+ Sharing the internal pool results in situation where the pool may have
+ two upstream owners. This create a race upon deactivation. Instead,
+ always offer a new pool, and keep the internal pool internal in case
+ we absolutely need it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748344
+
+2014-11-26 21:06:57 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/gstvideoencoder.h:
+ * win32/common/libgstvideo.def:
+ videoencoder: Add gst_video_encoder_set_min_pts()
+ For streams with reordered frames this can be used to ensure that there
+ is enough time to accomodate first DTS, which may be less than first PTS
+ https://bugzilla.gnome.org/show_bug.cgi?id=740575
+
+2015-06-12 19:58:34 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ Update .def file for new API
+
+2015-06-13 01:35:52 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Copy multiview-mode, flags and view count from ref info
+ When copying info from the reference input state, duplicate
+ all the fields of the video info. The sub-class will have the
+ chance to override them later.
+
+2015-06-12 16:57:39 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: enforce same taps when combining scalers
+
+2015-06-12 16:52:27 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: make sure to clamp to max width
+ When estimating the area that should first be vertically scaled, make
+ sure we clamp to the max input size or else we get invalid reads.
+
+2015-06-12 16:47:03 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/video-scaler.h:
+ video-scaler: Enforce same taps on Y and UV scalers for merged formats
+ Make sure we have the same number of taps for the Y and UV scalers so
+ that the scalers can be merged correctly.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749764
+
+2015-06-12 12:50:35 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Add a FIXME 2.0 for gst_rtsp_connection_create_from_socket()
+ There's a couple of redundant arguments from the pre-GIO days.
+
+2015-06-11 23:32:55 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: check for full UTF-8 BOM in MSS typefinder
+ https://bugzilla.gnome.org/show_bug.cgi?id=750802
+
+2015-06-11 18:14:47 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: UTF-8 MSS Manifest detection support
+ Check if the first bytes of data contain an UTF-8 BOM.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750802
+
+2015-06-11 16:18:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Check in autoplug_continue against the subtitle factory caps correctly
+ 6a2f017bfa9cb73c6db65eea0b84b1d5b56febb7 changed it to check the subtitle
+ factory caps if there is a text-sink but we fail to get its sinkpad. What
+ actually should be done here is to use the factory caps if there is no
+ text-sink at all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750785
+
+2015-06-11 23:01:48 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Fix some warnings with clang around multiview enums
+ There is the GstVideoMultiviewMode enum and the
+ GstVideoMultiviewFramePacking, which is a subset of the
+ multiview modes, with the same values as the corresponding
+ types from the full enum. Do some casts and use the right
+ times to avoid implicitly using/passing GstVideoMultiviewFramePacking
+ when a GstVideoMultiviewMode is needed.
+
+2015-06-11 12:21:08 +1000 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/libs/video.c:
+ tests: Fix video libs test for multiview GstVideoInfo change
+ The GstVideoInfo struct was changed late in integrating the
+ multiview changes, and I forgot to run and fix the unit test.
+
+2015-06-11 11:12:39 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Implement multiview frame-packing overrides
+ Add GstVideoMultiviewFramePacking enum, and the
+ video-multiview-mode and video-multiview-flags
+ properties on playbin.
+ Use a pad probe to replace the multiview information in
+ video caps sent out from uridecodebin.
+ This is a part implementation only - for full
+ correctness, it should also modify caps in caps events,
+ accept-caps and allocation queries.
+ https://bugzilla.gnome.org/show_bug.cgi?id=611157
+
+2015-06-11 11:12:39 +1000 Jan Schmidt <jan@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/video-frame.h:
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-info.h:
+ * gst-libs/gst/video/video-multiview.c:
+ * gst-libs/gst/video/video-multiview.h:
+ * gst-libs/gst/video/video.h:
+ * tests/check/libs/video.c:
+ * win32/common/libgstvideo.def:
+ video: Add multiview/stereo support
+ Add flags and enums to support multiview signalling in
+ GstVideoInfo and GstVideoFrame, and the caps serialisation and
+ deserialisation.
+ videoencoder: Copy multiview settings from reference input state
+ Add gst_video_multiview_* support API and GstVideoMultiviewMeta meta
+ https://bugzilla.gnome.org/show_bug.cgi?id=611157
+
+2015-06-10 14:33:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c:
+ rtpbaseaudiopayload: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
+
+2015-06-10 12:26:38 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: fix the channel of color balance element
+ When traversing the color balance element channel list to find the one that
+ matches with the playsink proxy, the assignation was set to iterator of the
+ playsink proxy, not the balance element. Thus, the mapping to the values of
+ the balance element channel was wrong.
+ This patch fixes the assignation of the color balance element channel, so the
+ mapping to the channel of the color balance element is fixed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750691
+
+2015-06-10 15:50:12 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: cannot enable text flag while playing
+ when text playbin is not enabled in the beginning, then
+ video_srcpad_stream_synchronizer gets linked to videochain->sinkpad
+ and when we try to enable text bin during play, since it is already linked to videochain,
+ text chain does not get linked properly. Hence unlinking the same
+ before linking to text chain
+ https://bugzilla.gnome.org/show_bug.cgi?id=748908
+
+2015-06-10 09:59:49 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * win32/common/libgstrtsp.def:
+ win32: Update defs file
+
+2015-06-05 22:04:24 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ GstRTSPConnection: Add GTlsInteraction support
+ https://bugzilla.gnome.org/show_bug.cgi?id=750471
+
+2015-06-09 21:24:07 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * tools/gst-play.c:
+ tools: gst-play: don't print 64 whitespaces next to the time indication
+ Printing 64 whitespaces to erase the "Paused" message (after \r) would make
+ it wrap to the next line on shorter terminals. Instead we only print the
+ amount of spaces needed. Also mark the "Paused" string for translation
+ while we're at it.
+
+2015-06-09 14:31:15 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * Makefile.am:
+ cruft: add the obsolete tmpl dir to cruft-dirs
+
+2015-06-09 22:03:37 +1000 Jan Schmidt <jan@centricular.com>
+
+ * win32/common/libgstaudio.def:
+ Update win32 exports
+
+2013-12-09 18:46:14 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * ext/alsa/gstalsasink.c:
+ alsa: report recoverable device failures to base class
+ This gives custom slave methods in the base class a chance to
+ resynchronize themselves
+ Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
+ https://bugzilla.gnome.org/show_bug.cgi?id=708362
+
+2013-12-09 17:08:15 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ * gst-libs/gst/audio/gstaudiobasesink.h:
+ audiobasesink: added custom clock slaving method
+ This new clock slaving method allows for installing a callback that is
+ invoked during playback. Inside this callback, a custom slaving
+ mechanism can be used (for example, a control loop adjusting a PLL or an
+ asynchronous resampler). Upon request, it can skew the playout pointer
+ just like the "skew" method. This is useful if the clocks drifted apart
+ too much, and a quick reset is necessary.
+ Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
+ https://bugzilla.gnome.org/show_bug.cgi?id=708362
+
+2015-06-09 11:30:15 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From d9a3353 to 6015d26
+
+2015-06-09 10:16:34 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play-1.0.1:
+ * tools/gst-play.c:
+ tools: gst-play: add shortcuts to switch audio/subtitle/video tracks
+
+2014-11-05 09:41:36 +0200 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst/playback/gstplaybackutils.c:
+ playback: Skip 'ANY' capsfeature while finding the count of common capsfeatures
+ https://bugzilla.gnome.org/show_bug.cgi?id=687182
+
+2014-11-05 09:40:43 +0200 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplaybackutils.c:
+ * gst/playback/gstplaybackutils.h:
+ * gst/playback/gstplaybin2.c:
+ playback: Add gstplaybackutils.{h,c} to deploy the common subroutines
+ Bring some of the helper functions in gstplaybin2.c to new files
+ gstplaybackutils.{h,c} which can be utilized by other files
+ in gst/playback too.
+ https://bugzilla.gnome.org/show_bug.cgi?id=687182
+
+2015-06-08 23:07:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From d37af32 to d9a3353
+
+2015-06-08 20:32:02 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * tools/gst-play.c:
+ tools: gst-play: sort directory entries
+ When adding a directory to the playlist, the order would be whatever
+ g_dir_read_name returned. Sorting these using natural sort order.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750585
+
+2015-06-08 20:17:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvcontext.c:
+ ximagesink, xvimagesink: fix string leaks when setting class hint
+ https://bugzilla.gnome.org/show_bug.cgi?id=750455
+
+2015-06-08 13:01:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/video-color.c:
+ video: Allow using bt2020 by name in colorimetry
+ As the lookup stops at the first element in the array with a NULL
+ name, bt2020 could not be used by name. Moving up this entry
+ fixes the issue.
+
+2015-06-05 16:01:05 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/ximage/ximagesink.c:
+ ximagesink: set WM_CLASS of window
+ Set WM_CLASS of the ximagesink window so window managers can apply rules
+ based on xprop filtering.
+
+2015-06-05 15:58:39 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/xvimage/xvcontext.c:
+ xvimagesink: set WM_CLASS of window
+ Set WM_CLASS of the xvimagesink window so window managers can apply rules
+ based on xprop filtering.
+
+2015-06-07 23:06:08 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 21ba2e5 to d37af32
+
+2015-06-07 18:49:48 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ libvisual: clean dereferences of private structures
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-06-07 18:23:23 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ * ext/libvisual/gstaudiovisualizer.h:
+ libvisual: make private all variable subclasses don't need
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-06-07 17:31:55 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From c408583 to 21ba2e5
+
+2015-06-07 17:00:05 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ docs: remove variables that we define in the snippet from common
+ This is syncing our Makefile.am with upstream gtkdoc.
+
+2015-06-07 17:16:13 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 241fcb7 to c408583
+
+2015-06-07 16:44:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.1 ===
+
+2015-06-07 10:04:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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-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/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/gstrtsp-enumtypes.h:
+ * win32/common/pbutils-enumtypes.c:
+ * win32/common/pbutils-enumtypes.h:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ Release 1.5.1
+
+2015-06-07 09:35:03 +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
+
+2015-06-05 16:44:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Always prefer downstream's ssrc suggestion if any
+ Otherwise ssrc changes via rtpsession's (deprecated!) internal-ssrc property
+ are not possible anymore. rtpsession was now patched to only suggest an ssrc
+ if it makes sense to do so.
+ In 2.0 we should get rid of all the properties that are also negotiated via
+ caps, the code and behaviour is too confusing otherwise.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749581
+
+2015-06-05 10:16:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * win32/common/libgstrtp.def:
+ rtcpbuffer: Improve documentation of new functions a bit
+ Also actually add them to the documentation.
+
+2015-06-03 11:20:35 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ * tests/check/libs/rtp.c:
+ rtcpbuffer: Update package validation to support reduced size rtcp packets
+ According to this section of the rfc.
+ https://tools.ietf.org/html/rfc5506#section-3.4.2
+ The validation should be updated to accept more types of RTCP
+ packages, with this mask change feedback packages will be also
+ accepted.
+ Change-Id: If5ead59e03c7c60bbe45a9b09f3ff680e7fa4868
+
+2015-06-04 19:03:51 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: copy metadata that only has the "audio" tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750406
+
+2015-06-04 19:00:45 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: copy metadata that only has the "audio" tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750406
+
+2015-06-04 17:59:17 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: copy metadata that only has the "audio" tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750406
+
+2015-05-20 18:16:07 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Serialize the top level DiscovererInfo
+ Which contains fields such as duration, uri and tags.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749673
+
+2015-06-04 16:31:12 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/codec-utils.c:
+ codec-utils: Add AAC channel configurations 11, 12 and 14 and levels 6 and 7
+
+2015-06-04 11:54:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: If channel/rate negotiation fails, fall back to stereo and 48kHz
+
+2015-06-04 11:45:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: gst_structure_fixate_field_nearest_int() only works if the structure has this field
+ Just set the rate/channels directly if the caps don't have this field.
+
+2015-06-02 16:14:39 +0200 Edward Hervey <edward@centricular.com>
+
+ * tests/check/generic/clock-selection.c:
+ * tests/check/libs/allocators.c:
+ * tests/check/libs/audio.c:
+ * tests/check/libs/fft.c:
+ * tests/check/libs/navigation.c:
+ * tests/check/libs/rtp.c:
+ * tests/check/libs/rtsp.c:
+ * tests/check/libs/rtspconnection.c:
+ * tests/check/libs/tag.c:
+ * tests/check/libs/xmpwriter.c:
+ * tests/check/pipelines/basetime.c:
+ * tests/check/pipelines/capsfilter-renegotiation.c:
+ * tests/check/pipelines/gio.c:
+ * tests/check/pipelines/simple-launch-lines.c:
+ * tests/check/pipelines/theoraenc.c:
+ * tests/check/pipelines/vorbisdec.c:
+ * tests/check/pipelines/vorbisenc.c:
+ check: Use GST_CHECK_MAIN () macro everywhere
+ Makes source code smaller, and ensures we go through common initialization
+ path (like the one that sets up XML unit test output ...)
+
+2015-06-02 16:02:37 +0200 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/opus.c:
+ check: Use GST_CHECK_MAIN () macro everywhere
+ Makes source code smaller, and ensures we go through common initialization
+ path (like the one that sets up XML unit test output ...)
+
+2015-06-02 12:47:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for video/x-cavs caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=727731
+
+2015-06-02 12:28:19 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * win32/common/libgstpbutils.def:
+ win32: Update def file for new encoding API
+
+2015-05-29 14:15:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: optimise payload mapping for buffers with one memory
+ Micro-optimisation: if the buffer consist of just one memory, we
+ know we have already mapped that memory to read the headers, so
+ no need to map it another time to get to the payload data, we
+ can just set up the payload data details right there and then
+ and avoid another map call in gst_rtp_buffer_get_payload().
+ Adds up when receiving RTP-payloaded raw video which can easily
+ be thousands of packets per frame.
+
+2015-05-21 13:59:55 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ * gst-libs/gst/rtp/gstrtpbasedepayload.h:
+ rtpbasedepayload: provide chain_list function on sink pad
+ Implement a chain_list function, which avoids lots of locking
+ compared to the default fallback implementation in GstPad.
+ We may also want to do some more sophisticated timestamp
+ tracking here at some point, but for now leave it up to the
+ jitterbuffer and/or subclasses (in case buffers in the
+ buffer list have no timestamp set on them, there may only
+ be a timestamp for the whole list on the first buffer).
+ This provides the exact same behaviour as the default
+ fallback implementation.
+
+2015-05-07 10:26:47 +0200 Thibault Saunier <tsaunier@gnome.org>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-profile.h:
+ * gst/encoding/gstencodebin.c:
+ encodebin: Add a way to enable/disabled a GstEncodingProfile
+ Summary:
+ So that the user can easily use the same encoding profile to render
+ with/without audio/video stream.
+ API:
+ gst_encoding_profile_is_disabled
+ gst_encoding_pofile_set_enabled
+ https://bugzilla.gnome.org/show_bug.cgi?id=749056
+
+2015-05-30 15:34:51 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: remove unnecessary variable
+ The second assignment of sret is never used. We can remove the first assignment
+ and use the value directly instead.
+
+2015-05-30 08:12:03 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/tag/id3v2frames.c:
+ id3v2frames: Fix compiler warnings
+ id3v2frames.c:951:20: error: unused variable 'utf16enc' [-Werror,-Wunused-const-variable]
+ static const gchar utf16enc[] = "UTF-16";
+ ^
+ id3v2frames.c:952:20: error: unused variable 'utf16leenc' [-Werror,-Wunused-const-variable]
+ static const gchar utf16leenc[] = "UTF-16LE";
+ ^
+ id3v2frames.c:953:20: error: unused variable 'utf16beenc' [-Werror,-Wunused-const-variable]
+ static const gchar utf16beenc[] = "UTF-16BE";
+ ^
+
+2015-05-30 01:03:46 +1000 Jan Schmidt <jan@centricular.com>
+
+ * docs/design/part-stereo-multiview-video.markdown:
+ part-stereo-multiview-video: Add a section of open design questions
+
+2015-05-30 00:58:38 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/video-format.h:
+ video-format: Fix minor docs typo
+
+2015-03-16 19:37:26 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/videotestsrc/gstvideotestsrc.h:
+ videotestsrc: Document the solid-color pattern
+
+2015-03-16 19:28:35 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstplay-enum.h:
+ playback: Document GST_PLAY_FLAG_SOFT_COLORBALANCE
+
+2014-10-09 01:13:29 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/gstvideometa.h:
+ * win32/common/libgstvideo.def:
+ video: Make gst_buffer_get_video_meta() a real function, Return lowest id
+ Instead of returning the first video meta found on a buffer, return the
+ one with the lowest id (which is usually the same thing, except on
+ multi-view buffers)
+
+2015-05-29 15:30:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: don't crash on unknown info types when deserializing
+ Handle unknown info types when deserializing instead of
+ dereferencing NULL pointers.
+ Coverity CID 1302394
+
+2015-05-29 13:15:59 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: prevent the sdp message parser from reading past the end of the buffer
+ Otherwise, a malformed SDP message could crash the application,
+ or even maliciously gather data from the memory located after
+ this buffer...
+ https://bugzilla.gnome.org/show_bug.cgi?id=750096
+
+2015-05-28 19:49:31 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * tests/check/elements/videorate.c:
+ tests: add test for videorate caps renegotiation after a framerate has been calculated and added to caps
+ The original 0/1 framerate must still be allowed to be configured
+ on the upstream side of videorate, otherwise future caps renegotiation
+ is going to fail.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750032
+
+2015-05-28 12:51:35 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: update the caps framerate only in the GST_PAD_SINK transform_caps direction
+ When a stream has a variable framerate, videorate calculates it and
+ forces it on the output caps. However, the code in _transform_caps()
+ currently also does that if the transform is going in the opposite
+ direction (GST_PAD_SRC), so during a renegotiation it tries to force
+ upstream to use the calculated framerate and it fails.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750032
+
+2015-05-26 08:06:50 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: use queue to avoid lock in audiotee audio branches
+ This part of pipeline is:
+ tee name=t ! visualizationbin ! streamsynchronizer name=s
+ t. ! s.
+ streamsynchronizer might block and it could starve the visualization
+ branch of the pipeline when it is enabled.
+ The visualization bin has queues internally but the other branch
+ that links the audiotee directly to the synchronizer is vulnerable
+ to block. Adding a queue between "t. ! s." fixes deadlocks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749676
+
+2015-05-26 13:11:00 +0300 Claudiu Florin Lazar <lazar.claudiu.florin@gmail.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: make deltax and deltay properties controllable
+ This will be more useful once we have absolute direct
+ control bindings.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749824
+
+2015-05-05 18:01:46 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix chain leak
+ Don't leak the building_chain when destroying.
+ Fix leaks with the validate.http.playback.reverse_playback.vorbis_theora_1_ogg
+ scenario.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748964
+
+2015-05-25 22:37:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/tag/id3v2frames.c:
+ tag: id3v2: fix parsing of UTF-16 text on systems with crippled iconv
+ Use g_utf16_to_utf8() instead of the more generic g_convert(), so
+ that we can extract text in UTF-16 format even on embedded systems
+ with crippled iconv support.
+ This code path is exercised by the id3demux test_unsync_v23
+ check in gst-plugins-good.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741144
+
+2015-05-25 22:37:06 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Add new generated rtp enum files to .gitignore
+
+2015-05-24 18:58:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: keep configured playback rate and trick mode when seeking
+ Instead of resetting rate to 1.0
+
+2015-05-24 18:47:25 +0100 Tim-Philipp Müller <tim@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 for new translatable strings
+
+2015-05-24 18:46:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: mark more strings for translation
+
+2015-05-23 01:50:11 +0900 danny song <danny.song.ga@gmail.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: add keyboard shortcut help
+ https://bugzilla.gnome.org/show_bug.cgi?id=749740
+
+2015-05-23 12:02:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: add back videoscale unit test
+ Has been removed in 835422b2 as part of porting
+ things over to the new videoscale API.
+
+2015-05-21 12:10:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play-1.0.1:
+ * tools/gst-play.c:
+ tools: gst-play: enable interative mode by default
+ And change --interactive option to --no-interactive.
+
+2015-05-21 13:07:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/Makefile.am:
+ rtp: Clean G-I files on make clean too
+
+2015-05-20 16:23:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/Makefile.am:
+ rtp: Add builddir to the include path for gobject-introspection
+ And also add missing headers/sources
+ https://bugzilla.gnome.org/show_bug.cgi?id=749632
+
+2015-05-20 15:40:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstrtp.def:
+ * win32/common/libgstrtsp.def:
+ win32: Update exports
+
+2015-05-20 13:36:30 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtp/gstrtpdefs.h:
+ * gst-libs/gst/rtp/rtp.h:
+ rtp: Add GstRTPProfile enum
+
+2015-05-20 13:35:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: Add FIXME 2.0 comment about GstRTSPTransport being an enum instead of flags
+
+2015-05-20 13:33:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: Use glib-mkenums to generate GstRTSPProfile and GstRTSPLowerTrans GTypes
+
+2015-05-20 10:22:48 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/ogg/gstoggdemux.c:
+ Revert "oggdemux: Prevent seeks when _SCHEDULING_FLAG_SEQUENTIAL is set"
+ This reverts commit 76647f2710d718e27f207b005956b7dba72c2d19.
+ Avoiding pull mode activation is a feature regression, and
+ demuxers should always use pull mode where that is possible,
+ e.g. if there's an upstream queue2 with a ring buffer or
+ a download buffer.
+ This patch made reverse playback no longer possible over http.
+ If the goal is to minimise seeks, then that can still be done
+ by making the demuxer behave differently in pull mode if
+ the SEQUENTIAL flag is set. If there are bugs, like the demuxer
+ needlessly scanning the entire file on start-up in pull mode,
+ then those should be fixed instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746010
+
+2015-05-19 19:48:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstpbutils.def:
+ win32: update .def file for new API
+
+2014-10-24 17:49:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: don't use soon-to-be-deprecated g_cancellable_reset()
+ From the API documentation: "Note that it is generally not
+ a good idea to reuse an existing cancellable for more
+ operations after it has been cancelled once, as this
+ function might tempt you to do. The recommended practice
+ is to drop the reference to a cancellable after cancelling
+ it, and let it die with the outstanding async operations.
+ You should create a fresh cancellable for further async
+ operations."
+ https://bugzilla.gnome.org/show_bug.cgi?id=739132
+
+2014-10-24 17:49:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/gio/gstgiobasesink.c:
+ * gst/gio/gstgiobasesrc.c:
+ gio: don't use soon-to-be-deprecated g_cancellable_reset()
+ From the API documentation: "Note that it is generally not
+ a good idea to reuse an existing cancellable for more
+ operations after it has been cancelled once, as this
+ function might tempt you to do. The recommended practice
+ is to drop the reference to a cancellable after cancelling
+ it, and let it die with the outstanding async operations.
+ You should create a fresh cancellable for further async
+ operations."
+ https://bugzilla.gnome.org/show_bug.cgi?id=739132
+
+2014-10-24 17:48:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/tcp/gstmultioutputsink.c:
+ * gst/tcp/gstmultisocketsink.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversrc.c:
+ tcp: don't use soon-to-be-deprecated g_cancellable_reset()
+ From the API documentation: "Note that it is generally not
+ a good idea to reuse an existing cancellable for more
+ operations after it has been cancelled once, as this
+ function might tempt you to do. The recommended practice
+ is to drop the reference to a cancellable after cancelling
+ it, and let it die with the outstanding async operations.
+ You should create a fresh cancellable for further async
+ operations."
+ https://bugzilla.gnome.org/show_bug.cgi?id=739132
+
+2015-05-19 18:53:09 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ gstdiscoverer: Add since annotation.
+ Forgot to add the since annotation to the
+ GstDiscovererSerializeFlags in the previous commit.
+
+2015-05-03 03:18:28 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ * tests/check/libs/discoverer.c:
+ * win32/common/libgstpbutils.def:
+ discoverer: Add serialization methods.
+ [API] gst_discoverer_info_to_variant
+ [API] gst_discoverer_info_from_variant
+ [API] GstDiscovererSerializeFlags
+ + Serializes as a GVariant
+ + Adds a test
+ + Does not serialize potential GstToc (s)
+ https://bugzilla.gnome.org/show_bug.cgi?id=748814
+
+2015-05-19 16:32:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Try harder to reuse previously configured caps values and give more preference to anything set as properties
+ This affects the pt, ssrc, seqnum-offset and timestamp-offset properties. If
+ they were set from a property, or we configured caps before, we try to use
+ that value for them. Even if the first structure of the downstream caps
+ specifies a different value, we check if the value is supported by other
+ structures.
+ Only if all this fails, we use the values given by downstream in the first
+ structure, i.e. if no properties were set and these are the first caps we
+ negotiate or downstream does not support our values.
+ By doing this we ensure that we don't spuriously change ssrcs or other fields
+ in the middle of the stream (and also consider property values more). Ssrc
+ changes would currently happen after sending an RTX packet (thus creating a
+ new internal source inside the rtpsession), and then renegotiating the
+ payloader (which then gets the RTX ssrc from rtpsession).
+ https://bugzilla.gnome.org/show_bug.cgi?id=749581
+
+2015-05-18 21:09:25 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video-scaler.c:
+ docs: a random set of trivial fixes for the library docs
+ Warnings down to 35, unused symbols doen to 112.
+
+2015-05-18 20:56:28 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ * gst-libs/gst/allocators/gstfdmemory.h:
+ docs: add fdmemory to docs
+
+2015-05-18 20:45:45 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/allocators/gstfdmemory.h:
+ * gst-libs/gst/video/colorbalance.h:
+ * gst-libs/gst/video/video-scaler.c:
+ docs: a random set of trivial fixes for the library docs
+ All those where super straight forward from the warnings gtkdoc prints. It kind
+ of makes sense to apply them before the list of warnings is >100 and people
+ complain that gtkdoc is noisy.
+
+2015-05-18 20:31:30 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/sdp/gstmikey.h:
+ mikey: fix a bunch of doc warnings
+ Rename header/source mismatch of parameters. Update the exposed API in
+ sections.txt.
+
+2015-05-18 20:01:49 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/playback/gstplaybin2.c:
+ Revert "doc: Workaround gtkdoc issue"
+ This reverts commit df7ef3c35d34352257a28307c07d4673f239452e.
+ This is fixed by the gtk-doc 1.23 release.
+
+2015-05-18 11:23:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ * tests/check/elements/appsrc.c:
+ appsrc: optimise caps changing when previously-set caps have not taken effect yet
+ Only negotiate/change caps once when setting caps twice and
+ the first-set caps have not been used yet.
+ Based on patch by Eunhae Choi.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747517
+
+2015-05-18 16:16:10 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix pool leak
+ During set caps when config fails, the referenced newpool
+ is not unref ed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749530
+
+2015-05-18 15:45:01 +0900 eunhae choi <eunhae1.choi@samsung.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: check the flags before set again
+ check the previous flags of playsink to avoid the reconfigure of playsink repeatedly
+ https://bugzilla.gnome.org/show_bug.cgi?id=749528
+
+2015-05-16 23:33:55 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ doc: Workaround gtkdoc issue
+ With gtkdoc 1.22, the XML generator fails when a itemizedlist is
+ followed by a refsect2. Workaround the issue by wrapping the refsect2
+ into para.
+
+2015-05-15 14:49:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstsubtitleoverlay.c:
+ playback: use the new gst_object api
+ Use gst_object_has_as_anchestor instead of the now deprecated _has_ancestor.
+
+2015-05-10 11:42:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/ogg/gstoggmux.c:
+ docs: fix up example pipeline
+
+2015-05-09 22:33:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisenc.c:
+ * ext/vorbis/gstvorbisparse.c:
+ * ext/vorbis/gstvorbistag.c:
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/gio/gstgiosink.c:
+ * gst/gio/gstgiosrc.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/videoconvert/gstvideoconvert.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/volume/gstvolume.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ docs: update element example pipelines
+ - gst-launch -> gst-launch-1.0
+ - use autoaudiosink and audiovideosink more often
+ - review pipeline examples and descriptions
+
+2015-05-10 10:51:09 +1000 Jan Schmidt <jan@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ video: Update win32 exports for new libgstvideo API
+
+2015-05-08 15:21:16 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/videoconvert/gstvideoconvert.c:
+ * gst/videoconvert/gstvideoconvert.h:
+ videoconvert: Expose some properties from the videoconverter API
+ Expose chroma resampler, alpha mode, alpha value, chroma mode, matrix mode,
+ gamma mode and primaries mode from the videoconverter API.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749105
+
+2015-05-08 14:57:03 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ * gst-libs/gst/video/video-resampler.h:
+ * gst/videoscale/gstvideoscale.c:
+ video-converter: Change some implicit string enums to real enums
+ GST_VIDEO_CONVERTER_OPT_ALPHA_MODE, GST_VIDEO_CONVERTER_OPT_CHROMA_MODE,
+ GST_VIDEO_CONVERTER_OPT_MATRIX_MODE, GST_VIDEO_CONVERTER_OPT_GAMMA_MODE and
+ GST_VIDEO_CONVERTER_OPT_PRIMARIES_MODE were G_TYPE_STRING with only a few valid
+ options. Changed those to real enums.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749104
+
+2015-05-08 15:06:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Also negotiate with downstream if needed before handling a GAP event
+
+2015-05-08 15:02:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Also negotiate with downstream if needed before handling a GAP event
+
+2015-05-06 12:40:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Try to be smarter when clipping buffers without duration/framerate to the segment
+ 2 second frame duration is rather unlikely... but if we don't clip
+ away buffers that far before the segment we can cause the pipeline to
+ lockup. This can happen if audio is properly clipped, and thus the
+ audio sink does not preroll yet but the video sink prerolls because
+ we already outputted a buffer here... and then queues run full.
+ In the worst case we will clip one buffer too many here now if no
+ framerate is given, no buffer duration is given and the actual
+ framerate is less than 0.5fps.
+ Fixes seeking on HLS/DASH streams, when seeking into the middle of
+ fragments and having no framerate/buffer duration.
+
+2015-05-04 17:59:30 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix navigation event leak when early returning
+ Create the event *after* the early return check so it's not leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748903
+
+2015-05-04 18:00:18 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix navigation event leak when not handled
+ gst_navigation_message_new_event() is *not* consuming the event so we should
+ always drop our extra reference.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748903
+
+2015-05-04 17:58:38 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst-libs/gst/video/navigation.c:
+ navigation: fix structure leak if subclass doesn't implement send_event()
+ The send_event() implementation is supposed to consume @structure.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748903
+
+2015-05-05 15:35:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Don't override segment.base from upstream with 0
+ Upstream might want to use it to properly map timestamps to running/stream
+ times, if we just override it with 0 synchronization will be just wrong.
+ For this we remove some old 0.10 code related to segment accumulation, and
+ remove some more code that is useless now, and accumulate the group start time
+ (aka segment.base offset) manually now.
+ https://bugzilla.gnome.org/show_bug.cgi?id=635701
+
+2015-05-05 13:14:12 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasedepayload: Add some debug output
+
+2015-03-19 10:50:22 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * docs/design/part-mediatype-video-raw.txt:
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-scaler.c:
+ video: add NV61 format support
+ https://bugzilla.gnome.org/show_bug.cgi?id=746466
+
+2015-05-04 20:33:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: add new video API to docs
+
+2015-05-04 10:35:55 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * ext/opus/gstopusheader.c:
+ opusheader: Do not include rate in caps if it is 0
+ As expressed in gst_opus_header_create_caps, value 0 means unset.
+ Setting rate value to 0 make negotiation with decoder fail.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748875
+
+2015-05-04 02:18:22 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-info.h:
+ video: check colorimetry and chroma_site equality in gst_video_info_is_equal()
+ Add VideoInfo accessors for colorimetry and chroma_site and use them
+ when checking the equality of two GstVideoInfo
+
+2015-05-04 02:10:17 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ * win32/common/libgstvideo.def:
+ video-color: Add gst_video_colorimetry_is_equal()
+ Add a function for comparing the equality of 2 colorimetry
+ structures.
+
+2015-04-10 16:05:45 +0900 Young Han Lee <y.lee@lge.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: remove unused code
+ These lines have done nothing for about 10 years.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748820
+
+2015-04-10 15:24:28 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst-libs/gst/pbutils/codec-utils.c:
+ pbutils: Use more strict profile checking for hevc
+ Use the profile_idc value to set the profile string in caps.
+ Don't use compatibility flags for this purpose.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747613
+
+2015-04-30 14:55:14 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Remove unused macro
+ Remove unused macro GET_TMP_LINE
+ https://bugzilla.gnome.org/show_bug.cgi?id=748687
+
+2015-04-29 15:44:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: add some more key navigation mappings
+ And don't feed multi-character key descriptors to the
+ event handler, it won't be what it expects.
+
+2015-04-29 15:30:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/navigation.c:
+ * gst-libs/gst/video/navigation.h:
+ * win32/common/libgstvideo.def:
+ navigation: sprinkle some since markers and add new API to .def file
+ https://bugzilla.gnome.org/show_bug.cgi?id=747245
+
+2015-04-02 16:16:58 +0200 Edward Hervey <edward@centricular.com>
+
+ * tools/gst-play.c:
+ tools: Add mouse/keyboard handling from messages
+ Allows the user to control playback with the window in focus
+ https://bugzilla.gnome.org/show_bug.cgi?id=747245
+
+2015-04-02 16:10:32 +0200 Edward Hervey <edward@centricular.com>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: Post unhandled navigation events on the bus
+ https://bugzilla.gnome.org/show_bug.cgi?id=747245
+
+2015-04-02 16:09:13 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/navigation.c:
+ * gst-libs/gst/video/navigation.h:
+ video: Add a new "event" navigation message type
+ This will be useful for elements that wish to post unhandled navigation
+ events on the bus to give the application a chance to do something with
+ it
+ https://bugzilla.gnome.org/show_bug.cgi?id=747245
+
+2015-04-28 17:24:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opus: fix includes and compilation against opus in non-standard prefix
+ https://bugzilla.gnome.org/show_bug.cgi?id=748594
+
+2015-04-28 16:58:21 +0200 Mersad Jelacic <mersad@axis.com>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opus: don't use deprecated gst_buffer_new_and_alloc
+ Use the helper function available in the base class instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748585
+
+2015-04-28 12:01:02 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-info.h:
+ * win32/common/libgstvideo.def:
+ video-info: expose InterlaceMode conversion to/from string
+ Expose the methods used to convert a GstVideoInterlaceMode to and
+ from a string.
+
+2015-04-27 11:26:10 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/encoding/gstsmartencoder.c:
+ Rename property enums from ARG_ to PROP_
+ Property enum items should be named PROP_ for consistency and readability.
+
+2015-04-27 11:06:58 +0200 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst/videoconvert/gstvideoconvert.c:
+ videoconvert: Keep colorimetry and chroma-site fields if passthrough
+ https://bugzilla.gnome.org/show_bug.cgi?id=748141
+
+2015-04-27 10:08:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiosink.h:
+ * gst-libs/gst/audio/gstaudiosrc.h:
+ audio: Change the remaining "samples" in the ::delay() vfunc docs to "frames"
+ https://bugzilla.gnome.org/show_bug.cgi?id=748289
+
+2015-04-26 20:13:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/pipelines/tcp.c:
+ tests: tcp: remove SOCK_CLOEXEC which causes build problems on OS/X
+ It's not needed here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747692
+
+2015-04-26 21:08:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.h:
+ * gst-libs/gst/audio/gstaudiosink.h:
+ * gst-libs/gst/audio/gstaudiosrc.h:
+ audio: The delay vfunc returns the number of frames, not samples
+ https://bugzilla.gnome.org/show_bug.cgi?id=748289
+
+2015-04-26 17:49:33 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Android.mk:
+ * android/NOTICE:
+ * android/alsa.mk:
+ * android/app.mk:
+ * android/app_plugin.mk:
+ * android/audio.mk:
+ * android/audioconvert.mk:
+ * android/audioresample.mk:
+ * android/audiotestsrc.mk:
+ * android/decodebin.mk:
+ * android/decodebin2.mk:
+ * android/gdp.mk:
+ * android/pbutils.mk:
+ * android/playbin.mk:
+ * android/queue2.mk:
+ * android/riff.mk:
+ * android/rtp.mk:
+ * android/rtsp.mk:
+ * android/sdp.mk:
+ * android/tag.mk:
+ * android/tcp.mk:
+ * android/typefindfunctions.mk:
+ * android/video.mk:
+ * android/videoconvert.mk:
+ * android/videoscale.mk:
+ * android/videotestsrc.mk:
+ * ext/ogg/Makefile.am:
+ * ext/vorbis/Makefile.am:
+ * 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:
+ * gst/adder/Makefile.am:
+ * gst/app/Makefile.am:
+ * gst/audioconvert/Makefile.am:
+ * gst/audiorate/Makefile.am:
+ * gst/audioresample/Makefile.am:
+ * gst/audiotestsrc/Makefile.am:
+ * gst/encoding/Makefile.am:
+ * gst/playback/Makefile.am:
+ * gst/tcp/Makefile.am:
+ * gst/typefind/Makefile.am:
+ * gst/videoconvert/Makefile.am:
+ * gst/videorate/Makefile.am:
+ * gst/videoscale/Makefile.am:
+ * gst/videotestsrc/Makefile.am:
+ * gst/volume/Makefile.am:
+ * tools/Makefile.am:
+ Remove obsolete Android build cruft
+ This is not needed any longer.
+
+2015-04-26 14:37:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/typefindfunctions.c:
+ tests: typefindfunctions: add test for UTF-16 MSS manifest typefinding
+
+2015-04-26 14:44:33 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: don't read more data than needed in MSS typefinder
+
+2015-04-26 14:27:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: detect MSS manifests without using g_convert()
+ Embedded systems often have limited charset conversion
+ functionality, so don't rely on g_convert() (i.e. iconv)
+ for UTF-16 to UTF-8 conversions, we can easily enough do
+ that ourselves by converting to native endianness and
+ then using GLib's helper functions.
+
+2015-04-25 18:45:50 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ * ext/libvisual/gstaudiovisualizer.h:
+ audiovisualizer: fix the license from GPL to LGPL
+ This was a copy'n'paste buf in the initial commit done by myself.
+
+2015-04-24 14:59:21 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ xmptag: fix invalid reads in GST_DEBUG statement
+ Don't try to print a string that is not NUL-terminated. This
+ log line does not really seem useful so let's just drop it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748413
+
+2015-04-24 17:10:59 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/encoding/gstencodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultihandlesink.c:
+ * gst/tcp/gstmultioutputsink.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ remove unused enum items PROP_LAST
+ This were probably added to the enums due to cargo cult programming and are
+ unused. Removing them.
+
+2015-04-03 00:44:12 +0900 Wonchul Lee <chul0812@gmail.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ audiodecoder: Add sink and src query virtual method
+ API: GstAudioDecoderClass::src_query()
+ API: GstAudioDecoderClass::sink_query()
+ https://bugzilla.gnome.org/show_bug.cgi?id=747293
+
+2015-04-23 15:57:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
+ Make sure the test environment is set up.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 15:42:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump automake requirement to 1.14 and autoconf to 2.69
+ This is only required for builds from git, people can still
+ build tarballs if they only have older autotools.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 15:14:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * tests/check/libs/.gitignore:
+ * tests/check/pipelines/.gitignore:
+ Update .gitignore
+
+2015-04-23 09:50:12 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: n_lines member should be a guint not a boolean
+ https://bugzilla.gnome.org/show_bug.cgi?id=748348
+
+2015-04-21 15:27:57 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix event leaks
+ gst_event_replace() takes its own reference on the event so we should drop
+ ours after creating and storing an event using it.
+ This fix leaks which can be reproduced using the
+ validate.http.media_check.vorbis_theora_1_ogg scenario.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748247
+
+2015-04-22 10:34:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * INSTALL:
+ Remove INSTALL file
+ autotools automatically generate this, and when using different versions
+ for autogen.sh there will always be changes to a file tracked by git.
+
+2015-04-22 10:33:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * LICENSE_readme:
+ Remove LICENSE_readme
+ It's completely outdated and just confusing, better if people are
+ forced to look at the actual code in question than trusting this file.
+
+2015-04-21 13:31:44 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix YUY2 scaling some more
+ Take into account the different steps between Y and UV when calculating
+ the line size for vertical resampling or else we might not resample
+ enough pixels and leave bad lines.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=747790
+
+2015-04-21 13:16:29 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: scale enough pixels in YUY2 (and friends) mode
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=747790
+
+2015-04-17 16:21:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * tests/check/libs/rtpbasedepayload.c:
+ tests: rtpbasedepayload: fix crash in test when passing varargs
+ Need to pass 64 bits where 64 bits are expected.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748027
+
+2015-04-17 11:18:22 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Remove unused variables
+ Remove unused variables n_taps, max_taps in setup_scale()
+ https://bugzilla.gnome.org/show_bug.cgi?id=748021
+
+2015-04-16 10:03:05 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideoutils.h:
+ video: add missing part of documentation text
+
+2015-03-31 13:26:21 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: fix GstToc leak when parsing toc messages
+ gst_message_parse_toc() returns a reffed GstToc which is owned by the
+ GstDiscovererInfo. But we have to make sure we unref its previous value before
+ setting the new one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747103
+
+2015-04-17 11:45:34 +0200 Edward Hervey <edward@centricular.com>
+
+ * win32/common/libgstallocators.def:
+ win32: Update defs for new API
+
+2015-04-17 09:31:40 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ * gst-libs/gst/allocators/gstfdmemory.h:
+ allocators: make GstFdAllocator non-abstract
+ Make the GstFdAllocator non-abstract because it is perfectly possible
+ to make memory from a generic fd. Mark the memory as simply "fd".
+
+2015-04-15 11:24:17 +0200 Bernhard Miller <bernhard.miller@streamunlimited.com>
+
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: fix mixed usage of gint and gint32 in int matrix
+ This is a fixup for b2db18cda2e4e7951655cb2a34108a8523b6eca9
+ audioconvert: avoid float calculations when mixing integer-formatted channels
+ The int matrix was using gint and gint32 synonymously, which can theoretically
+ cause problems if gint and gint32 are actually different types.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747005
+
+2015-04-14 12:47:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ * gst/gio/gstgio.c:
+ gio: fix gvfs plugin dependencies
+ Try harder to look for gvfs backend changes in the right
+ place, to make sure the plugin gets reloaded when backends
+ are removed or installed. We watch the gvfs mounts directory
+ because the files there contain absolute paths to the
+ backend executables, and those may not be in the usual gio
+ path.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747841
+
+2015-04-14 15:08:09 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/seek/scrubby.c:
+ examples: disconnect scale callback in scrubby
+ When the position slider's button is released, disconnect the "value_changed"
+ callback to avoid triggering false seek callbacks.
+
+2015-04-13 17:35:36 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/seek/scrubby.c:
+ examples: keep scrubby command consistent
+ scrubby has two options, wav and playbin. Wav takes a file location so make
+ the playbin option take a file location as well instead of an uri. This also
+ means the usage help string will be correct for the playbin option.
+
+2015-04-13 17:28:45 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/seek/scrubby.c:
+ examples: no need to set intermediate states
+
+2015-04-13 16:09:26 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/seek/scrubby.c:
+ examples: wavparse doesn't need dynamic linking
+ In scrubby, there is no need to link wavparse with the sink dynamically.
+ The pad is available when the element is generated.
+ Change video and audio sinks to the automatically detected sinks.
+
+2015-04-11 19:51:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Break instead of return if default negotiation on GAP events fails
+ Otherwise we're going to leak the event.
+
+2015-04-11 00:03:29 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/app/Makefile.am:
+ * gst/videorate/Makefile.am:
+ app, videorate: fix CFLAGS and LIBADD order
+ Make sure local headers are included before installed -base.
+
+2015-04-10 14:30:36 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/playrec/playrec.c:
+ examples: remove reference to 0.10 in playrec
+
+2015-04-10 13:41:39 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/overlay/gtk-videooverlay.c:
+ examples: remove deprecated function in gtk-videooverlay
+ gtk_widget_set_double_buffered () has been deprecated since GTK 3.14.
+ Also, widgets are realized automatically and gtk_wiget_realize () is only
+ meant to be used in widget implementations.
+
+2015-04-09 17:03:11 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: fix buffer leak in chain function
+ If we don't consume the buffer by passing its reference to
+ overlay->text_buffer then we need to unref it.
+ Fix a leak with validate.file.playback.fast_forward.test5_mkv
+ when running inside Valgrind.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747602
+
+2015-04-08 18:32:29 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: docs grammar fixes
+ https://bugzilla.gnome.org/show_bug.cgi?id=747516
+
+2015-04-09 16:49:44 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/gio/giosrc-mounting.c:
+ examples: add example description to giosrc-mounting
+ Also, use GST_MESSAGE_TYPE instead of accessing the GstMessage structure
+
+2015-04-09 13:00:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: fix ring buffer leak on open failure
+
+2015-04-09 12:59:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobasesrc: fix ring buffer leak on open failure
+
+2015-04-09 11:23:25 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/encoding/encoding.c:
+ examples: reuse variables in encoding example
+
+2015-04-08 20:49:24 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Don't post error messages while holding the stream lock
+
+2015-04-08 20:48:39 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Don't get and parse the current srcpad caps
+ We only get here if we don't have any srcpad caps, and we're going
+ to override the GstAudioInfo a few lines below anyway without ever
+ using it if for whatever reason we get caps here.
+
+2015-04-08 20:45:58 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Try to invent default caps instead of setting none at all when getting a GAP event before CAPS
+ Otherwise we would forward the GAP event without ever providing any caps,
+ which then would make decodebin expose a srcpad without any caps set. That's
+ confusing for applications and can lead to all kinds of interesting bugs.
+ Instead do the same as already is done in GstAudioDecoder, and try to invent
+ caps based on the sinkpad caps and the caps allowed by downstream and the
+ srcpad template caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747190
+
+2015-04-08 20:44:15 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Also log the pointer value of sticky events in debug output
+ Makes it easier to follow them in the debug logs.
+
+2015-04-08 17:12:22 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/dynamic/addstream.c:
+ examples: remove unused return value in addstream
+ Removing unused return value of pause_play_stream ().
+ Fixing code style to satisfy the git hook.
+
+2015-04-08 15:31:39 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/dynamic/sprinkle.c:
+ examples: avoid sprinkle running endlessly
+ Quit sprinkle when there are no more frequencies to remove.
+ Also rename for readability the check for linking elements.
+
+2015-04-08 16:15:43 +0200 Edward Hervey <edward@centricular.com>
+
+ * common:
+ * tests/check/Makefile.am:
+ tests: Use AM_TESTS_ENVIRONMENT
+ Needed by the new automake test runner
+
+2015-04-07 16:43:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ rtp: rtcpbuffer: fix typo in enum
+ and in docs. Spotted by Rob Swain.
+
+2015-04-07 15:32:35 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/app/appsink-src2.c:
+ tests: remove unused filename string from appsink-src2
+
+2015-04-07 15:30:30 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/app/appsink-src.c:
+ tests: check file exists before running appsink-src
+
+2015-04-07 15:16:41 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/app/appsink-src.c:
+ * tests/examples/app/appsink-src2.c:
+ * tests/examples/app/appsrc_ex.c:
+ tests: add missing license headers for example apps
+
+2015-04-06 19:20:00 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ {audio,video}decoder: Forward SEGMENT_DONE events immediately and drain decoders
+ Otherwise we're going to wait with draining until the next data comes, which
+ is a bit suboptimal and might take a long time... or maybe never happens.
+
+2015-04-05 13:53:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/appsrc.c:
+ tests: appsrc: clean up block_deadlock test and make it work in valgrind
+ Remove all the bus watch and main loop code from the block_deadlock
+ test, it's not needed: neither pipeline will ever post an EOS or ERROR
+ message on the bus, and we're the only ones posting an error, from a
+ timeout. Might just as well just sleep for a bit and then do whatever
+ we want to do.
+ Don't gratuitiously set tcase timeout, just use whatever is the
+ default (or set via the environment).
+ Make individual pipeline runs shorter.
+ Check for valgrind and only do a handful iterations when running
+ in valgrind, not 100 (each iteration takes about 4s on a core i7).
+ Make videotestsrc output smaller buffers than the default resolution,
+ we don't care about the buffer contents here anyway.
+ Fixes test timeouts when run in valgrind.
+
+2015-04-05 12:30:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/multisocketsink.c:
+ tests: multisocketsink: fix flaky unit test
+ On slower systems, or under high system load (e.g. check-valgrind),
+ the sending_buffers_with_9_gstmemories test would sometimes fail,
+ because the read call only returns 32 bytes instead of the full
+ 36 bytes expected. This is because multisocketsink might end up
+ doing a partial write of 32 bytes first, and then write the
+ missing 4 bytes later, but since we don't wait for all of data
+ to be written, there's a short window where our read call in the
+ unit test might then only receive the 32 bytes written so far,
+ which makes it deeply unhappy.
+ Instead, make sure we loop to read all bytes.
+
+2015-04-04 21:38:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/tcp/gstmultisocketsink.c:
+ tcpserversink: don't error out if clients send us something, just ignore it
+ We don't expect clients to send us any data, but if they do, just
+ ignore it. Web browsers might send us an HTTP request for example,
+ but some will still be happy if we just send them data without
+ a proper HTTP response.
+ There was a bug in the reading code path. We only have a small
+ read buffer and would provoke an EWOULDBLOCK trying to read
+ because we don't bail out of the loop early enough.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743834
+
+2015-04-04 01:23:48 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/pipelines/basetime.c:
+ tests: basetime: fix timeouts when running under valgrind
+ This test sets a rather short timeout, increase this when
+ we run under valgrind. Also add a short sleep to the
+ fakesrc ! fakesink pipeline to avoid thrashing the CPU,
+ which would often not stop the main loop when it should.
+ Also fix wrong (0.10) return value from pad probe callback.
+
+2015-04-04 00:46:46 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: downgrade left-over ERROR debug message
+
+2015-04-04 00:42:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ * tests/check/elements/videorate.c:
+ videorate: fix a couple of memory leaks
+ tests: videorate: fix leak in unit test
+
+2015-04-03 18:18:32 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ doc: Add gst_video_encoder_get_allocator() to doc
+
+2015-04-03 21:00:53 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exiftag: don't try to convert utf-8 to latin1 if string is ASCII already
+ Bypass g_convert/iconv if there's nothing to convert. That way,
+ conversion won't fail on systems where iconv doesn't support
+ converting utf-8 to latin1 and there's nothing to convert.
+ https://bugzilla.gnome.org/show_bug.cgi?id=723252
+
+2015-04-03 18:57:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From bc76a8b to c8fb372
+
+2015-03-12 16:01:48 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: fix wrong duration on partial streams with a skeleton index
+ When a stream has a skeleton index, the stream time is taken from that
+ index. However, when part of the stream is captured, the index is
+ invalid as its offsets are now wrong. To avoid this, we ignore the index
+ when the last offset points beyond the end of the stream (when its
+ byte length is known).
+ https://bugzilla.gnome.org/show_bug.cgi?id=744070
+
+2015-03-18 16:32:53 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/pango/gstbasetextoverlay.c:
+ textoverlay: fix disappearing text with high deltax
+ When deltax is large enough to cause the text to push past the
+ width of the frame, it would disappear due to a bug in setting
+ the layout width.
+ While there, fix a log printing an incorrect width to set.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739689
+
+2014-12-17 12:17:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: fix deadlock when not pulling a buffer from collectpads
+ oggmux keeps a cached buffer per pad, and pulls buffers from
+ collectpads to this cached buffer for all pads before processing
+ the best pad. In some cases, the move from collectpads buffer
+ to cached buffer is delayed till next call. However, when there
+ is only one pad, this can't be delayed till next call as there
+ will be a deadlock since collectpads has no other pad to push to.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740565
+
+2015-03-25 15:36:38 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: fix deadlock on chain shutdown
+ When shutting down the chain, we can get a deadlock when removing
+ a pad, if that chain was being busy streaming but blocked (eg, while
+ waiting for a queue to have free space).
+ https://bugzilla.gnome.org/show_bug.cgi?id=746480
+
+2015-04-03 13:20:58 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/seek/scrubby.c:
+ examples: add license header to scrubby
+
+2015-03-19 10:48:15 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ audio,video: use gst_segment_is_equal instead of memcmp
+ memcmp will blindly compare the reserved fields, as well as any
+ padding the compiler may choose to sprinkle in GstSegment.
+ Fixes valgrind complaints in unit tests, as well as some found via
+ https://bugzilla.gnome.org/show_bug.cgi?id=738216
+
+2014-04-04 12:32:14 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * sys/xvimage/xvimageallocator.c:
+ xvimagsink: fix failure to allocate large shared memory blocks
+ A previous patch increased allocations by 15 bytes in order to ensure
+ 16 byte alignment for g_malloc blocks. However, shared memory is
+ already block aligned, and this extra 15 bytes caused allocation
+ to fail when we were already allocating to the shared memory limit,
+ which is a lot smaller than typical available RAM.
+ Fix this by removing the alignment slack when allocating shared
+ memory.
+ https://bugzilla.gnome.org/show_bug.cgi?id=706066
+
+2014-04-04 12:40:14 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * sys/ximage/ximagepool.c:
+ ximage: do not allocate extra alignment slack for shared memory
+ A previous patch increased allocations by 15 bytes in order to ensure
+ 16 byte alignment for g_malloc blocks. However, shared memory is
+ already block aligned, and this extra 15 bytes is not needed. Since
+ shared memory limits are low compared to RAM, we remove this waste.
+ https://bugzilla.gnome.org/show_bug.cgi?id=727236
+
+2015-04-03 13:56:28 +0900 Chihyoung Kim <chihyoung2.kim@lge.com>
+
+ * configure.ac:
+ tests: require Gtk+ 3.10 for examples
+ Fixes build of playback and seek tests when an
+ older Gtk+ version is present on the system.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747283
+
+2015-04-03 11:46:12 +0530 Arun Raghavan <arun@centricular.com>
+
+ * ext/opus/gstopusenc.c:
+ opus: Fix incorrect fall-through condition in property getter
+
+2014-12-09 13:18:42 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/videorate/gstvideorate.c:
+ * gst/videorate/gstvideorate.h:
+ * tests/check/elements/videorate.c:
+ videorate: Detect framerate if not forced to variable downstream
+ In case upstream does not provide videorate with framerate information,
+ it will detect the current framerate from the buffer it received,
+ but if downstream forces the use of variable framerate (most probably
+ through the use of a caps filter with framerate = 0 / 1), videorate will
+ respect that.
+ And add some unit tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=734424
+
+2014-12-09 11:31:30 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Do not loop forever pushing first buffer when variable framerate
+ In the case the framerate is variable (represented by framerate=0/1),
+ we currently end up loop pushing the first buffer and then recompute
+ diff1 and diff2 without updating the videorate->next_ts at all
+ leading to infinitely looping pushing that first buffer.
+ In the case of variable framerate, we should just compute the next_ts
+ as previous_pts + previous_duration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734424
+
+2015-04-02 14:32:15 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/playback/playback-test.c:
+ playback-test: update deprecated API
+
+2015-04-02 11:33:12 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/icles/test-colorkey.c:
+ * tests/icles/test-videooverlay.c:
+ tests: fix deprecated API in colorkey and videooverlay
+
+2015-04-02 11:14:08 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/seek/scrubby.c:
+ examples: fix deprecated API in scrubby
+
+2015-03-19 14:34:07 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: don't use GST_ERROR() for debug messages
+ Fix https://bugzilla.gnome.org/show_bug.cgi?id=746457
+
+2015-04-01 15:58:28 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/audio/volume.c:
+ tests: use elapsed label of volume example
+
+2015-03-30 11:24:46 +0200 Bernhard Miller <bernhard.miller@streamunlimited.com>
+
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: avoid float calculations when mixing integer-formatted channels
+ The patch calculates a second channel mixing matrix from the current one. The
+ matrix contains the original values * (2^10) as integers. This matrix is used
+ when integer-formatted channels are mixed.
+ On a ARM Cortex-A8, single core, 800MHz this improves performance in a
+ testcase from 29s to 9s for downmixing 6 channels to stereo.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747005
+
+2015-04-01 15:02:13 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/audio/volume.c:
+ tests: fix deprecated API in audio volume example
+
+2015-04-01 14:37:23 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/seek/jsseek.c:
+ jsseek: update deprecated GTK API
+
+2015-04-01 13:50:51 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/seek/jsseek.c:
+ jsseek: switch deprecated GtkTable for GtkGrid
+
+2015-04-01 11:01:57 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/audio/audiomix.c:
+ tests: update deprecated GTK API in audiomix
+
+2015-03-31 11:21:25 +0200 Edward Hervey <bilboed@bilboed.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:
+ introspection: Don't use g-ir-scanner cache at compile time
+ It pollutes user directories and we don't need to cache it
+ https://bugzilla.gnome.org/show_bug.cgi?id=747095
+
+2014-04-10 12:03:05 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/tag/id3v2frames.c:
+ id3v2: ignore RVA2 tags with more than 64 peak bits
+ The spec for this does not say nor imply how this should be
+ interpreted. The previous code would try to shift by 64 bits,
+ which is undefined.
+ Coverity 1195119
+ https://bugzilla.gnome.org/show_bug.cgi?id=727955
+
+2015-03-30 10:50:45 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: avoid possible deference of null pointer
+ For safety, check the pointer playbin->curr_group is valid before
+ reading parameters of the structure.
+ CID #1291624
+
+2015-03-28 16:59:23 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: resurrect some flow return handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=744572
+
+2015-03-27 20:16:28 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: handle a sample not having caps or a buffer more gracefully
+ https://bugzilla.gnome.org/show_bug.cgi?id=746908
+
+2015-03-27 16:22:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ * tests/check/libs/rtpbasedepayload.c:
+ basedepay: Handle initial gaps and no clock-base
+ When generating segment, we can't assume the first buffer is actually
+ the first expected one. If it's not, we need to adjust the segment to
+ start a bit before.
+ Additionally, we if don't know when the stream is suppose to have
+ started (no clock-base in caps), it means we need to keep everything in
+ running time and only rely on jitterbuffer to synchronize.
+ https://bugzilla.gnome.org/show_bug.cgi?id=635701
+
+2015-03-26 23:53:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: improve debug message by printing the object
+ Print the pad object that EOS'd too early
+
+2015-03-27 13:39:43 +0800 Song Bing <b06498@freescale.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Keep sticky events around when doing a soft reset
+ The current code will first discard all frames, and then tries to copy
+ all sticky events from the (now discarded) frames. Let's change the order.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746865
+
+2015-03-26 18:03:12 -0700 David Schleef <ds@schleef.org>
+
+ * gst-libs/gst/riff/riff-ids.h:
+ riff: Add FLLR tag
+
+2015-03-25 18:40:25 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ * tests/check/libs/rtpbasedepayload.c:
+ basedepayload: Fix generated segment
+ This fixes playback position in RTSP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=635701
+
+2015-03-25 08:20:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: ignore new pads if it is shutting down
+ If a new pad is added after playbin has been put to READY/NULL it
+ should ignore new pads as it is shutting down.
+ This can happen when the pipeline fails to preroll (is still in READY)
+ and the user gives up on waiting or an error that doesn't reach
+ the demuxer occurs (on some event handling) and it will continue to
+ work and exposing pads while playbin has been put to NULL.
+ Without this check an input-selector is created and set to PAUSED
+ state, preventing playbin from properly shutting down in case it
+ has data blocked inside it.
+
+2015-03-24 15:47:20 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/theora/gsttheoradec.c:
+ Revert "theoradec: Disable usage of crop meta"
+ This reverts commit da52868f468bd75ddb595a3eb52aaa38ecbbac41.
+
+2015-03-24 15:18:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Don't leak the pools
+ gst_query_set_nth_alloction_pool() is transfer none on the pool, so we must
+ unref the pool when done.
+
+2015-03-01 11:44:22 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/theora/gsttheoradec.c:
+ theoradec: Disable usage of crop meta
+ This is a temporary workaround that simply disables usage of crop
+ meta for now.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741030
+
+2015-03-24 17:28:51 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * gst/audioconvert/gstaudioquantize.c:
+ audioconvert: Eliminate unsigned quantizers
+ audio_convert_convert unpacks to default format (signed) before calling
+ quantize, and the unsigned variants were equivalent to signed anyway,
+ so we just get rid of them.
+
+2015-03-24 03:01:22 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * gst/audioconvert/gstaudioquantize.c:
+ * gst/audioconvert/gstfastrandom.h:
+ audioconvert: Avoid int division in quantization
+ Since range size is always 2^n, we can simply use modulo (implemented
+ with a bitmask).
+ The previous implementation used 64-bit integer division, which is
+ done in software on ARMv7. Although the divisor was constant, the
+ division could not be transformed into "multiplication by magic number"
+ since the dividend was 64-bit.
+ The now-unused and not-so-fast gst_fast_random_(u)int32_range functions
+ were removed.
+ Also, implementing bug fixes:
+ 1) ADD_DITHER_TPDF_HF_I no longer discards bias.
+ 2) We change TPDF's noise range to be the same as RPDF's. Previously,
+ RPDF's noise ranged:
+ { bias - dither, bias + dither }
+ while TPDF's noise ranged:
+ { bias/2 - dither/2, bias/2 + dither/2 - 1 } +
+ { bias/2 - dither/2, bias/2 + dither/2 - 1 } =
+ { bias - dither, bias + dither - 2 }
+ Now, both range:
+ { bias - dither, bias + dither - 1 }
+ https://bugzilla.gnome.org/show_bug.cgi?id=746661
+
+2015-03-24 15:13:52 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fall through switch statement
+ Adding a comment makes coverity happy and quells the issue.
+ CID 1291629
+
+2015-02-16 09:25:03 +1000 Duncan Palmer <dpalmer@digisoft.tv>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Set multiqueue sizes before use-buffering.
+ This fixes a race where the use-buffering property on a multiqueue was
+ set before the queue depth was changed from it's high preroll limits to
+ lower playback limits. This resulted in buffering messages being emitted
+ by the multiqueue in the short window between use-buffering being
+ set and the queue depth being reset.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744308
+
+2015-03-24 10:46:44 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ Revert "fdmemory: freed pointer will always be 0"
+ This reverts commit 7fbcefb753f944a79eae6957ea2789c960eb9eea.
+
+2015-03-24 10:19:05 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ fdmemory: freed pointer will always be 0
+
+2015-03-23 13:15:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: Set output format immediately after creating the encoder instance
+ We know the caps by then, there's no need to wait until we actually receive
+ the first buffer.
+
+2015-03-23 13:13:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opusenc: Remove another unused variable
+
+2015-03-23 13:11:42 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ * ext/opus/gstopusheader.c:
+ opusenc: Remove useless headers and header_sent variables from the instance struct
+ They are only used inside a single function.
+
+2015-03-23 12:09:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: Take channels and sample rate from the caps if we have no stream header
+
+2015-03-23 12:07:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: Reset the decoder if the caps change
+
+2015-03-23 11:57:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: Take output sample rate from the stream headers too
+ This way we let opusdec do the resampling if needed and don't carry
+ around buffers with a too high sample rate if not required.
+ While Opus always uses 48kHz internally, this information from the
+ header specifies which frequencies are safe to drop.
+
+2015-03-23 11:56:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusheader.c:
+ opusheader: Put number of channels and sample rate into the caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-03-20 17:45:03 +0900 Wonchul Lee <chul0812@gmail.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix compiler warning
+ gstoggdemux.c:1233:11: error: format specifies type 'long' but the argument has type 'ogg_int64_t' (aka 'long long') [-Werror,-Wformat]
+ granule);
+ ^~~~~~~
+ https://bugzilla.gnome.org/show_bug.cgi?id=746512
+
+2015-03-19 13:31:07 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * win32/common/libgstallocators.def:
+ defs: update
+
+2015-03-19 12:42:23 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-convert: fix clamping for 16 bits alpha mult
+
+2015-03-18 20:38:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: fix height/width assertions
+ As commit 274984e8 states:
+ When doing CROP META it is expected that the width and/or height
+ in the GstVideoMeta is bigger or equal to the caps negotiated size.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741030
+
+2015-03-18 15:12:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/allocators/Makefile.am:
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ * gst-libs/gst/allocators/gstfdmemory.h:
+ fdmemory: make a base class for allocating fd-backed memory
+ Make a base class that can help with allocating fd-backed memory.
+ Make dmabuf extend from the base class.
+ We can now make methods to check if memory has an fd and get the fd for
+ all the different types of fd-backed memory.
+
+2015-03-16 20:41:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/multisocketsink.c:
+ multisocketsink: Allocate enough memory on the stack in the test
+ Otherwise we just overwrite other things on the stack and cause crashes.
+
+2015-03-16 11:53:24 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix playback regression on streams with clipped data at start
+ The code that was calculating the start granule from packet durations
+ was interpreting a negative value as an error, but this is actually a
+ valid case, to indicate clipping of data at start.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743900
+
+2015-03-15 17:27:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ * gst-libs/gst/allocators/gstfdmemory.h:
+ fdmemory: add flags to control behaviour
+ Add some flags to the GstFdMemory to control how memory is mapped and
+ unmapped.
+
+2015-03-15 16:41:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/allocators.c:
+ allocators: add allocators test
+
+2015-03-15 15:16:23 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/allocators/Makefile.am:
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ * gst-libs/gst/allocators/gstfdmemory.c:
+ * gst-libs/gst/allocators/gstfdmemory.h:
+ fdmemory: add fd backed GstMemory to separate file
+ Make a separate file for the code to handle the fd backed memory.
+ This would make it possible later to add other allocators also using
+ fd backed memory.
+
+2015-03-14 18:08:15 +0000 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: fix deadlock condition
+ The variables could have changed when the lock was released
+ to push a gap event. Streamsynchronizer needs to check them
+ again before going to sleep.
+ Bonus: fix a comment typo
+
+2015-03-13 18:07:12 +0000 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playsink: remove redundant else statements
+
+2015-03-13 18:23:46 +0000 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: don't escape percent sign in documentation code sample
+
+2014-11-03 12:47:18 +0000 William Manley <will@williammanley.net>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/tcp.c:
+ Add test_that_multisocketsink_and_socketsrc_preserve_meta
+ This test is in a seperate commit to the previous two because it depends
+ on and tests the functionality in both.
+
+2015-03-13 16:19:28 +0000 William Manley <will@williammanley.net>
+
+ * gst/tcp/gstsocketsrc.c:
+ socketsrc: Add support for GstNetControlMessageMeta
+ multisocketsink now understands the new GstNetControlMessageMeta to allow
+ sending control messages (ancillary data) with data when writing to Unix
+ domain sockets.
+ Thanks to glib's `GSocketControlMessage` abstraction the code introduced
+ in this commit is entirely portable and doesn't introduce and additional
+ dependencies or conditionally compiled code, even if it is unlikely to be
+ of much use on non-UNIX systems.
+
+2014-10-30 17:53:15 +0000 William Manley <will@williammanley.net>
+
+ * configure.ac:
+ * gst/tcp/gstmultisocketsink.c:
+ multisocketsink: Add support for GstNetControlMessageMeta
+ multisocketsink now understands the new GstNetControlMessageMeta to allow
+ sending control messages (ancillary data) with data when writing to Unix
+ domain sockets.
+ A later commit will introduce a new socketsrc element which will similarly
+ understand `GstNetControlMessageMeta`. This, when used with a
+ `GSocketControlMessage` of type `GUnixFDMessage` will allow GStreamer to
+ send and receive file-descriptions in ancillary data, the first step to
+ using memfds to implement zero-copy video IPC.
+ Thanks to glib's `GSocketControlMessage` abstraction the code introduced
+ in this commit is entirely portable and doesn't introduce and additional
+ dependencies or conditionally compiled code, even if it is unlikely to be
+ of much use on non-UNIX systems.
+
+2015-03-13 13:56:13 +0000 William Manley <will@williammanley.net>
+
+ * gst/tcp/gstsocketsrc.c:
+ * gst/tcp/gstsocketsrc.h:
+ * tests/check/pipelines/tcp.c:
+ socketsrc: Add `connection-closed-by-peer` signal
+ This provides notification that the socket in use was closed by the peer
+ and gives an opportunity to replace it with a new one which is not
+ closed, allowing reading from many sockets in order.
+ I use this in pulsevideo to implement reconnection logic to handle the
+ pulsevideo service dieing, such that is can be restarted without
+ disrupting downstream.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=739546
+
+2015-03-13 13:43:59 +0000 William Manley <will@williammanley.net>
+
+ * gst/tcp/gstsocketsrc.c:
+ socketsrc: Tidy up usage of `g_object_unref`/`g_clear_object` and locking
+ This is clearer, and should make future changes safer. No functional
+ change intended.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=739546
+
+2015-03-13 13:30:48 +0000 William Manley <will@williammanley.net>
+
+ * gst/tcp/gstsocketsrc.c:
+ socketsrc: Refactor to simplify
+ * Don't bother polling, just do a blocking read, the `GCancellable` will
+ take care of unlocking. This should also be faster on MS Windows where
+ the GIO documentation for `g_socket_get_available_bytes` states: "Note
+ that on Windows, this function is rather inefficient in the UDP case".
+ * Implement `GstPushSrc.fill` rather than `GstPushSrc.create`. This means
+ that we will be using the downstream allocator which may be more
+ efficient. It also means that socketsrc is likely to respect its
+ "blocksize" property (assuming that there is enough data available).
+ See https://bugzilla.gnome.org/show_bug.cgi?id=739546
+
+2014-11-03 02:47:14 +0000 William Manley <will@williammanley.net>
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * gst/tcp/Makefile.am:
+ * gst/tcp/gstsocketsrc.c:
+ * gst/tcp/gstsocketsrc.h:
+ * gst/tcp/gsttcpplugin.c:
+ * tests/check/pipelines/tcp.c:
+ * win32/vs7/libgsttcp.vcproj:
+ * win32/vs8/libgsttcp.vcproj:
+ tcp: Add element socketsrc
+ `socketsrc` can be considered a source counterpart to `multisocketsink`.
+ It can be considered a generalization of `tcpclientsrc` and
+ `tcpserversrc`: it contains all the logic required to communicate over
+ the socket but none of the logic for creating the sockets/establishing
+ the connection in the first place, allowing the user to accomplish this
+ externally in whatever manner they wish making it applicable to other
+ types of sockets besides TCP.
+ This commit essentially copies the implementation directly from
+ tcpserversrc. Later patches will tidy the implementation up and
+ re-implement `tcpclientsrc` and `tcpserversrc` in terms of `socketsrc`.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=739546
+
+2015-03-13 23:24:23 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: Log with the ringbuffer object where possible
+
+2015-03-13 12:49:31 +0000 William Manley <will@williammanley.net>
+
+ * gst/tcp/gstmultisocketsink.c:
+ * tests/check/elements/multisocketsink.c:
+ multisocketsink: Map `GstMemory`s individually when sending
+ If a buffer is made up of non-contiguous `GstMemory`s `gst_buffer_map`
+ has to copy all the data into a new `GstMemory` which is contiguous. By
+ mapping all the `GstMemory`s individually and then using scatter-gather
+ IO we avoid this situation.
+ This is a preparatory step for adding support to multisocketsink for
+ sending file descriptors, where a GstBuffer may be made up of several
+ `GstMemory`s, some of which are backed by a memfd or file, but I think this
+ patch is valid and useful on its own.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=746150
+
+2015-03-13 10:30:43 +0000 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: Relax width/height assertion
+ When doing CROP META it is exepcted that the width and/or height in the
+ GstVideoMeta is bigger or equal to the caps negotiated size.
+
+2015-03-12 16:32:31 +0000 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/gstvideopool.c:
+ videopool: Choose the biggest buffer size
+ We should respect what has been negotiated.
+
+2015-03-12 10:06:15 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: recover from EOS when searching for chain in push mode
+ If we get EOS when we're trying to build a chain, we disable seeking
+ and continue instead of posting an error. This can happen for corner
+ cases such as a stream with a video that stops before the end, for
+ instance.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745980
+
+2015-03-11 16:46:38 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix seeking in files with a "missing" stream
+ When looking for pages when seeking, we stop looking for non sparse
+ streams if we don't find one within a given threshold. This fixes
+ seeking filling up queues and blocking in corner cases such as an
+ audio file with a pathological 1 frame video stream (yes, I saw one).
+ https://bugzilla.gnome.org/show_bug.cgi?id=745980
+
+2015-03-13 01:06:57 +1100 Jan Schmidt <jan@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/video-chroma.c:
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-dither.c:
+ * gst-libs/gst/video/video-resampler.c:
+ * gst-libs/gst/video/video-resampler.h:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst/videoscale/gstvideoscale.h:
+ docs: Add new video functions and objects. Cleanup a little.
+ Add GstVideoChroma, GstVideoDither, GstVideoScaler and friends to the docs.
+ Remove and clean up a few obsolete/deleted refs and typos
+
+2015-03-12 12:49:40 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opusenc: replace cbr and constrained-vbr properties with an enum
+ It was deemed confusing before.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744909
+
+2015-03-12 12:17:11 +0000 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Disconnect signals and invalidate group if it fails to activate
+ Otherwise playbin might move to the group directly after EOS of the next
+ group, and then error out again.
+
+2015-02-01 03:39:07 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoradec.h:
+ theoradec: Fix decoding in the presence of GstVideoCropMeta
+ Store the video info of the internal frame decode width/height
+ separate to the exposed (cropped) frame info, so that it can be
+ used for mapping the downstream allocated video frame buffer correctly
+ when using GstVideoCropMeta.
+ Fixes playback of files with sizes that aren't a multiple of 16-pixels
+ width or height.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741030
+
+2015-03-03 15:18:04 +0800 Song Bing <b06498@freescale.com>
+
+ * tests/check/pipelines/streamsynchronizer.c:
+ streamsynchronizer: Should wait state change complete before start another state change
+ Should wait state change complete before start another state change.
+ Can't ensure can received async-done message when state change from PLAYING to PAUSED.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-02-27 16:40:23 +0800 Song Bing <b06498@freescale.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Remove unnecessary ERROR message.
+ Remove unnecessary ERROR message.
+ Push GAP will fail as flushing. Needn't ERROR message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-03-05 17:42:53 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: do not send seek events from the streaming thread
+ This will usually deadlock, despite this patch being in master for
+ quite some time and working fine. Nevertheless, we deem it to be
+ not working, disregarding facts.
+ As such, we fix it by keeping track of seek events, and sending
+ them upstream from a separate thread. Buffers are then discarded
+ till we get a new segment with the expected seqnum.
+
+2015-02-23 13:07:41 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: set correct seqnum on segment events after a seek in push mode
+ There is already a seqnum field for this, which was used to overwrite
+ the seqnum that was set by the push specific code.
+
+2015-02-23 11:30:36 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: try harder to query duration from upstream
+ READY->PAUSED can be too early as souphttpsrc can get the HTTP
+ headers after this. Try again in the chain function.
+ Also use seeking query to disable seeking if upstream reports
+ being unseekable.
+
+2014-10-31 10:55:14 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: add non flushing time seeking in push mode
+ Some resetting code has to be done in the NEW_SEGMENT
+ event handler, instead of the missing FLUSH_STOP one.
+ Segment base was also wrongly accounted for. This was hidden
+ by the fact that flushing resets the base.
+ A discontinuity is now also signalled on seeking. We have to
+ also ensure that the discontinuity "sticks" till a buffer
+ with a valid timestamp goes out, or the audio decoder base
+ class will ignore the discontinuity for purposes of keeping
+ track of the current time.
+ This allows using non flushing segment seeks for looping
+ HTML audio in particular, and more generally non flushing seeks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729198
+
+2015-02-04 17:13:44 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix wrong first granule
+ The code was using the first nonnegative granulepos to seed the
+ granule tracking, which appeared to work since headers have zero
+ granulepos. However, this does not work for files with a hole at
+ start, which are common in live streaming.
+ The correct behavior is to look for the first granule, and subtract
+ the duration of all the packets finishing on this page.
+ The function which does this relies on the fact that the ogg_stream
+ structure can be duplicated by shallow copy, in order to pull the
+ packets from the first page(s) on the copy without affecting the
+ original stream state.
+
+2015-03-11 09:48:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: fix border handling of YUY2 and friends
+ Don't draw the border in groups of 4 pixels for YUY2 but instead in
+ groups of 2 with alternating U and V. This avoids a crash on odd width
+ borders.
+
+2015-03-11 09:47:23 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: force yuv conversion for border
+ Make sure we always do yuv conversion for the border.
+
+2015-03-10 17:29:51 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: fix A422 subsampling description
+
+2015-03-10 15:12:30 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add table based matrix8 implementation
+ Based on patch from Mozzhuhin Andrey <nopscmn at gmail.com>
+ Add a table based matrix8 multiplication implementation. The algorithm
+ does not do any clipping so we need to make sure we never call this on
+ input that might need to be clipped. In general, this algorithm is
+ 2 times faster than the orc optimized one and would be chosen for all
+ RGB -> YUV conversions and some YUV->YUV and RGB->RGB conversions.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732186
+
+2015-03-10 11:55:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: add all colors mode
+
+2015-03-10 10:19:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ video: Add support for 10 bit planar AYUV formats
+
+2015-03-10 09:27:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vorbis/gstvorbisparse.c:
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/volume/gstvolume.c:
+ * sys/xvimage/xvimagepool.c:
+ * tests/check/libs/rtpbasedepayload.c:
+ * tests/check/libs/video.c:
+ Fix double semicolons
+
+2015-03-09 21:35:59 -0400 Olivier Crete <olivier.crete@collabora.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Accept any capsfeatures
+
+2015-03-09 16:28:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-info.c:
+ video-info: validate parsed colorimetry
+ Validate the parsed colorimetry and reset to defaults when we get RGB
+ with a matrix or YUV without a matrix.
+
+2015-03-09 16:01:19 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: detect identity matrix
+ Do nothing if we have an identity matrix conversion.
+
+2015-03-09 15:58:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-info.c:
+ video-info: use default colorimetry on error
+ When we fail to parse the colorimetry property, fall back to the default
+ colorimetry for the format and dimension instead of leaving things
+ undefined.
+
+2015-03-09 11:25:41 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: unused value
+ Value set in ret is immediately overwritten in the next line outside of the if
+ block. Run reset but don't store return.
+ CID #1226470
+
+2015-03-09 12:13:44 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: only convert to/from rgb when needed
+ Only use the YUV->RGB matrix when we have YUV as input and only use the
+ matrix when we need to make YUV output.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745780
+
+2015-03-09 11:12:46 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: Link to an explanation why the seqnum comparison function does the right thing even for wraparounds
+
+2015-02-22 21:13:35 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: only return EOS upon clipping if applicable
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2015-02-22 21:11:50 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: only return EOS upon clipping if applicable
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2015-03-07 16:49:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video: Update orc generated C files
+
+2015-03-06 12:54:56 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add transfer full annotation for config
+
+2015-03-06 09:30:51 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: correct right-border location for YUY2, YVYU, UYVY
+ Remove 'r_border /= 2' in convert_fill_border(). It doesn't
+ take the right border to correct location.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745719
+
+2015-03-05 12:31:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/volume/gstvolume.c:
+ volume: Explicitly cast integers to doubles and then back to integers after multiplication
+ gcc 4.9.1 on ARM seems to have a bug that causes it to cast the float to an
+ integer first, resulting in a 0 scale factor for volume < 1.0.
+ As a side effect this change here will also improve accuracy of the result a
+ bit because we go via doubles instead of floats.
+ https://gcc.gnu.org/bugzilla/show_bug.cgi?id=65325
+ https://bugzilla.gnome.org/show_bug.cgi?id=745667
+
+2015-03-05 09:52:18 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: avoid scaler when size is unchanged
+
+2015-03-04 16:45:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add horizontal 2tap u16 orc function
+ Add slightly faster u16 horizontal resampler orc function.
+
+2015-03-04 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/libs/video.c:
+ check: add another generic converter test
+ Run conversion and scaling with borders.
+
+2015-03-04 12:21:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * tests/check/libs/video.c:
+ video-converter: don't reuse the input line when adding borders
+ When we need to add borders, we need a writable input line, so
+ don't reuse the source memory directly.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745207
+
+2015-03-04 09:24:27 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: fix latency query in FEC case
+ The max latency parameter is "the maximum time an element
+ synchronizing to the clock is allowed to wait for receiving all
+ data for the current running time" (docs/design/part-latency.txt).
+ https://bugzilla.gnome.org/show_bug.cgi?id=744338
+
+2015-03-03 16:36:20 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ textoverlay: Re-render if video size changed
+ https://bugzilla.gnome.org/show_bug.cgi?id=745554
+
+2015-03-03 22:56:37 +0530 Arun Raghavan <arun@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ audiobasesink: Reset audio clock if necessary
+ When the ringbuffer is deactivated and then acquired, if the audio clock
+ provided by the sink gets reset to zero, we need to add an offset to the
+ clock to make sure that subsequent samples are written out at the right
+ times. While we need to leave this to derived classes to take care of
+ when they provide their own clock (since that clock may or may not be
+ reset to zero), we can do this ourselves if we know the provided clock
+ is our own (which does reset to zero on a re-acquire).
+
+2015-03-02 16:42:23 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: avoid making scalers for outsize == 0
+
+2015-03-02 16:33:09 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-scaler.c:
+ video-converter: v-resample enough pixels
+ When we are using the fast linear resampler, use the ->inc to calculate
+ the first and last pixel we need so that we can do vertical resampling
+ on the right amount of pixels.
+
+2015-03-02 15:07:34 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-orc: fix unpack functions for RGB/RGB15 on BE
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745337
+
+2015-03-02 13:27:23 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ video-format: more fixes for big endian
+
+2015-03-02 12:26:23 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ video-format: add big-endian versions of RGB/BGR 15/16 pack/unpack
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745337
+
+2015-02-28 13:31:41 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: fix compiler warning
+ ‘return’ with no value, in function returning non-void
+
+2015-02-28 12:26:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play-1.0.1:
+ * tools/gst-play.c:
+ gst-play: add keyboard shortcut to cycle through trick modes
+ Make "t" activate trick modes and cycle through the various
+ modes.
+
+2015-02-28 11:37:27 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: fix indentation
+ Prevent gst-indent from messing up indentation, it
+ really doesn't like the G_GNUC_PRINTF thing here.
+
+2015-02-27 20:22:59 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/audiodecoder.c:
+ * tests/check/libs/audioencoder.c:
+ * tests/check/libs/videodecoder.c:
+ * tests/check/libs/videoencoder.c:
+ tests: fix crashes in {audio,video}{decoder,encoder} tests on 32-bit
+ Don't feed 64-bit integer variable into vararg function that expects
+ an unsigned integer to go with GST_TAG_TRACK_NUMBER. This would
+ cause crashes on 32-bit platforms, and if not that then test
+ failures if the comparisons fail later (at least on big endian
+ platforms).
+
+2015-02-27 15:07:36 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: description: Make static strings static
+ Otherwise, they're not guaranteed to still be valid when leaving the scope.
+ https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-27 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/pbutils.c:
+ tests: pbutils: more checking of returned description strings
+ https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-27 00:36:43 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst/adder/gstadder.c:
+ adder: Drop custom latency querying logic
+ The default latency query handler now implements the same logic already.
+
+2015-02-26 14:47:28 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: remove check for below zero for unsigned int
+ CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
+ number since it in an unsigned integer. Removing that check and only checking
+ if it is bigger than max and setting it appropriately.
+ CID #1271606
+
+2015-02-26 12:06:23 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/playback/gstdecodebin2.c:
+ playback: Fix broken GList modification
+ When we modify a GList (via g_list_delete_link), always reassign the
+ new head to the original GList. Otherwise we end up with
+ filtered_errors being corrupt (the head might have been the element
+ removed)
+
+2015-02-26 11:06:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play-1.0.1:
+ gst-play: add new keyboard shortcuts to man page
+
+2015-02-26 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: more fine-grained playback rate control
+ Use smaller steps for lower rates to allow more
+ fine-grained control. Handle jump across 0 properly
+ from both sides (just flip direction where we would
+ have gone down to 0 instead). Don't artificially
+ limit rates to +/- 10x. Print new rate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745174
+
+2015-02-26 10:20:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: stash current playback rate in app structure
+ https://bugzilla.gnome.org/show_bug.cgi?id=745174
+
+2015-02-25 18:52:11 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * tools/gst-play.c:
+ gst-play: support changing the playback rate in interactive mode
+ It is fun to have this feature, also it is useful for testing decoders.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745174
+
+2015-02-25 17:00:34 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: we can use the scaler without scalers to copy
+
+2015-02-25 16:50:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: only make a scaler when we are scaling
+ Only make a scaler when we are actually doing any scaling. Without
+ scalers, the scale function will simply do a copy.
+
+2015-02-25 16:49:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add support for copy
+ When no scalers are given, simply do a copy of the requested area.
+
+2015-02-25 16:15:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: activate scaler fastpath depending on method
+ Only activate the scaler fastpath for x2 up and downscale when the
+ scaler method is respectively nearest and linear because that is what
+ those fastpaths really implement.
+
+2015-02-25 15:33:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add scaler optimization
+ If we are vertically downscaling, it is better to first downscale and
+ then do the horizontal scaling in most cases.
+
+2015-02-25 15:32:57 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: remove unused case
+
+2015-02-25 11:38:17 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ video-converter: don't overwrite border alpha
+ Let border alpha and image alpha be independent.
+
+2015-02-24 17:33:57 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: use 1.0 as default alpha
+
+2015-02-24 17:26:31 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: add alpha handling
+ Add support for alpha. Make it possible to copy, set and multiply the
+ alpha value of a frame during conversion.
+ Set the border alpha to 0xff by default.
+ Go over some of the fastpaths and add alpha handling.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745006
+
+2015-02-24 17:20:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: fix chroma subsampling
+ Also adjust the output line number with the offset.
+
+2015-02-24 10:01:18 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: disable fastpath when scaling and gamma
+ Disable the fastpath when scaling and doing gamma remap.
+
+2015-02-24 09:54:18 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: don't do gamma on alpha channel
+ The alpha channel is not supposed to be gamma encoded.
+
+2015-02-24 16:06:08 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: fix deadlock when resetting buffering
+ This function is static, and only ever called with the expose lock
+ taken. It thus has no reason to take this lock itself.
+ This was introduced by one of my locking fixes from 741355.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741355
+
+2015-02-24 12:38:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: minor docs fix
+
+2014-05-27 13:54:06 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: forward template and ring buffer settings to existing decodebins
+ https://bugzilla.gnome.org/show_bug.cgi?id=744844
+
+2015-02-23 17:24:52 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: move null check
+ Check if dbin->decode_chain is NULL before running drain_and_switch_chains()
+ because if it is, we shouldn't run that function or it will segfault.
+ CID #1271074
+
+2015-02-23 01:32:14 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Don't send pending events before decode
+ Make sure to update the output segment to track the segment
+ we're decoding in, but don't actually push it downstream until
+ after buffers are decoded.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744806
+
+2015-02-08 05:19:25 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ videodecoder: Add drain() vfunc
+ drain() is a new vfunc which does what finish() does, while
+ explicitly requiring the decoder be able to continue processing
+ data afterward.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734617
+
+2015-02-22 16:57:57 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ Revert "videodecoder: drain current segment upon new one to ensure correct flow return"
+ This reverts commit cc1b4eaf9ebe4568f9c2c64338cef1b2edbdca3f.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=734617
+
+2015-02-22 16:57:50 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ Revert "audiodecoder: drain current segment upon new one to ensure correct flow return"
+ This reverts commit 696b8cdc40f033ff0a45ebe620279130152fb2f8.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=734617
+
+2015-02-21 17:42:08 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: drain current segment upon new one to ensure correct flow return
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2015-02-21 17:41:50 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: drain current segment upon new one to ensure correct flow return
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2015-02-20 12:34:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Only consider non-parser factories for generating the post-parser capsfilter caps
+ Otherwise if there are multiple parsers we would most likely break negotiation
+ of the stream-format/alignment wanted by the decoders as parsers generally
+ support all possible stream-formats and alignments.
+
+2015-02-19 15:51:19 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ audio: video: fix a few GI annotations
+ transfer-full -> transfer full
+ @Since -> Since
+
+2015-02-05 12:07:50 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: fix deadlock between downward state change and pad addition
+ If caps on a newly added pad are NULL, analyze_new_pad will try to
+ acquire the chain lock to add a probe to the pad so the chain can
+ be built later. This comes from the streaming thread, in response
+ to headers or other buffers causing this pad to be added, so the
+ stream lock is taken.
+ Meanwhile, another thread might be destroying the chain from a
+ downward state change. This will cause the chain to be freed with
+ the chain lock taken, and some elements are set to NULL here, which
+ can include the parser. This causes pad deactivation, which tries
+ to take the element's pad's stream lock, deadlocking.
+ Fix this by keeping track of which elements need setting to NULL,
+ and only do this after the chain lock is released. Only the chain
+ manipulation needs to be locked, not the elements' state changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741355
+
+2015-02-04 11:46:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: guard against the decode chain going while a pad is added
+ https://bugzilla.gnome.org/show_bug.cgi?id=741355
+
+2015-02-03 17:06:43 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: possible fix for deadlock when spamming "next song"
+ There was a deadlock between a thread changing decodebin/demuxer
+ state from PAUSED to READY, and another thread pushing data
+ when starting.
+ From the stack trace at
+ https://bug741355.bugzilla-attachments.gnome.org/attachment.cgi?id=292471,
+ I deduce the following is happening, though I did not reproduce the
+ problem so I'm not sure this patch fixes it.
+ The streaming thread (thread 2 in that stack trace) takes the demuxer's
+ sink pad's stream lock in gst_ogg_demux_perform_seek_pull and will
+ activate a new chain. This ends up causing the expose lock being taken
+ in _pad_added_cb in decodebin.
+ Meanwhile, a state changed is triggered on thread 1, which takes the
+ expose lock in decodebin in gst_decode_bin_change_state, then frees
+ the previous chain, which ends up calling gst_pad_stop_task on the
+ demuxer's task, which in turn takes the demuxer's sink pad's stream
+ lock, deadlocking as both threads are now waiting for each other.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741355
+
+2015-02-18 20:58:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: ensure tags have been fetched before pulling data
+ Otherwise upstream can get confused about offsets as there will
+ be a jump once the tags have been parsed due to the stripped area.
+ If upstream pulls from 0 to 100, and then tagdemux does the
+ tag reading and finds out that the first 200 bytes are the tag, the
+ next pull from upstream will have an offset of 200 bytes. So
+ upstream will get the following data:
+ 0 - 100, 300 - (EOS), as it will continue requesting from where
+ it has last stopped, but tagdemux will add an offset to skip the
+ tags.
+ This patch makes sure that the tags have been parsed and skipped
+ since the first pull range call.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744580
+
+2015-02-19 01:30:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Reset the default query return value when the iterator has to resync
+
+2015-02-19 01:21:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Let the latency query fail if one of the source queries fails
+
+2015-02-18 17:41:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: Remove g_warnings() for the deprecated audio property
+ Otherwise there are g_warnings() already when just using gst-inspect or
+ dumping a pipeline graph.
+
+2015-02-18 11:34:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: description: fix MPEG-2 video profiles in description
+ We would accidentally use the profile nick as profile name
+ in the description for MPEG video that's not version 4.
+
+2015-01-29 18:49:45 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Pass object, not GValue to debug print
+
+2015-02-16 23:54:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ audiovisualizer: don't use private GMutex implementation details
+ Don't use private GMutex implementation details to check
+ whether it has been freed already or not. Just turn dispose
+ function into finalize function which will only be called
+ once, that way we can just clear the mutex unconditionally.
+
+2015-02-15 13:51:36 +0800 Song Bing <b06498@freescale.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Use the same waiting function for EOS and stream switches
+ Also improve the waiting condition for stream switches, which was assuming
+ before that the condition variable will only stop waiting once when it is
+ signaled. But the documentation says that there might be spurious wakeups.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-01-26 11:14:13 +0800 Song Bing <b06498@freescale.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/streamsynchronizer.c:
+ streamsynchronizer: Unit test for streamsynchronizer's EOS handling
+ Test that a pipeline can change from PLAYING to PAUSED and back in
+ the following scenarios:
+ 1. One track reach EOS after pushed some buffers while another track
+ still pushes buffers
+ 2. One track reach EOS without buffers while another track still pushes
+ buffers
+ https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-01-12 17:40:25 +0800 Song Bing <b06498@freescale.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Send GAP events from the pads' streaming threads
+ Change the GAP events that are currently sent from the chain function of
+ the current pad to all other EOS pads. They should instead be sent from
+ their own streaming threads.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-01-12 16:08:33 +0800 Song Bing <b06498@freescale.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ * gst/playback/gststreamsynchronizer.h:
+ streamsynchronizer: Send GAP event to finish preroll when change state from PLAYING to PAUSED
+ Wait in the event function when EOS is received until all pads are EOS
+ and then forward the EOS event from each pads own event function.
+ Also send a new GAP event for EOS pads from the event function whenever
+ going from PLAYING->PAUSED by shortly waking up the GCond. This is needed
+ to allow sinks to pre-roll again, as they did not receive EOS yet because
+ we blocked that, but also will never get data again.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-02-16 09:48:03 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/codec-utils.c:
+ Revert "codec-utils: Handle the two rext profiles for h265"
+ This reverts commit 19b93566801a56e7b043a670b7edcf8f2da06619.
+ These two "profiles" are actually a complete set of profiles, which we will
+ need to handle separately. Unfortunately it seems like we need information
+ from the SPS to detect the exact profile.
+
+2015-02-15 20:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: description: move some code into utility function
+
+2015-02-15 20:05:13 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * tests/check/libs/pbutils.c:
+ pbutils: descriptions: add H.265 profile to description if available
+ https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-15 19:03:38 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * tests/check/libs/pbutils.c:
+ pbutils: descriptions: add MPEG-4 video profile to description if available
+ https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-15 18:37:38 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * tests/check/libs/pbutils.c:
+ pbutils: descriptions: add Dirac/VC-2 profile to description if available
+ https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-15 18:14:18 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * tests/check/libs/pbutils.c:
+ pbutils: descriptions: add H.264 profile to description if available
+ https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-13 22:56:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/install-plugins.c:
+ install-plugins: fix indentation and add Since marker
+ Forgot to squash this into the actual patch before pushing.
+
+2015-02-13 22:49:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * win32/common/libgstpbutils.def:
+ install-plugins: add new API to exports .def and to docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=744465
+
+2015-02-03 10:47:11 +0100 Kalev Lember <kalevlember@gmail.com>
+
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * gst-libs/gst/pbutils/install-plugins.h:
+ install-plugins: Add API to suppress confirmation before searching
+ The new gst_install_plugins_context_set_confirm_search() API can be used
+ to pass a hint to modify the behaviour of the external installer
+ process.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744465
+
+2015-02-02 16:16:46 +0100 Kalev Lember <kalevlember@gmail.com>
+
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * gst-libs/gst/pbutils/install-plugins.h:
+ install-plugins: Add API for passing desktop ID and startup ID
+ The new gst_install_plugins_context_set_desktop_id() and
+ gst_install_plugins_context_set_startup_notification_id() API can be
+ used to pass extra details to the external installer process.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744465
+
+2015-02-12 12:08:16 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video-orc: update with new methods
+
+2015-02-12 11:38:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-format: add orc function for RGB15/16 unpack
+
+2015-02-10 21:57:02 -0800 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: improve debug log
+ Log the human readable pad_link_return desc as well.
+
+2015-02-11 15:57:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/codec-utils.c:
+ codec-utils: Handle the two rext profiles for h265
+ These values are for now taken from x265 and need to be checked against
+ the spec. Especially we need to check if information from other fields
+ need to be taken into consideration too, e.g. the bit depth and chroma
+ index from the SPS.
+ This however makes 4:4:4 output of x265enc actually work.
+
+2015-02-11 13:43:11 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst/adder/gstadder.c:
+ * gst/playback/gsturidecodebin.c:
+ Improve and fix LATENCY query handling
+ This now follows the design docs everywhere, especially the maximum latency
+ handling.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744106
+
+2015-02-11 14:16:21 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ Improve and fix LATENCY query handling
+ This now follows the design docs everywhere, especially the maximum latency
+ handling.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744106
+
+2015-02-11 13:32:25 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/video-scaler.h:
+ * win32/common/libgstvideo.def:
+ video-scaler: add 2d scaler
+ Make a convenience function that combines 2 scalers to perform a 2d
+ scale. This removes quite a bit of overhead in method calls when doing a
+ typical scale and it also can reuse a piece of unused memory in the
+ vertical scaler.
+ Use the 2d scaler in video-converter and remove the other scalers and
+ temp memory.
+
+2015-02-10 16:43:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Fix YUY2 formats and friends
+ Only merge scalers for selected formats.
+ Use nearest neighbour scaling for chroma when doing nearest neighbour
+ for the luma.
+ Also fastpath GRAY16_OE in nearest neighbour.
+ configure parameters correctly for packed fastpath.
+
+2015-02-10 16:40:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: Small performance tweaks
+ Small performance tweaks for RGB and friends.
+ Add, but ifdef out, alternative nearest neighbour scaling, it is slower
+ than the current table based version.
+ Use memcpy instead of orc_memcpy because it is measurably faster.
+ Fix YUY2 and friends vertical scaling.
+
+2015-02-10 16:44:38 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: Guard against (impossible) bits!=16 && bits!=8 case to fix compiler warning with clang
+ video-scaler.c:1331:14: error: variable 'func' is used uninitialized whenever 'if' condition is false
+ [-Werror,-Wsometimes-uninitialized]
+ } else if (bits == 16) {
+ ^~~~~~~~~~
+ video-scaler.c:1348:3: note: uninitialized use occurs here
+ func (scale, src_lines, dest, dest_offset, width, n_elems);
+ ^~~~
+ video-scaler.c:1331:10: note: remove the 'if' if its condition is always true
+ } else if (bits == 16) {
+ ^~~~~~~~~~~~~~~~
+ video-scaler.c:1260:27: note: initialize the variable 'func' to silence this warning
+ GstVideoScalerVFunc func;
+ ^
+ = NULL
+
+2015-02-10 16:38:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Use correct enum type to fix compiler warnings with clang
+ video-converter.c:3406:12: error: implicit conversion from enumeration type 'GstVideoFormat' to different
+ enumeration type 'GstFormat' [-Werror,-Wenum-conversion]
+ format = convert->fformat[plane];
+ ~ ^~~~~~~~~~~~~~~~~~~~~~~
+ video-converter.c:3413:44: error: implicit conversion from enumeration type 'GstFormat' to different enumeration
+ type 'GstVideoFormat' [-Werror,-Wenum-conversion]
+ gst_video_scaler_horizontal (h_scaler, format,
+ ~~~~~~~~~~~~~~~~~~~~~~~~~~~ ^~~~~~
+ video-converter.c:3471:12: error: implicit conversion from enumeration type 'GstVideoFormat' to different
+ enumeration type 'GstFormat' [-Werror,-Wenum-conversion]
+ format = convert->fformat[plane];
+ ~ ^~~~~~~~~~~~~~~~~~~~~~~
+ video-converter.c:3487:42: error: implicit conversion from enumeration type 'GstFormat' to different enumeration
+ type 'GstVideoFormat' [-Werror,-Wenum-conversion]
+ gst_video_scaler_vertical (v_scaler, format, lines, d + out_x, i,
+ ~~~~~~~~~~~~~~~~~~~~~~~~~ ^~~~~~
+ video-converter.c:3551:12: error: implicit conversion from enumeration type 'GstVideoFormat' to different
+ enumeration type 'GstFormat' [-Werror,-Wenum-conversion]
+ format = convert->fformat[plane];
+ ~ ^~~~~~~~~~~~~~~~~~~~~~~
+ video-converter.c:3569:46: error: implicit conversion from enumeration type 'GstFormat' to different enumeration
+ type 'GstVideoFormat' [-Werror,-Wenum-conversion]
+ gst_video_scaler_horizontal (h_scaler, format,
+ ~~~~~~~~~~~~~~~~~~~~~~~~~~~ ^~~~~~
+ video-converter.c:3577:42: error: implicit conversion from enumeration type 'GstFormat' to different enumeration
+ type 'GstVideoFormat' [-Werror,-Wenum-conversion]
+ gst_video_scaler_vertical (v_scaler, format, lines, d + out_x, i,
+ ~~~~~~~~~~~~~~~~~~~~~~~~~ ^~~~~~
+
+2015-02-10 15:25:04 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-converter: bits variable always set
+ In function gst_video_scaler_vertical() the bits variable is always
+ set to either 8 or 16 in every possible format. No need to initialize it.
+ If the format isn't valid it goes to no_func, so there is no need to
+ handle the case of bits not being 8 or 16.
+ CID #1268401
+
+2015-02-10 11:15:22 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: only enable backlog for interlaced video
+ Skip lines we don't need.
+
+2015-02-10 09:30:44 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add fastpath for NV formats
+
+2015-02-10 09:20:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: fix pstride of NV16 and NV24 formats
+
+2015-02-09 18:01:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * tests/check/libs/rtsp.c:
+ rtspmessage: map headers we know that are added by string to their enum
+ That way we can look them up by their field enum later as well.
+
+2015-02-09 17:49:12 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/rtsp.c:
+ tests: rtsp: add some unit tests for new GstRTSPMessage API
+
+2015-02-09 16:24:19 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ * win32/common/libgstrtsp.def:
+ rtspmessage: add API to add and get custom headers
+ Add API to add and get custom headers that are not
+ covered by our header fields enum. This is backwards
+ compatible in that it will also work for our defined
+ fields, so if we ever add a new header field to the
+ enum, get_header_by_name() for the same header string
+ will still work.
+ API: gst_rtsp_message_add_header_by_name()
+ API: gst_rtsp_message_take_header_by_name()
+ API: gst_rtsp_message_remove_header_by_name()
+ API: gst_rtsp_message_get_header_by_name()
+
+2015-02-09 17:51:00 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: Add more fastpaths
+ Add fastpaths for all planar conversion and scaling.
+ Improve gray and alpha handling.
+ Add option to specify the chroma resampler method and set to linear as
+ default.
+
+2015-02-09 13:20:43 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add generic planar scaler/converter
+ Add code to convert and scale between any planar format and use it in
+ the fastpaths of some planare converters.
+
+2015-02-09 10:20:37 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Fix compiler warnings by using the correct enum type
+ video-converter.c:3645:24: error: implicit conversion from enumeration type
+ 'GstFormat' to different enumeration type 'GstVideoFormat'
+ [-Werror,-Wenum-conversion]
+ convert->fformat = fformat;
+ ~ ^~~~~~~
+ video-converter.c:3667:24: error: implicit conversion from enumeration type
+ 'GstFormat' to different enumeration type 'GstVideoFormat'
+ [-Werror,-Wenum-conversion]
+ convert->fformat = fformat;
+ ~ ^~~~~~~
+ video-converter.c:3963:50: error: implicit conversion from enumeration type
+ 'const GstVideoFormat' to different enumeration type 'GstFormat'
+ [-Werror,-Wenum-conversion]
+ if (!setup_scale (convert, transforms[i].fformat))
+ ~~~~~~~~~~~ ~~~~~~~~~~~~~~^~~~~~~
+
+2015-02-07 03:56:05 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: Don't pass GstCollectData as a GstObject to GST_DEBUG
+
+2015-02-06 13:39:04 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-scaler.c:
+ video-converter: add more scaler fastpaths
+
+2015-02-06 13:25:51 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ video-orc: fix loading of param
+ param loading ignores the x4, loading only part of the param.
+
+2015-02-06 12:35:01 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add border and crop to more fastpaths
+
+2015-02-06 12:28:54 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: fix border for YUY2 and friends
+ Convert as many pixels as the max subsampling so that we convert a
+ complete group of pixels.
+
+2015-02-06 15:39:14 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: support AYUV border
+ Convert the border color from ARGB to AYUV, using
+ colorimetry matrix when output format is YUV.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741640
+
+2015-02-06 10:57:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: fix swapped border width
+ And also do nothing when there is no border.
+
+2015-02-06 10:56:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: actually draw the border in some fastpaths
+ Don't forget to draw the border after doing the fastpath conversion.
+
+2015-02-06 10:53:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: clamp width and heigth
+ Clamp the width and height based on the in and out offsets.
+
+2015-02-06 10:50:09 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: add unaligned fallbacks
+ Add fallback C implementations for when we can't call the ORC function
+ because of bad alignment.
+
+2015-01-28 05:20:19 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Where possible, skip decode for GST_SEGMENT_FLAG_TRICKMODE_NO_AUDIO
+ If we have timestamps on input buffers and are in trickmode no-audio
+ mode, then don't pass anything to the subclass for decode and simply
+ send gap events downstream
+ Only for forward playback for now - reverse requires accumulating
+ GAP events and pushing out in reverse order.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735666
+
+2015-02-05 17:44:59 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: Re-work GAP buffer and trick-mode handling
+ In trickmode no-audio mode, or when receiving a GAP buffer,
+ discard the contents and render as a GAP event instead.
+ Make sure when rendering a gap event that the ring buffer will
+ restart on PAUSED->PLAYING by setting the eos_rendering flag.
+ This mostly reverts commit 8557ee and replaces it. The problem
+ with the previous approach is that it hangs in wait_preroll()
+ on a PLAYING-PAUSED transition because it doesn't commit state
+ properly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735666
+
+2015-02-03 20:38:44 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Add a little timestamping debug output
+
+2015-02-03 01:19:05 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/theora/gsttheoradec.c:
+ theora: If no header packets in stream, look for them in the caps
+ Makes theora work in cases where the header packets are only in the caps
+ (because theoradec was connected to oggdemux late and missed the
+ beginning of the stream)
+
+2015-02-02 22:23:51 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/theora/gsttheoradec.c:
+ theora: Remove FIXME and return GST_CUSTOM_FLOW_DROP for header packet handling
+ This FIXME is easily fixed :)
+
+2015-01-31 05:12:10 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Remove pointless else{} around some code
+
+2015-01-31 05:09:46 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Fix reverse playback when there's only one gather set.
+ The decoder can fail to drain on EOS if there was only one gather
+ set, because it will never have sent the segment event downstream
+ and set the output segment, and fail to detect that the rate < 0.0
+ Make sure to send pending events before sending all the gather data
+ for decode.
+
+2014-10-09 03:31:58 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/video-frame.h:
+ video: Fix simple typo in GstVideoFrameMapFlags docs
+
+2015-02-05 17:49:55 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add crop and border to some fastpaths
+
+2015-02-05 17:18:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: add support for borders in scale fastpath
+ Add support for borders and cropping in the scaler fastpaths.
+
+2015-02-05 15:03:24 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: disable fastpath for crop and border
+ Add crop and border properties to the fastpath table and only select
+ fastpath functions when it can handle the cropping or borders.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=744028
+
+2015-02-04 18:01:51 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-scaler.c:
+ video-converter: add fastpath for some gray formats
+
+2015-02-04 17:44:31 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-scaler.c:
+ video-converter: add fastpath for some more RGB formats
+ Add fastpath for RGB and BGR.
+ Add fastpath for nearest resampling for RGB15 and RGB16 formats.
+
+2015-02-04 16:37:22 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: skip lines we don't need
+ Make sure to skip unused lines instead of doing a useless horizontal
+ resampling.
+
+2015-02-04 12:08:21 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: fix memory leak
+ In gst_video_scale_fixate_caps () it can goto done without freeing the memory
+ of the tmp GstStructure. This makes it go out of scope and leak.
+ CID #1265766
+
+2015-02-04 11:25:54 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/video/video-resampler.c:
+ video-resampler: make sure params.envelope is initialized
+ In gst_video_resampler_init () if method is GST_VIDEO_RESAMPLER_METHOD_NEAREST
+ then params.envelope is not initialized but still used later in line 382.
+ Make sure this variable is initiliazed to avoid undefined behaviour.
+ CID #1256568
+
+2015-02-03 12:23:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ video{enc,dec}oder: Don't reset latency all the time and handle max=GST_CLOCK_TIME_NONE correctly
+ max=NONE means that *this* element has no maximum latency. If upstream had a
+ maximum latency we must not override it with NONE.
+
+2015-02-03 12:15:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audio{enc,dec}oder: Always directly post latency messages on the bus when the subclass sets the latency
+ Instead of doing it only in setcaps for the encoder, and never at all for the
+ decoder.
+
+2015-02-03 12:12:18 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audio{enc,dec}oder: Handle max_latency == GST_CLOCK_TIME_NONE
+ And initialize the latencies with 0 and NONE.
+
+2015-01-28 05:26:06 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: Don't render a GAP silence buffer
+ Don't render out silence samples to a buffer, just
+ start the clock running, since any buffer with the
+ GAP flag will be discarded in render() now anyway.
+
+2015-01-28 22:42:17 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: Make sure the ringbuffer is started before waiting
+ Don't call the basesink wait_event implementation until we're sure
+ the ringbuffer is running, because it might wait on a non-running
+ clock.
+
+2015-01-27 02:04:22 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: drop GAP buffers, or all buffers in trickmode no-audio mode
+ Make the base audio sink throw away buffers marked GAP, or all
+ incoming buffers when performing a trick play with
+ GST_SEGMENT_TRICKMODE_NO_AUDIO flag set, and make sure to start
+ the ringbuffer when that happens so the clock starts running.
+ Preserve the timing calculations when rendering, so state is all
+ updated the same, but just don't render samples.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735666
+
+2015-01-29 17:58:27 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: do not throw a flow error on flushing
+ If the streaming task attempts to read a chain while the pipeline
+ is stopping (which can happen if the pipeline stops shortly after
+ start or a new URI being setup in gapless playback case), it will
+ see a flushing return from upstream, and should then also return
+ flushing to the caller, rather than emit a flow error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=722442
+
+2015-01-28 16:43:59 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opusenc: change audio property to audio-type
+ This is now an enum with values generic (default) and voice.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740891
+
+2015-01-28 17:44:57 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Fix compiler warnings
+ video-converter.c:3073:48: error: implicit conversion from enumeration type 'GstFormat' to different enumeration type 'GstVideoFormat'
+ [-Werror,-Wenum-conversion]
+ gst_video_scaler_horizontal (h_scaler, format,
+ ~~~~~~~~~~~~~~~~~~~~~~~~~~~ ^~~~~~
+ video-converter.c:3081:44: error: implicit conversion from enumeration type 'GstFormat' to different enumeration type 'GstVideoFormat'
+ [-Werror,-Wenum-conversion]
+ gst_video_scaler_vertical (v_scaler, format, lines, d, i, out_w);
+ ~~~~~~~~~~~~~~~~~~~~~~~~~ ^~~~~~
+ video-converter.c:3137:24: error: implicit conversion from enumeration type 'const GstVideoFormat' to different enumeration type 'GstFormat'
+ [-Werror,-Wenum-conversion]
+ convert->fformat = GST_VIDEO_INFO_FORMAT (in_info);
+ ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ ../../../gst-libs/gst/video/video-info.h:125:43: note: expanded from macro 'GST_VIDEO_INFO_FORMAT'
+ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ ../../../gst-libs/gst/video/video-format.h:361:59: note: expanded from macro 'GST_VIDEO_FORMAT_INFO_FORMAT'
+ ~~~~~~~~^~~~~~
+ video-converter.c:3157:24: error: implicit conversion from enumeration type 'GstVideoFormat' to different enumeration type 'GstFormat'
+ [-Werror,-Wenum-conversion]
+ convert->fformat = GST_VIDEO_FORMAT_GRAY8;
+
+2015-01-28 17:43:59 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video: Update orc files
+
+2015-01-28 17:37:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * win32/common/libgstvideo.def:
+ defs: update
+
+2015-01-28 17:32:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/video-scaler.h:
+ video-converter: add fast-path scaler for some packed YUV formats
+ Add fast path scaling for YUY2 and other packed YUV formats. Add a new
+ method to merge the scalers of the Y and UV components into one scaler.
+ Add faster horizontal 2tap scaler.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=741987
+
+2015-01-28 17:30:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: don't do dithering
+
+2015-01-28 17:30:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.h:
+ video-converter: the default is BAYER dithering
+
+2015-01-28 17:29:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: don't do dither when set to NONE
+
+2015-01-28 11:38:16 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix taps calculation for pstride == 1
+ Take pstride into consideration when calculating the scaler taps.
+
+2015-01-28 04:51:25 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: Make sure the ringbuffer really starts when we need it to
+ Some audio sink sub-classes (pulsesink) don't start their clock
+ when the ringbuffer starts, but always have to on EOS. When we
+ explicitly need to start the ringbuffer, make sure sub-classes will
+ do it by (ab)using the existing eos_rendering flag.
+
+2014-12-11 01:54:07 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/examples/playback/playback-test.c:
+ playback-test: Support new skip seek flags
+ Support the new SEEK_TRICKMODE_KEY_UNITS and SEEK_TRICKMODE_NO_AUDIO
+ flags added to core
+ https://bugzilla.gnome.org/show_bug.cgi?id=735666
+
+2015-01-27 13:39:14 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/audio/gstaudiopack-dist.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst/adder/gstadderorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.c:
+ orc: update orc files
+
+2015-01-27 10:28:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add fastpath for planar scaling
+ Add fastpaths for scaling of planar subsampled formats.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=741987
+
+2015-01-27 10:04:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add support for monochroma formats
+ Add support for scaling of images with pstride == 1. This can be used
+ to scale individual planes later.
+ Rework some of the scaling code to take the pstride as a parameter.
+
+2015-01-27 09:51:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: disable chroma and matrix operations
+ Ignore chroma subsampling and color matrix transformations like the
+ old videoscale used to do. This is to make the performance like it was
+ before.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=741987
+
+2015-01-26 12:52:40 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: fix GBR unpack
+
+2015-01-27 01:31:50 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ audiodecoder: Fix typo in documentation
+ Fix a couple of harmless warnings in the gtk-doc parsing
+
+2015-01-23 12:46:41 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/video/video-dither.c:
+ video: Fix leaked dither object in error cases
+ Coverity CID : 1256564
+
+2015-01-21 15:22:15 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ visual: fix caps leak
+ Fix leak of caps event and of caps objects when setting caps on sink and src
+ pads. Sync audiovisualizer class implementation to the one in gst-plugins-bad.
+ This commit matches c5ef1bee7318f057aa1f542d5a1474b75e85131a in that module.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-01-21 14:46:15 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ visual: post QoS messages when dropping frames due to QoS
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-01-21 09:49:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ * gst-libs/gst/video/video-format.c:
+ * gst/audioconvert/audioconvert.c:
+ * gst/audioconvert/gstaudioquantize.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/resample.c:
+ Constify some static arrays everywhere
+
+2015-01-21 09:42:21 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/alsa/gstalsa.c:
+ alsa: Constify channel position table
+
+2015-01-21 09:41:23 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/alsa/gstalsa.c:
+ alsa: Fix indention
+
+2015-01-21 08:33:57 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * ext/alsa/gstalsa.c:
+ alsa: Allow to use 8 bit samples with ALSA
+ 8 bit samples have no (0) as endianness, not the native endianness.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739446
+
+2015-01-21 09:39:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-format.c:
+ audio-format: Constify the audio format table
+
+2015-01-21 09:37:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosrc: Fill in the correct silence
+ For unsigned raw formats this is not all zeroes, and for non-raw formats
+ we just continue to assume all zeroes for now.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739446
+
+2015-01-21 08:47:26 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * gst-libs/gst/audio/gstaudiosink.c:
+ audiosink: Fill in the correct silence
+ For unsigned raw formats this is not all zeroes, and for non-raw formats
+ we just continue to assume all zeroes for now.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739446
+
+2015-01-20 19:14:21 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: Only emit EOS signal after all buffers are consumed
+ Otherwise the application will possibly shut down the pipeline already
+ because EOS is received, while there are still some buffers pending.
+
+2015-01-20 15:08:24 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ dcodebin2: fix lock/unlock mismatch on multiqueue overrun
+
+2015-01-13 16:07:06 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/audioresample/resample.c:
+ audioresample: Try to prevent endless looping
+ Speex may decide not to consume any samples because it can't write any. I've
+ seen a hang during draining caused by the resample loop never terminating.
+ In that case, resampling happened as normal until olen was 0 but ilen was
+ still 1. _process_native then reduced ichunk to 0, so ilen never decreased
+ below 1 and the loop never terminated.
+ Instead of reverting 684cf44 ({audioresample: don't skip input samples),
+ break only if all output samples have been produced and speex refuses
+ to consume any more input samples.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732908
+
+2015-01-19 11:17:18 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videorate/Makefile.am:
+ videorate: Add $(GST_PLUGINS_BASE_CFLAGS) to be able to find gst/video/video.h
+
+2015-01-18 14:58:36 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/videorate/Makefile.am:
+ * gst/videorate/gstvideorate.c:
+ videorate: Implement allocation query
+ The videorate element keeps 1 buffer internally. This buffer need
+ to be requested during allocation query otherwise the pipeline may
+ stall.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738302
+
+2015-01-18 14:17:07 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/videorate/Makefile.am:
+ * gst/videorate/gstvideorate.c:
+ Revert "videorate: Implement allocation query"
+ This reverts commit 3c04db4a307048db70ee1d08c1d62e26ad9569d8.
+
+2015-01-18 11:02:00 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/videorate/Makefile.am:
+ * gst/videorate/gstvideorate.c:
+ videorate: Implement allocation query
+ VideRate keeps 1 buffer in order to duplicate base on closest buffer
+ relative to targeted time. This extra buffer need to be request
+ otherwise the pipeline may stall when fixed size buffer pool is used.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738302
+
+2015-01-17 14:51:48 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Fix compilation
+
+2015-01-12 14:38:09 +0100 Branislav Katreniak <bkatreniak@nuvotechnologies.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: do call set_queue_size in no_more_pads_cb
+ Consider pipeline: gst-launch-1.0 playbin uri=http://example.com/a.ogg
+ Consider 128kbit audio stream.
+ As soon as uridecodebin detects the bitrate, it configures its input
+ queue2 max-size to 32000 bytes.
+ The 2MB buffer in multiqueue is nearly 2 orders of magnitude bigger.
+ This non-deterministically drives queue2 buffer anywhere from
+ 100% to 0% until multiqueue is filled.
+ This patch sets multiqueue size to 5 buffers early in no_more_pads_cb.
+ Partly reverts commit db771185ed750627a6a1824c42b651d739e1b4a4.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740689
+
+2015-01-16 15:21:14 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: free old groups when switching groups
+ Old groups are freed with one switch's delay when switching groups.
+ They're freed in a scratch thread to avoid delaying the switch.
+
+2014-12-12 17:02:35 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: fix clipped duration determination for non 0 based segments
+ https://bugzilla.gnome.org/show_bug.cgi?id=740422
+
+2015-01-15 10:51:37 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioutilsprivate.c:
+ audio: Keep caps features when building the downstream filter
+ Based on 5fd4e3e0b6cc4f30d7b1489a105db946b43f1a9f for video
+ by Alessandro Decina.
+
+2015-01-15 13:54:14 +1100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst-libs/gst/video/gstvideoutilsprivate.c:
+ videoutils: keep caps features in account when building the downstream filter
+ See 00c2ce6 and https://bugzilla.gnome.org/show_bug.cgi?id=741263 for reference.
+
+2015-01-14 10:35:34 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/examples/playback/playback-test.c:
+ examples: playback: add labels with supported seek range
+ Add the supported seeking range in the advanced seek area.
+ Also implement seeking querying the pipeline to retrieve those
+ values and show to the user. It is done in a smaller frequency
+ compared to the position/duration querying.
+
+2015-01-13 19:25:52 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: disable pad link checks as it has already been done
+ Decodebin has already added the element to the bin and should only
+ select caps compatible pads. It should disable the pad link checks
+ to avoid doing those again.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742885
+
+2015-01-13 16:58:34 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ visual: cleanup
+ Shameful fix to a silly mistake in the previous commit. Above email address for
+ any mockery
+
+2015-01-13 16:36:09 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ visual: handle the return of the setup function
+ Make the baseclass future proof by handling the gboolean return of the setup
+ function. So if/when a child class uses this the base class is ready.
+
+2015-01-13 16:09:49 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ Revert "visual: remove unnecessary variable"
+ This reverts commit a91d521a3602f33083405467db9454d422b9da1b.
+ Being a base class it is better to check the value instead of ignoring it since
+ a child class could be created that returns valuable information.
+
+2015-01-13 15:07:56 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ visual: remove unnecessary variable
+ klass->setup (scope) will always return TRUE since all children of this class
+ do so, no need to store the return. Besides, the value is overwritten a few
+ lines down before it is ever used. Save the unnecessary memory and instructions.
+ CID #1226467
+
+2015-01-12 15:27:18 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ visual: use unused value
+ ret is assigned but not used and in the next cycle of the loop it is overwritten
+ with default_prepare_output_buffer (). If there is a flow error the function
+ should return instead.
+ CID #1226475
+
+2015-01-12 15:56:06 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f2c6b95 to bc76a8b
+
+2015-01-08 21:20:14 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: start ringbuffer if needed upon commit
+ ... to provide for a running clock.
+
+2015-01-02 14:34:41 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: fix comment typo
+
+2015-01-09 15:38:09 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/video/video-dither.c:
+ video-dither: remove check for below zero for unsigned value
+ CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
+ number since it is an unsigned integer. Removing that check and only checking if
+ it is bigger than max and setting it appropriately.
+ CID 1256559
+
+2015-01-09 15:28:06 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/video/video-resampler.c:
+ video-resampler: remove check for below zero for unsigned value
+ CLAMP checks both if n_taps is '< 0' and '> max_taps'. n_taps will never be a
+ negative number because it is an unsigned integer. Removing that check and only
+ making sure it isn't set bigger than max.
+ CID 1256558
+
+2015-01-08 10:45:46 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ * gst-libs/gst/video/video-info.c:
+ video: Add support for BT2020 colorspace (UHD)
+
+2015-01-07 15:54:58 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: remove useless debug
+
+2015-01-07 15:52:57 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ video-converter: add options to control chroma resampling
+ Add an option to disable chroma resampling.
+ Improve the matrix option values so that you can choose to use the input
+ or output matrix or disable conversion.
+
+2015-01-02 15:27:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: remove unused enum
+
+2014-12-31 19:40:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: fix silly GQueue iteration code
+
+2014-12-26 20:48:55 +0000 Sam Thursfield <sam@afuera.me.uk>
+
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ Fix documentation that incorrectly says a return value should be freed
+ The gst_discoverer_info_get_missing_elements_installer_details()
+ documentation and annotation says that the return value should be freed
+ with g_strfreev(), but actually it's owned by the GstDiscovereInfo
+ object and should definitely not get freed by the caller as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742006
+
+2014-12-27 14:44:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobasesrc: Explicitly document that buffer-time and latency-time may be ignored
+
+2014-12-26 18:55:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: only clip by duration if end of buffer is ahead of segment
+ It might happen that the timestamp is before the segment and the
+ check would succeed. In this case reducing the duration makes no
+ sense and would lead to broken results.
+
+2014-12-22 22:04:41 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Report our latency properly in live mode
+ While we have no latency at all in theory, any other live source has the
+ duration of one buffer as minimum latency. Do the same in videotestsrc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741879
+
+2014-12-22 22:00:26 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: Report our latency properly in live mode
+ While we have no latency at all in theory, any other live source has the
+ duration of one buffer as minimum latency. Do the same in audiotestsrc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741879
+
+2014-12-22 09:25:04 -0500 Song Bing <b06498@freescale.com>
+
+ * gst-libs/gst/video/gstvideopool.c:
+ * sys/ximage/ximagepool.c:
+ * sys/xvimage/xvimagepool.c:
+ videopool: update video alignment after applying
+ Video buffer pool will update video alignment to respect stride alignment
+ requirement. But haven't updated it to video alignment in configure.
+ Which will cause user get wrong video alignment.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=741501
+
+2014-11-28 14:36:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: get the internal time before the clock reset
+ Otherwise calls to get the clock time might change its internal state
+ and the internal/external time for calibration get unbalanced leading to
+ a clock jump
+ https://bugzilla.gnome.org/show_bug.cgi?id=740834
+
+2014-12-22 11:45:53 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * MAINTAINERS:
+ MAINTAINERS: Update my mail address
+
+2014-12-22 11:38:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ video{en,de}coder: Call reset() before the start() vfunc
+ This makes sure that the element is in the same state before start() is called
+ the very first time and every future call after the element was used already.
+ Also it ensure that we always have a clean state before start(), cleaned the
+ same way in every case.
+
+2014-12-22 11:36:58 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: Call reset() before the start() vfunc to guarantee a clean state
+ The same was done already in the decoder, and we cleaned some state just above
+ manually that would also be taken care of by reset().
+ This makes sure that the element is in the same state before start() is called
+ the very first time and every future call after the element was used already.
+
+2014-12-22 11:33:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ video{en,de}coder: Reset the codec after calling the stop() vfunc
+ The stop() vfunc might mess with some of our fields we have just
+ reset, which could cause memory leaks or invalid state taken over
+ to later.
+ Also the stop() vfunc, or anything called until it from another thread,
+ might want to be able to use the fields that were just resetted and
+ become confused because of that.
+ In the decoder we already had a workaround for things like this happening,
+ this workaround is not needed anymore.
+
+2014-12-22 10:45:37 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobase{sink,src}: Don't hold the object lock while calling create_ringbuffer() vfunc
+ The implementation of that vfunc might want to use the object lock for
+ something too. It's generally not a good idea to keep the object lock while
+ calling any function implemented elsewhere.
+ Also the ringbuffer can only be NULL at this point, remove a useless if block.
+ And in the sink actually hold the object lock while setting the ringbuffer on
+ the instance. Code accessing this is expected to use the object lock, so do it
+ here ourselves too.
+
+2014-12-18 13:24:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: Error out early if we observe an invalid audio format
+
+2014-12-18 13:22:17 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Also handle invalid block aligns for raw audio
+ Fixes audio playback of
+ http://demo.archermind.com/Test%20Sample/Video/MPEG%204/Divx3/Low-Motion/576-320.avi
+ Audio and video together is still broken because of other issues.
+
+2014-12-18 10:57:13 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Fix private header include/dist
+ We want to dist it, but we don't want to install it.
+ Fixes make dist/distcheck
+
+2014-12-18 10:53:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From ef1ffdc to f2c6b95
+
+2014-12-17 21:52:13 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: plug ref leak of template caps
+ the pad template caps is already a new ref. No need to copy.
+
+2014-12-17 19:14:38 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ video: audio: fix GI annotations for proxy caps function
+ Add the annotations to parameters that can be null and also for stating
+ the ownership of the returned caps
+
+2014-12-17 15:21:48 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: tests for caps query implementation
+ Copied from videodecoder tests and updated to audio features
+
+2014-12-17 15:21:16 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ * win32/common/libgstaudio.def:
+ audiodecoder: expose getcaps virtual function
+ Allows subclasses to do custom caps query replies.
+ Also exposes the standard caps query handler so subclasses can just
+ extend on top of it instead of reimplementing the caps query proxying.
+
+2014-12-16 18:36:57 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: implement caps and accept-caps queries
+ Allows decoders to proxy downstream restrictions on caps.
+ Also implements accept-caps query to prevent regressions caused by the
+ new fields on the return of a caps query that would cause the accept-caps
+ to fail as it uses subset caps comparisons
+
+2014-12-16 11:13:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/audio/gstaudioutilsprivate.c:
+ * gst-libs/gst/audio/gstaudioutilsprivate.h:
+ audioencoder: refactor getcaps proxy function to be reusable
+ Makes the audioencoder's getcaps function that proxies downstream
+ restriction available to other elements in the audio module to use it
+
+2014-12-17 14:18:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * tests/check/libs/videodecoder.c:
+ * win32/common/libgstvideo.def:
+ videodecoder: expose getcaps virtual function
+ Allows subclasses to do custom caps query replies.
+ Also exposes the standard caps query handler so subclasses can just
+ extend on top of it instead of reimplementing the caps query proxying.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-15 18:46:21 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: accept-caps should only require fields from the template
+ With the new caps query results the caps returned might have extra fields
+ that are not required by the decoder (framerate for image decoders) and it
+ causes a regression making, for example, jpegdec reject caps that don't
+ have framerates.
+ The accept-caps implementation will do 2 checks:
+ 1) Do subset check with the template caps, making sure all the required
+ fields that are present on the template are present on the received caps.
+ 2) Do a intersection check with the result of a caps query, making sure
+ that downstream can accept the fields in the received caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-09 16:08:12 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideoutilsprivate.c:
+ videoutils: proxy filter when doing a caps query downstream
+ Allows downstream to use the filter and possibly reduce caps complexity
+ to speed up negotiation
+ https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-09 16:05:27 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideoutilsprivate.c:
+ videoutils: return empty if the element has no possible allowed caps
+ Instead of returning the template caps and having a failure happen
+ later because there are no possible caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-08 16:33:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/gstvideoutilsprivate.c:
+ * gst-libs/gst/video/gstvideoutilsprivate.h:
+ * tests/check/libs/videodecoder.c:
+ videodecoder: implement caps query
+ Refactor the encoder's caps query proxying function to a common place
+ and use it in the videodecoder to proxy downstream restrictions.
+ The new function is private to the gstvideo lib.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-17 12:01:19 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: require release version of orc now that there is one
+
+2014-12-16 12:57:55 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ ximagesink: clear src and dest rectangles
+ Now that the center function also takes into account the x and y
+ coordinates of the dest rectangle, better clear all the fields before
+ using them.
+
+2014-12-16 12:10:53 +0100 Song Bing <b06498@freescale.com>
+
+ * gst-libs/gst/video/gstvideopool.c:
+ * sys/ximage/ximagepool.c:
+ * sys/xvimage/xvimagepool.c:
+ videopool: update buffer size after video alignment
+ Update the new buffer size after alignment in the pool configuration
+ before calling the parent set_config. This ensures that the parent knows
+ about the buffer size that we will allocate and makes the size check
+ work in the release_buffer method.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=741420
+
+2014-12-15 20:57:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.h:
+ * gst-libs/gst/audio/gstaudiobasesrc.h:
+ audiobasesrc/sink: Add _CAST macros
+
+2014-12-15 14:10:17 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/video/gstvideosink.c:
+ * tests/check/libs/video.c:
+ video: Fix non-default usage of gst_video_sink_center_rect
+ Make sure we take into account non-0 x/y destination rectangles
+
+2014-12-15 12:12:44 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/examples/playback/playback-test.c:
+ examples: improve playback-test help text a little
+ And allow pipeline type to be specified as string.
+
+2014-12-15 10:35:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.h:
+ pango: Add license/copyright header to header file
+
+2014-12-15 09:45:43 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ Revert "decodebin: Only emit the drain signal for the main decode chain, not any subchains"
+ This reverts commit a391dfe17f1a325f60e1d51a6d40c1a68eb196de.
+ It breaks gapless playback: https://bugzilla.gnome.org/show_bug.cgi?id=740045
+
+2014-12-09 03:18:37 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Fill gap events
+ https://bugzilla.gnome.org/show_bug.cgi?id=741281
+
+2014-12-10 16:10:58 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audio: Add error handling to gst_audio_decoder_drain()
+ https://bugzilla.gnome.org/show_bug.cgi?id=740686
+
+2014-12-13 16:14:49 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: Fix redundant definitions compiler warning
+ gstaudioclock.c:51:31: error: redundant redeclaration of 'gst_audio_clock_init' [-Werror=redundant-decls]
+ G_DEFINE_TYPE (GstAudioClock, gst_audio_clock, GST_TYPE_SYSTEM_CLOCK);
+ gstaudioclock.c:51:31: error: redundant redeclaration of 'gst_audio_clock_class_init' [-Werror=redundant-decls]
+ G_DEFINE_TYPE (GstAudioClock, gst_audio_clock, GST_TYPE_SYSTEM_CLOCK);
+
+2014-12-13 16:04:40 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: No need to get the parent class in class_init, G_DEFINE_TYPE does that for us
+
+2014-12-13 16:01:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: Use G_DEFINE_TYPE instead of a custom get_type() function
+
+2014-12-12 08:32:15 -0800 Zaheer Abbas Merali <zaheermerali@gmail.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: fix spelling of word in comment
+
+2014-12-12 14:59:49 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/rtpbasedepayload.c:
+ tests: rtpbasepayload: fix indentation
+
+2014-12-12 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: fix indentation
+
+2014-12-12 14:56:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: fix broken refcounting in unit test
+ The set_format vfunc does not pass ownership of the caps
+ to the decoder, so we mustn't unref the caps there.
+ gst_event_new_caps() does not take ownership of the caps
+ passed, so we must unref the caps afterwards.
+ Fixes leaks when running test in valgrind in 1.4 branch.
+
+2014-12-12 10:02:43 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ video: Update disted orc source files
+
+2014-12-12 10:01:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ Revert "video-converter: Fix compiler warning because of missing prototype of non-static function"
+ This reverts commit 406f32a9468c837a4d71f988de10dc2198a8edc9.
+ The problem was apparently that my video-orc.h was not updated and did not
+ include the prototype for that function. Only a "make clean" caused it to
+ be regenerated.
+
+2014-12-12 09:51:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Fix compiler warning because of missing prototype of non-static function
+ video-converter.c:838:1: error: no previous prototype for function
+ '_custom_video_orc_matrix8' [-Werror,-Wmissing-prototypes]
+
+2014-12-09 22:47:31 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: do not use fixed caps on source pad
+ decoders can change the caps on their source pads, so they don't
+ use fixed caps. Having fixed caps can cause renegotiation issues.
+
+2014-12-09 22:46:42 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: do not use fixed caps on source pad
+ decoders can change the caps on their source pads, so they don't
+ use fixed caps. Having fixed caps can cause renegotiation issues.
+
+2014-12-11 13:45:38 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Do not mix up stream type when getting stream combiner element
+ We were always returning the video stream combiner whatever stream type
+ combiner was wanted.
+
+2014-12-10 13:23:23 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin2: always unref the combiner sinkpad when removing the srcpad
+ Create a function to do the pad cleanup of the GstSourceCombine struct
+ and use it to not forget to also cleanup the sink pad and fix a memory
+ leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741198
+
+2014-12-10 16:42:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ video-orc: make RGB pack/unpack faster
+ Avoid all the merging and splitting and use a pair of shifts and or
+
+2014-12-11 01:53:15 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.h:
+ videodecoder: Add GST_VIDEO_DECODER_CAST macro
+ It's used in some macros already, so let's make it exist.
+
+2014-11-25 13:31:48 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: No remove child if destroyed.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740730
+
+2014-12-08 18:53:35 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/icles/test-reverseplay.c:
+ reverse-play: fix seek to end when starting reverse
+ Start reverse playback by actually seeking to the end of
+ the file.
+
+2014-12-06 21:02:37 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: set bits and format after conversion
+ Update the current format, bits and pstride.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=741187
+
+2014-12-05 22:09:45 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: free dither_lines
+ Avoid a memory leak
+
+2014-12-05 18:16:53 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ Bump ORC requirement to 4.22.1
+ We now depend on git commit f1cfa5, "orcc: allow setting custom
+ backup function"
+
+2014-12-05 14:51:28 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: use custom backup function
+ Use the new orc feature to set a custom backup function.
+
+2014-12-05 12:18:42 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: improve matrix8 function
+ Avoid using a constant.
+ Avoid doing saturated adds, results are not supposed to overflow here.
+ Rework the C backup function a little in preparation for custom backup
+ functions in ORC.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=741015
+
+2014-11-28 15:06:27 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * tests/check/libs/audiodecoder.c:
+ audiodecoder: Push pending events before sending EOS.
+ Segments are added to the pending events, and pushing a segment
+ is mandatory before sending EOS.
+ + Adds a test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740853
+
+2014-11-27 05:53:20 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix seeking before the first frame.
+ The previous code was setting keytarget to target
+ to make sure the keyframe found for each pad was
+ indeed before the target.
+ Then if target == keytarget, it assumed a keyframe had been
+ found, which was not the case if target was before the first frame
+ in the file.
+ This patch checks that a keyframe was indeed found, and if not
+ seeks to 0, without bisecting again.
+ Assuming default gst qa assets in $HOME/gst-validate
+ seek_before_first_frame.scenario:
+ description, seek=true, handles-states=true
+ pause, playback-time=0.0
+ seek, playback-time=0.0, start=0.0, flags=accurate+flush
+ seek, playback-time=0.0, start=0.01, flags=accurate+flush
+ seek, playback-time=0.0, start=0.1, flags=accurate+flush
+ GST_DEBUG=*theoradec*:2 gst-validate-1.0 playbin \
+ uri=file://$HOME/gst-validate/gst-qa-assets/medias/ogg/vorbis_theora.0.ogg \
+ --set-scenario seek_before_first_frame.scenario
+ https://bugzilla.gnome.org/show_bug.cgi?id=741097
+
+2014-10-08 08:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Only check sinks which are in >= GST_STATE_READY
+ Otherwise we endup with bogus caps intersection (from the pad template
+ caps and not from what the actual hardware/device supports)
+ https://bugzilla.gnome.org/show_bug.cgi?id=738131
+
+2014-12-03 10:15:18 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: fix chroma resampling check
+ Decide if we need chroma resampling by checking if we have a progressive
+ or interlaced chroma resampler.
+
+2014-12-03 10:14:34 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: only do dithering when needed
+ Only do dithering when one of the quantizers is > 1.
+
+2014-12-02 15:58:00 -0500 Chad <crh184@psu.edu>
+
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Use gst_util_uint64_scale_int_round()
+ Using gst_util_uint64_scale_int() causes slight drift
+ which accumulates over time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741045
+
+2014-12-02 13:39:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * win32/common/libgstvideo.def:
+ defs: update defs file
+
+2014-12-02 11:51:19 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoconvert/gstvideoconvert.c:
+ * gst/videoconvert/gstvideoconvert.h:
+ videoconvert: add dither-bits option
+ Fix the dither option.
+ Add a new option to set the quantizer
+
+2014-12-02 11:48:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add where orc functions could go
+ Add the disabled orc functions in #if 0 lines for when we can enable
+ them.
+
+2014-12-02 11:40:59 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ * gst-libs/gst/video/video-dither.c:
+ video-converter: add dithering
+ Use the new dither object to perform dithering.
+ Add option to select dithering method.
+ Add option to quantize to a specific value
+
+2014-12-02 11:39:42 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add palette when needed
+
+2014-12-02 11:32:28 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video-dither.c:
+ * gst-libs/gst/video/video-dither.h:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video.h:
+ video-dither: add video dither helper object
+ Add a new object that implements various dithering methods.
+
+2014-12-01 22:28:52 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tools/gst-play.c:
+ gst-play: do not set system's volume to 100% by default
+ Only change the volume if requested
+
+2014-12-01 09:50:24 +0100 Thomas Klausner <wiz@danbala.tuwien.ac.at>
+
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsa: Use EPIPE instead of ESTRPIPE if the latter does not exist
+ NetBSD does not have ESTRPIPE.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740952
+
+2014-11-28 14:28:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/alsa/gstalsasrc.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/vorbis/gstvorbisdec.c:
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ * gst-libs/gst/tag/id3v2frames.c:
+ * gst-libs/gst/video/navigation.c:
+ * gst-libs/gst/video/video-converter.c:
+ * gst/adder/gstadder.c:
+ * gst/encoding/gstencodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/tcp/gstmultihandlesink.c:
+ * gst/tcp/gstmultioutputsink.c:
+ * tests/examples/playback/playback-test.c:
+ * tests/examples/seek/jsseek.c:
+ * tools/gst-discoverer.c:
+ Don't compare booleans for equality to TRUE and FALSE
+ TRUE is 1, but every other non-zero value is also considered true. Comparing
+ for equality with TRUE would only consider 1 but not the others.
+
+2014-11-16 15:54:56 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-profile.h:
+ * gst/encoding/gstencodebin.c:
+ * win32/common/libgstpbutils.def:
+ encodebin: Add a way to disable caps renegotiation for output stream format
+ In some cases, the user might want the stream outputted by encodebin to
+ be in the exact same format during all the stream. We should let the
+ user specify when this is the case. This commit add some API in the
+ GstEncodingProfile to determine whether the format can be renegotiated
+ after the encoding started or not.
+ API:
+ gst_encoding_profile_set_allow_dynamic_output
+ gst_encoding_profile_get_allow_dynamic_output
+ https://bugzilla.gnome.org/show_bug.cgi?id=740214
+
+2014-11-28 13:31:39 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: remove libs/video and videoconvert test from valgrind blacklist
+ Seem to work fine.
+
+2014-11-28 13:29:37 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: don't run orc/* tests under valgrind
+ They just seem to blow up for some reason that needs investigating.
+
+2014-11-28 13:11:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/tag/gsttagmux.c:
+ tagmux: fix criticals when there are no tags at all
+
+2014-11-21 01:47:35 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/icles/test-reverseplay.c:
+ test-reverseplay: Use uridecodebin for input
+ Work with any installed URI handler
+ Add some more debug output
+
+2014-11-28 10:27:28 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: Mapping a frame with inconsistent values between GstVideoMeta and GstVideoInfo is a bug
+ It will cause the frame to be initialized with inconsistent values that then
+ later can cause crashes or any other kind of interesting and hard to debug
+ bugs.
+
+2014-11-27 17:10:31 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 7bb2bce to ef1ffdc
+
+2014-11-27 15:28:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-blend.c:
+ video-blend: make use of x offset when unpacking overlay image pixels
+ Now that it's implemented we can use it, which is a minor
+ optimisation when the image to overlay gets cropped on the
+ left.
+
+2014-11-27 15:04:12 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: sprinkle some 'restrict' keywords in pack/unpack functions
+ In cases where we just call orc directly this is somewhat
+ superfluous, but let's do it anyway for consistency. In
+ other cases the compiler can hopefully use this to optimise
+ memory access a little.
+
+2014-11-27 13:01:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: handle x offset in unpack
+ Add support for x offset in almost all unpack methods.
+ Fix naming of source and dest pixels.
+ Add const to source pixels.
+
+2014-11-27 10:51:58 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: improve unpack i420
+ unpack_i420 does not need extra code to handle odd widths, the orc code
+ already handles it fine.
+
+2014-11-27 09:45:07 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: use old property name
+ Unbreak ABI by changing to the old property name again.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740798
+
+2014-11-25 13:39:07 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Analyze source pad before setting to PAUSED for 'simple demuxers'
+ Before we were setting them to PAUSED and (much) later connecting to
+ their source pad caps notify signal.
+ There was a race where that demuxer was pushing a caps and later a buffer
+ on its source pad when we were not even connected to its source pad caps notify
+ signal leading to decodebin missing the information and not keeping on
+ building the pipeline on CAPS event thus the demuxer was posting an ERROR
+ (not linked) message on the bus. This need to be done for 'simple
+ demuxers' because those have one ALWAYS source pad, not like usual demuxers
+ that have several dynamic source pads.
+ A "simple demuxer" is a demuxer that has one and only one ALWAYS source
+ pad.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740693
+
+2014-11-25 16:46:50 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Take STREAM_LOCK before sending sticky events.
+ There was a race where:
+ 1) we would put the element to PAUSED
+ 2) It would get data sent to it from upstream
+ 3) It would thus send caps
+ 3) caps_notify_cb would continue autoplugging
+ 4) caps would flow downstream, the last pad would get exposed
+ 5) we were still not done sending the sticky events
+ Taking the stream lock on the new element's sinkpad and only
+ releasing it when sticky events have all been sent prevents
+ the caps from reaching the source pad of the element before
+ we're all set.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740694
+
+2014-08-06 19:31:25 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: detect mp4 common file format variant
+ Used e.g. by UltraViolet.
+
+2014-11-25 22:01:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/alsa/gstalsasrc.c:
+ alsasrc: debug message fixes
+ In the same vein as 74e9640a.
+
+2014-11-25 17:42:07 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scale: combine adds when max_taps equals combine size
+ When the amount of pixels/lines matches the amount we can combine,
+ combine the adds and multiplies and do the scale as a separate
+ operation.
+
+2014-11-25 17:25:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: combine scaling operations
+ Combine add and scale of multiple lines/pixels to reduce the amount of
+ read and writes to temporary memory.
+
+2014-11-25 14:45:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pango/gsttimeoverlay.c:
+ * ext/pango/gsttimeoverlay.h:
+ timeoverlay: add "time-line" property
+ So we can also show running time or stream time, not just the
+ buffer time stamps.
+
+2014-11-25 11:54:51 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: add property to do scaling after gamma-decode
+
+2014-11-25 11:28:42 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: add more scaling filters
+ Adjust the filter parameters so that they use the same number of taps
+ and method as the old ones.
+ Add some new filters
+
+2014-11-25 10:36:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-resampler.c:
+ video-resampler: remove print
+
+2014-11-25 10:32:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-resampler.c:
+ video-resampler: improve variable taps
+ Improve quality of variable taps on all methods by reusing the lanczos
+ parameters where possible.
+
+2014-11-25 09:11:31 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-resampler.c:
+ video-resampler: Fix lanczos parameters for variable taps
+ when using variable taps and when we are limiting the number of taps,
+ recalculate the lanczos parameters to match the clamped value.
+ Set the max number of taps to 128
+
+2014-11-25 11:38:34 +0300 Andrei Sarakeev <sarakusha@gmail.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Reset mute property of the sink to playsink's value when setting up the audio chain
+ Otherwise the following can happen:
+ 1. set mute=true
+ 2. play media1 (Ok)
+ 3. play media without audio (audiochain removed)
+ 4. play media2 (audiochain created, mute=*false*)
+ https://bugzilla.gnome.org/show_bug.cgi?id=740675
+
+2014-11-25 11:38:34 +0300 Andrei Sarakeev <sarakusha@gmail.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ discoverer: fix typo in header file
+ https://bugzilla.gnome.org/show_bug.cgi?id=740675
+
+2014-11-25 09:08:18 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for audio/x-audible
+
+2014-11-25 01:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: improve 'audible' audio typefinder a little
+ Don't return NEARLY_CERTAIN just based on 4 bytes.
+ Also change media type to audio/x-audible.
+ https://bugzilla.gnome.org/show_bug.cgi?id=715050
+
+2013-11-23 11:36:43 +1000 Jonathan Matthew <jonathan@d14n.org>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: add audio/audible typefinder
+ https://bugzilla.gnome.org/show_bug.cgi?id=715050
+
+2014-06-16 11:46:18 +0200 Branislav Katreniak <bkatreniak@nuvotechnologies.com>
+
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsa: Change the log messages in xrun_recovery() from DEBUG to WARNING
+ xrun_recovery() runs when there is an error
+ https://bugzilla.gnome.org/show_bug.cgi?id=740615
+
+2014-11-24 12:47:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: keep track of required temp lines
+ Make a small object to hold a pool of allocated temp lines.
+ Keep track of how many temp lines each conversion stage needs and use
+ this to allocate just enough temp lines from the temp lines object. from
+ the temp lines object.
+
+2014-11-24 12:45:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: use err line in fastpath
+ Use the error line for temporary storage in the fastpath so that we
+ don't have to allocate any other temp lines.
+
+2014-11-22 21:51:33 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: don't complain about PTS != DTS on keyframes
+ It is valid for streams with b-frames
+ https://bugzilla.gnome.org/show_bug.cgi?id=740556
+
+2014-11-21 16:06:54 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: handle mixed interlaced
+ When dealing with mixed interlaced, setup a scaler and chroma-resampler
+ for both interlaced and progressive frames and switch between them
+ depending on the interlace mode of the input frame.
+
+2014-11-21 16:04:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Cleanup options parsing
+ Cleanup option parsing
+ Add some debug
+
+2014-11-21 15:59:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: there is no need to apply x offset to temp lines
+
+2014-11-21 15:58:34 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: ensure both fields have the same number of taps
+
+2014-11-21 11:15:04 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: rework the options a little
+ Rework the options a little to make it nicer to set defaults.
+
+2014-11-21 11:12:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-resampler.c:
+ * gst-libs/gst/video/video-resampler.h:
+ video-resampler: add option to limits taps
+ Add an option to limit the number of taps to use in automatic mode. The
+ problem is that for lanczos, we might use more taps than what we can
+ handle with the current precision.
+ Rework the other options a little to make it nicer to set defaults.
+
+2014-11-20 18:20:00 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video: update orc files
+
+2014-11-20 15:53:23 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * win32/common/libgstvideo.def:
+ win32: Update defs file
+
+2014-11-19 21:18:04 +0900 Hyunjun Ko <zzoonis@gmail.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtspconnection: fix warning on param name mismatch
+ https://bugzilla.gnome.org/show_bug.cgi?id=740013
+
+2014-11-18 00:04:59 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-reverseplay.c:
+ tests: Add reverse playback verification test
+ Plays a requested URI forward to EOS, then backward and
+ checks that the same timestamp range(s) are covered.
+
+2014-11-12 15:23:37 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Operate in a zero-latency mode if drop-only is set to TRUE
+ There's no reason why we would have to wait for the next buffer to decide
+ whether to output the current one or not. We just have to check if the
+ current one is earlier than our expected next time, which is the previous
+ frame timestamp plus the expected frame duration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740018
+
+2014-11-19 14:38:03 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Use correct enum, GstVideoFormat instead of GstFormat
+
+2014-11-19 13:25:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: fix size check
+ Add some debug, fix size check that decides what scaling to do first and
+ when to do conversion.
+
+2014-11-19 12:53:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: avoid primaries conversion when asked
+ Don't do conversion between primaries when the option is disabled.
+ Only do some matrix code when needed.
+
+2014-11-19 12:41:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-info.c:
+ video-info: add a note about subsampled formats
+ Add a note about gst_video_info_set_format() and interlaced formats.
+
+2014-11-19 12:05:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-info.c:
+ video-info: handle interlaced size correctly
+ Refactor GstVideoInfo init, make function to set default colorimetry.
+ Call fill_planes after we configure the GstVideoInfo with parameters
+ from the caps.
+ The size of the chroma planes for interlaced vertically subsampled
+ formats needs to be rounded up to 2, we have 2 fields with each
+ the same anount of chroma lines.
+
+2014-11-19 12:04:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-color.c:
+ video-color: return FALSE on unparsable colorimetry
+
+2014-11-19 09:40:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: handle unpack interlaced subsampled formats
+ For interlaced vertically subsampled formats the check for even lines
+ needs to take into account the two fields.
+
+2014-11-19 09:39:32 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix interlaced shift
+
+2014-11-19 09:30:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: keep a small backlog of lines
+ Allow lines to jump backwards slightly, usefull for interlaced content.
+
+2014-11-19 09:28:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ video-chroma: Fix interlaced chroma resampling
+ Use the interlaced flag to select the right resampler.
+
+2014-11-18 16:36:08 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-resampler.c:
+ * gst-libs/gst/video/video-scaler.c:
+ video: add some more debuging
+
+2014-11-18 16:35:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix interlacing some more
+ Use the right phase.
+ Take the right lines from interlaced content.
+
+2014-11-18 12:53:06 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ video-converter: fix dither method
+
+2014-11-18 12:52:27 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: fix some leaks
+ And remove some unused fields.
+
+2014-11-18 12:20:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ video-converter: add support for gamma and primaries
+ Keep only 1 structure with all matrix information.
+ Add structure to hold gamma information.
+ Add more options to control gamma, primaries and color matrix handling.
+ Add functions to compute transformations to and from XYZ and use this
+ to convert between primaries.
+ Merge gamma into the convert to and from RGB stage.
+ Fix border val.
+ Simplify the fastpath table, remove unused fields, add some more checks.
+
+2014-11-18 11:09:40 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ video-color: add method to get primaries info
+
+2014-11-18 11:08:10 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-info.c:
+ video-color: fix default 601 primaries
+
+2014-11-18 11:06:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix interlaced taps setup
+
+2014-11-14 09:15:22 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ * gst-libs/gst/video/video-info.c:
+ video-color: make sRGB colorimetry the default for RGB
+
+2014-11-13 12:03:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: split YUV to and from RGB conversions
+ Prepare for doing full gamma corrected conversion and scaling by first
+ splitting the conversions from and to RGB into separate steps.
+ split scaling in downscaling and upscaling steps to be performed before
+ and after conversion respectively.
+
+2014-11-13 12:02:07 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: don't convert too much
+ because we do conversion after downscaling we only need to convert the
+ smallest width.
+
+2014-11-13 12:00:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: add orc splat functions to draw border
+
+2014-11-05 21:52:44 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ Revert "basetextoverlay: Fix segfault when overlay outside the frame"
+ This is not correct. overlay->silent is a property and we
+ should not just flip the property forever because one text
+ we render is outside of the frame. The next one might not
+ be, the positioning properties can be changed after all.
+ The lower layers should handle clipping, and now do.
+ This reverts commit 1cc311156cc3908d1d9888fbcda67305fc647337.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738984
+ https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-05 21:46:47 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ Revert "basetextoverlay: segfault when xpos >= video size"
+ This is not right, even if it might avoid a crash. We don't
+ want to just set xpos/ypos to 0 in those cases. Clipping
+ should be done properly, see bug #739281 for that.
+ This reverts commit 900d0267d511e9553eec44d948d7e33ead7dc903.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738984
+ https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-16 23:26:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-blend.c:
+ video-blend: minor optimisation
+ Only need to run matrix on those pixels which
+ will actually be used.
+
+2014-11-16 19:28:54 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/Makefile.am:
+ * tests/icles/test-overlay-blending.c:
+ tests: make overlay blending test slightly less boring
+
+2014-11-16 16:34:31 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-blend.c:
+ video-blend: fix clipping of overlay images on the left
+ Fix clipping of images that are partially left of the video
+ surface, they would get clipped on the right side instead of
+ the left side, because the video unpack functions currently
+ ignore the x offset parameter. Work around that until that
+ is implemented.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-16 16:31:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-blend.c:
+ video-blend: fix allocation of temp src line for wide sources
+ Fix allocation of temporary source line buffers for source
+ images that are wider than the video overlay surface.
+
+2014-11-16 01:34:09 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-overlay-blending.c:
+ tests: add visual overlay composition blending test
+ Shows visual result of blending a logo on top of
+ a video surface, esp. when the logo is partially
+ outside of the video surface and needs to be
+ clipped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-16 01:32:55 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/video.c:
+ tests: fix leak in video unit test
+
+2014-11-10 16:36:35 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst-libs/gst/video/video-blend.c:
+ video-blend: fix blending of rectangles partially or fully outside of the video
+ In case of overlay being completely or partially outside
+ the video frame, the offset calculations are not right,
+ which resulted in the overlay not being displayed as
+ expected, or crashes due to invalid memory access.
+ When the overlay rectangle is completely outside,
+ we need not render the overlay at all.
+ For partial display of overlay rectangles, src_yoff
+ was not being calculated, hence it was always clipping
+ the bottom half of the overlay, By calculating the
+ src_yoff, now the overlay is clipped properly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-10 12:12:42 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * tests/check/libs/video.c:
+ tests: video: add video blend test
+ Add test to check rendering of overlays of different sizes
+ that are completely or partially outside the video surface.
+ Once the overlay is blended to the video, verify if the
+ position of the blended overlay is as expected, by comparing
+ the pixels of the blended video with the expected values.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-15 23:15:06 +0000 Tim-Philipp Müller <tim@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-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:
+ docs: update to git
+
+2014-11-15 23:13:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/gio/gstgiostreamsink.c:
+ * gst/gio/gstgiostreamsrc.c:
+ * gst/playback/gstplaybin2.c:
+ docs: fix some gtk-doc warnings
+ Deprecated entities found in documentation for xyz:Long_description
+ .
+
+2014-11-12 09:57:38 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: take offset into account when unpacking
+ When we can directly take the input line from the source frame when
+ unpacking, also take into account the x offset.
+
+2014-11-12 09:57:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add some notes
+
+2014-11-11 16:19:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * win32/common/libgstvideo.def:
+ defs: update defs and docs
+
+2014-11-11 16:11:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ * tests/check/libs/video.c:
+ video-color: add gamma encode/decode functions
+ Add functions to encode and decode gamma.
+ Add unit test to check that encode and decode are eachothers inverse
+ and that the limits are respected.
+
+2014-11-10 14:53:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/libs/video.c:
+ test: add scaling test
+ Sort pack and unpack performance measurements
+
+2014-11-10 12:01:48 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-orc: update disted file
+ and disable one failing function
+
+2014-10-24 17:08:43 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoscale/Makefile.am:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ * gst/videoscale/vs_fill_borders.c:
+ * gst/videoscale/vs_fill_borders.h:
+ * gst/videoscale/vs_image.c:
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_lanczos.c:
+ * gst/videoscale/vs_scanline.c:
+ * gst/videoscale/vs_scanline.h:
+ * tests/check/Makefile.am:
+ videoscale: port to new API
+
+2014-11-10 11:40:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ video-orc: use faster saturating conversions
+ saturating conversions are generally faster.
+
+2014-11-07 15:45:04 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-chroma: add ORC version of UP_H2_CS
+ It is however slower than the C version and thus disabled.
+
+2014-11-09 14:44:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for Apple Core Audio Format
+ https://bugzilla.gnome.org/show_bug.cgi?id=739840
+
+2014-11-09 12:53:32 +0100 Peter G. Baum <peter@dr-baum.net>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: recognize Apple Core Audio Format
+ (CAF) Specification 1.0
+ https://bugzilla.gnome.org/show_bug.cgi?id=739840
+
+2014-11-09 10:47:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/pipelines/capsfilter-renegotiation.c:
+ capsfilter-renegotiation: Use assertions from libcheck for more information on failures
+
+2014-11-07 12:06:10 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ * tests/check/libs/video.c:
+ video-chroma: ORCify 2x vertical upsampling
+ Make an ORC version of the 2x vertical upsampling code.
+ Improve unit tests, test chroma up and down sampling.
+ memset buffer in conversion to make valgrind happy.
+
+2014-11-06 14:14:22 +0000 William Manley <will@williammanley.net>
+
+ * gst/tcp/gstmultihandlesink.c:
+ * gst/tcp/gsttcpserversink.c:
+ tcpserversink: Don't leak a `GSocket` and a `GInetSocketAddress`
+ when accepting a connection.
+ Discovered by `make check-valgrind` with the new `socketintegrationtest`.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739544
+
+2014-11-03 01:08:27 +0000 William Manley <will@williammanley.net>
+
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/.gitignore:
+ * tests/check/pipelines/tcp.c:
+ tests: Add TCP pipelines test
+ There don't seem to be any unit tests for the socket handling elements. As
+ I am about to attempt some refactorings I've added some basic tests which
+ exercise some of the happy-paths in tcpclientsrc, tcpserversrc,
+ tcpserversink and tcpclientsink. They should let me know if I've caused
+ serious breakage.
+ They are far from exhaustive but are sufficient for me to have caught a few
+ memory-leaks in the existing code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739544
+
+2014-11-06 18:18:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/libs/video.c:
+ tests: add video conversion test
+ Go through all conversions and make a list of performance.
+
+2014-11-06 18:13:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-info.c:
+ video-info: use h-cosited chroma for HD video by default
+
+2014-11-06 18:09:04 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: clamp lines
+
+2014-11-06 16:29:16 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video-orc: update disted files
+
+2014-11-06 16:18:25 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: ORCify 8<->16 conversion
+
+2014-11-06 15:30:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: unpack into the destination when needed
+ Make sure we write into the destination line when we can propose the
+ dest allocator.
+
+2014-11-06 15:29:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add more debug
+
+2014-11-06 15:01:27 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video: Update disted orc files
+
+2014-11-06 13:08:42 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ * gst-libs/gst/video/video-orc.orc:
+ * tests/check/libs/video.c:
+ video-chroma: optimize chroma subsampling a little
+ Combine multiplies in 4x filters.
+ Rename conversion functions to make them nicer in orc.
+ Add ORC versions for various downsampling algorithms
+ Add unit test chroma resampler
+
+2014-11-06 10:43:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/libs/video.c:
+ tests: make pack/unpack test
+ Make a more complete pack/unpack test, check if the image after
+ pack/unpack has the same color and precision, and has correctly
+ duplicated subsampled pixels.
+
+2014-11-06 10:42:09 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/libs/video.c:
+ tests: get the correct number of video formats
+ Make a method to get the number of formats (including the last one).
+
+2014-11-06 09:44:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.h:
+ video-format: update some docs and add a FIXME(2.0)
+
+2014-11-06 09:38:06 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: add range extension to BGR_10XE format
+
+2014-11-06 09:34:59 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-format: fix pack of 4:2:0 formats
+ When packing 4:2:0 formats, we need to take the chroma from the even
+ lines, for the odd lines we only take luminance.
+
+2014-11-06 09:32:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: fix range extension of UYVP
+ We need to shift the top 6 bits to the lower 6 bits
+
+2014-11-06 09:28:06 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ video-chroma: do h subsampling after v subsampling
+ We only need to do the horizontal subsampling on 1 line if we do it
+ after vertical subsampling and we also avoid doing vertical subsampling
+ on unused pixels.
+
+2014-11-06 09:39:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: dist header file needed for ABI checks on powerpc32
+ Fixes 'make check' on debian powerpc32 buildbot:
+ libs/libsabi.c:95:26: fatal error: struct_ppc32.h: No such file or directory
+
+2014-11-05 04:34:44 +0900 Danny Song <danny.song.ga@gmail.com>
+
+ * tests/check/elements/adder.c:
+ test : fix leaks in adder unit test
+ https://bugzilla.gnome.org/show_bug.cgi?id=739640
+
+2014-11-05 11:54:31 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: keep separate lines with border
+ Make separate with a border around them so that we can avoid a memcpy.
+
+2014-11-05 11:52:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: avoid memcpy when not needed
+
+2014-11-05 11:51:44 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: pass output line correctly
+
+2014-11-04 09:30:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: rework the converter to allow more optimizations
+ Rework the converter, keep track of the conversion steps by chaining the
+ cache objects together. We can then walk the chain and decide the
+ optimal allocation pattern.
+ Remove the free function, we're not going to need this anytime soon.
+ Keep track of what output line we're constructing so that we can let the
+ allocator return a line directly into the target image when possible.
+ Directly read from the source pixels when possible.
+
+2014-11-04 11:03:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix temp line allocation
+ We need to allocate the templine with the amount of pixels we are going
+ to handle, which we only know for the vertical resampler when we are
+ asked to resample.
+
+2014-11-04 11:02:49 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix taps in interlaced mode
+
+2014-11-04 11:01:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix phases in interlaced mode
+
+2014-11-04 09:29:58 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ video-orc: fix v_2tap_u16
+
+2014-11-03 16:18:41 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add extra pixels for the border
+ We need extra pixels for the border.
+
+2014-11-03 15:36:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add support for 16bits formats
+ Add scaler functions for 16 bits formats.
+ Rename the scaler functions so that 16bits versions don't look too
+ weird.
+ Remove old unused h_2tap functions
+ Fix v_ntap functions, it was using 1 tap too little.
+
+2014-11-03 15:33:24 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Add support for 16 bits formats
+ Rework the way we track the current state of the video through the
+ different conversion phases and use this to make sure we use the right
+ format and pstride where needed.
+
+2014-10-22 13:37:40 +0100 William Manley <will@williammanley.net>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ docs: gst_dmabuf_allocator_alloc: Improve documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=739545
+
+2014-11-03 10:07:56 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ video-orc: comment out unused function
+ A faster version of 4tap horizontal scaling causes segfaults in ORC
+ presumably because it uses too many registers so disable it to avoid
+ crashing in the ORC tests.
+
+2014-11-02 21:45:30 +0100 Andreas Frisch <fraxinas@opendreambox.org>
+
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: return available factory CAPS instead of ANY on CAPS query
+ https://bugzilla.gnome.org/show_bug.cgi?id=739536
+
+2014-11-03 08:12:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: Fix compiler warning
+ video-scaler.c:151:58: error: implicit conversion from enumeration type
+ 'GstVideoScalerFlags' to different enumeration type
+ 'GstVideoResamplerFlags' [-Werror,-Wenum-conversion]
+ gst_video_resampler_init (&scale->resampler, method, flags, out_size,
+ ~~~~~~~~~~~~~~~~~~~~~~~~ ^~~~~
+
+2014-11-01 20:08:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: Do not use deprecated gtk-doc 'Rename to' tag
+ GObject introspection GTK-Doc tag "Rename to" has been deprecated, changing to
+ rename-to annotation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739514
+
+2014-11-01 14:58:13 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/video-scaler.h:
+ video: fix some g-i / gtk-doc warnings
+
+2014-11-01 14:47:26 +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 functions
+ Fixes build without orc.
+
+2014-11-01 14:28:55 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video-blend.c:
+ video: add video blend helper functions to docs
+ I don't think those were ever meant to be made public,
+ but they are, so we might as well document them.
+
+2014-11-01 13:14:32 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: ORCify vertical ntap function
+
+2014-11-01 12:58:01 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: handle 4tap interlaced
+
+2014-10-31 16:53:06 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video-orc: update dist files
+
+2014-10-31 16:49:43 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add ORC optimized ntap horizontal scalers
+
+2014-10-29 16:28:28 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tests/icles/playback/test.c:
+ * tests/icles/playback/test2.c:
+ * tests/icles/playback/test4.c:
+ tests/playback: quit from main loop
+ Listen for eos and error signal to quit main loop.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739346
+
+2014-10-29 16:26:07 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tests/icles/playback/test2.c:
+ * tests/icles/playback/test4.c:
+ tests/playback: correct state change checking
+ Correct the test apps check if result of state change is not failure as the
+ state change can happen async
+ https://bugzilla.gnome.org/show_bug.cgi?id=739346
+
+2014-10-31 22:52:43 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video: Update disted orc files for new functions.
+ Fixes the build when building without ORC
+
+2014-10-31 11:07:06 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: align offsets to subsampling
+ Only apply an offset that is a multiple of the subsampling. To handle
+ arbitrary offsets in the future, we need to be able to chroma-resample
+ part of the borders.
+
+2014-10-31 10:38:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: clamp output lines
+
+2014-10-31 10:34:46 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: add alignment checks
+ Some of the ORC functions need specific alignment
+
+2014-10-31 10:33:42 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix offset check
+
+2014-10-30 18:41:01 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: also chroma up/downsample when scaling
+
+2014-10-30 18:40:43 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: clamp input lines correctly
+
+2014-10-30 23:53:39 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: fix build without orc
+ https://bugzilla.gnome.org/show_bug.cgi?id=739433
+
+2014-10-30 17:30:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add border color
+
+2014-10-30 16:57:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ video-converter: add support for src/dest regions
+ Add support for cropping the source and placing the converted image
+ into a rectangle in the destination frame.
+ Add an option to add a border and border color.
+
+2014-06-10 09:33:40 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opusenc: update output segment stop time to match clipped samples
+ This will let oggmux generate a granpos on the last page that properly
+ represents the clipped samples at the end of the stream.
+
+2014-06-05 14:50:15 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/vorbis/gstvorbisenc.c:
+ vorbisenc: push an updated segment stop time when we know it
+ When encoding, libvorbis will tell us how many samples are encoded
+ in the buffer it returns. This number may be less than the maximum
+ of samples in the block, if this is the last packet. In we have no
+ segment end time, we set it to the end time of that last sample to
+ tell downstream that the buffer contains less samples.
+
+2014-06-05 14:54:31 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: set correct granpos on last page when samples are clipped
+ Samples may be clipped at the end, and this is conveyed by a
+ granulepos that's smaller than it would otherwise be. Use the
+ segment stop time to detect this, and calculate the right
+ granulepos.
+
+2014-06-05 11:26:08 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: fix last buffer timestamp when samples are clipped
+ The end of a stream can be clipped by setting the granulepos of
+ the last page to a lower value that it otherwise would be.
+
+2014-10-30 14:48:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/libs/video.c:
+ tests: fix test
+
+2014-10-03 12:42:46 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * tools/gst-discoverer.c:
+ gst-discoverer: error out on failure to copy
+ This should not really fail, but let's check return value
+ anyway as it guards against future changes.
+ Coverity 1135731
+
+2014-10-03 12:28:30 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: add a const where appropriate
+
+2014-10-03 12:08:05 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: remove unneeded test
+ We've already bailed out if we have less than 5 bytes.
+ Coverity 1226441
+
+2014-10-30 11:33:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ Update libgstvideo.def for resampler -> video_resample renaming
+
+2014-10-30 11:46:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add more ORC functions
+ Add the old ORC functions for nearest and linear. Label them as Low
+ quality because they are not as accurate but ORC lacks opcodes to
+ express this for now.
+
+2014-10-30 11:43:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-resampler.c:
+ * gst-libs/gst/video/video-resampler.h:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/video-scaler.h:
+ video-scaler: rename resampler to video-resampler
+ Prefix the resampler with video-. It we would like to reuse the
+ resampler for audio later, we can copy/move it and deprecate this
+ one.
+
+2014-10-29 17:38:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/video-scaler.h:
+ video-scaler: remove color range argument
+ We just need to clip to the format limits, if there is extra headroom in
+ the range we can use that without problems.
+
+2014-10-29 17:14:51 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * win32/common/libgstvideo.def:
+ defs: update defs
+
+2014-10-29 16:20:56 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add ORC optimized versions
+ Add ORC optimized versions of 2 and 4tap vertical scaling. Provide
+ a high quality 12 bits and a low quality 6 bits version.
+
+2014-10-29 16:13:02 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: add precision to make_s16_taps
+
+2014-10-29 13:19:00 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: copy config fields
+ When setting a new config, copy all the fields into our own config and
+ not only the ones we know about.
+
+2014-10-29 13:17:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/resampler.c:
+ * gst-libs/gst/video/resampler.h:
+ * gst-libs/gst/video/video-scaler.c:
+ resampler: make offset/phase/n_taps uint32
+ Make various resizer fields uint32 so that we can use them in ORC
+ functions later.
+
+2014-10-27 11:59:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: don't convert too much
+ Always convert the smallest width.
+
+2014-10-27 10:13:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/resampler.c:
+ * gst-libs/gst/video/video-scaler.c:
+ * tests/check/libs/video.c:
+ resampler: make shift easier to use
+
+2014-10-26 05:58:56 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/resampler.c:
+ * gst-libs/gst/video/resampler.h:
+ * gst-libs/gst/video/video-converter.c:
+ resampler: add parameters to cubic filter
+ Improve cubic filter and add parameters. Switch to mitchell filter
+ by default.
+
+2014-10-24 16:51:37 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/video-scaler.h:
+ * tests/check/libs/video.c:
+ video-scaler: add extra options
+
+2014-10-24 16:42:11 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ video-converter: define some options
+
+2014-10-24 16:23:53 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/resampler.c:
+ * gst-libs/gst/video/resampler.h:
+ resampler: add some options
+
+2014-10-24 15:42:31 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/resampler.c:
+ resampler: limit max number of taps
+ Don't use more taps than the input size.
+
+2014-10-24 15:28:22 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: add scaling support
+ Add scaling support for the video-converter object
+
+2014-10-24 15:25:33 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video-scaler.c:
+ * gst-libs/gst/video/video-scaler.h:
+ * gst-libs/gst/video/video.h:
+ * tests/check/libs/video.c:
+ video-scaler: add video scaler helper object
+ Add a video scaler object build on top of the resampler. It has
+ implementation to deal with interlaced video as well as horizontal and
+ vertical scaling functions.
+
+2014-10-24 13:01:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/resampler.c:
+ * gst-libs/gst/video/resampler.h:
+ video: add generic resampler
+ Add an object that can generate a set of resample filter coefficients.
+
+2014-10-24 12:11:43 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: rework the generic converter function
+ Use a LineCache object to track and process lines between unpack,
+ upsample, convert, downsample and pack stages. This simplifies the
+ main core processing function a lot and allows for future additions
+ easily.
+ Add support for interlaced formats in chroma up and downsampling.
+
+2014-10-24 11:45:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ * gst/videoconvert/gstvideoconvert.c:
+ video-convert: swap src and dest
+ It is more natural and consistent with other uses.
+
+2014-10-24 11:35:31 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ video-chroma: fix typo
+
+2014-10-27 17:56:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 84d06cd to 7bb2bce
+
+2014-10-23 14:41:13 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst-libs/gst/video/video-blend.c:
+ video-blend: segfault when xpos >= video size
+ When the xpos is given as greater than or equal to the video size,
+ we get a segfault, due to improper condition.
+ Hence adding proper conditions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738984
+
+2014-10-23 14:38:07 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: segfault when xpos >= video size
+ When the xpos is given as greater than or equal to the video size,
+ we get a segfault, due to improper condition.
+ Hence adding proper conditions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738984
+
+2014-10-26 21:31:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/examples/app/.gitignore:
+ examples: add new appsink example to .gitignore
+
+2014-10-26 11:04:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ Revert "decodebin: fix the autoplugging of parser elements"
+ This reverts commit 2b0d3927410ae24e6b0fce100bd4ebbbe805a66f.
+ This breaks cases where an actual second parser is required after the parser,
+ e.g. to do timestamp corrections.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=738416
+
+2014-10-26 11:04:38 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ Revert "decodebin: Fix locking"
+ This reverts commit aa94d5dc9aa6ef381da6b60a67f218117c662958.
+
+2014-10-24 13:09:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/playbin-complex.c:
+ tests: fix playbin-complex test on big endian
+
+2014-10-24 13:04:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/struct_ppc32.h:
+ tests: fix expected GstRTSPTimeRange structure size for ABI test for ppc32
+ Also see https://bugzilla.gnome.org/show_bug.cgi?id=695276
+
+2014-10-24 12:26:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/adder.c:
+ tests: fix adder check on big-endian
+
+2014-10-24 10:17:47 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * android/rtsp.mk:
+ * gst-libs/gst/rtsp/.gitignore:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtsp-marshal.list:
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ rtsp: use generic marshaller
+
+2014-10-23 11:22:35 +0200 Thibault Saunier <tsaunier@gnome.org>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Make GstBaseTextOverlay::font-desc readable
+
+2014-10-21 13:01:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From a8c8939 to 84d06cd
+
+2014-10-21 13:30:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Fix locking
+ The chain mutex needs to be locked when looking at chain->elements. Move code
+ around a bit to require only one lock() and unlock().
+
+2014-10-21 12:58:41 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: fix the autoplugging of parser elements
+ If there are two parser elements available for the same media format,
+ then decodebin is autoplugging an extra capsfilter and parser irrespective
+ of caps and rank. So restrict the decodebin from autoplugging multiple parser
+ elements back to back in adjacent positions with in a single DecodeChain
+ for the same media format.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738416
+
+2014-10-21 12:57:59 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 6e75498 to a8c8939
+
+2014-10-21 14:43:30 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ videotestsrc: assertion error
+ timestamp_offset is being declared as an int64 variable,
+ for which the min
+ value of G_MININT64 is -9223372036854775808
+ Changing the minimum and maximum limit for the offset variable.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738568
+
+2014-10-13 00:03:55 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: optimize the code a bit by avoiding unnecessary string comparisons
+ https://bugzilla.gnome.org/show_bug.cgi?id=738416
+
+2014-10-13 00:03:20 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Fix typo in comment
+ https://bugzilla.gnome.org/show_bug.cgi?id=738416
+
+2014-10-01 15:04:09 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: call watch notify before freeing any watch resources
+ This gives control to the notify function allowing it to finish other
+ watch related functionality.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737752
+
+2014-10-20 15:31:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: Fix gst_app_sink_pull() docs to transfer full for the return value
+ Also we get a GstSample, not a GstBuffer here.
+
+2014-10-17 12:10:44 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: use gslice for typefine data
+ Also use our free function in the failure case.
+
+2014-10-13 15:58:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: fix some leaks in error code path
+ Fixes test_encodebin_sink_pads_nopreset_static
+ running under valgrind.
+
+2014-10-13 05:08:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * common:
+ tests: parallelise 'make valgrind'
+ Use $(MAKE) instead of 'make' inside the Makefile,
+ otherwise the make will run as if -j1 had been
+ specified and complain about the job server not
+ being available, and with $(MAKE) in inherits the
+ parent make's settings it seems.
+ Upgrade common submodule for parallel check-valgrind.
+
+2014-10-03 12:57:52 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: allow more channel_masks
+ Allow partial valid channel masks.
+ Set channel mask to 0 for non-valid channel masks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733405
+
+2014-10-03 12:54:17 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst-libs/gst/audio/audio-channels.c:
+ audio-channels: allow partially valid channel_mask
+ Since WAVEFORMATEXTENSIBLE allows to have more channels than
+ bits in the channel mask we should allow this, too, to avoid
+ loss of information.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733405
+
+2014-10-13 22:24:31 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: should post DECODE errors and not ENCODE
+ Fix error code for audio decoder
+
+2014-10-10 18:49:29 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst-libs/gst/video/video-blend.c:
+ videoblend: Avoid assigning a negative value to a guint
+ There are some few but certain conditions where it is possible for the
+ dest_width to be smaller than x. So we check this before assigning a negative
+ value to src_width, which is a unsigned and would be promoted to a number that
+ can segfault videoblend.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738242
+
+2014-10-10 10:05:19 +0530 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Fix segfault when overlay outside the frame
+ When the textoverlay is set outside the video frame by deltax or deltay the
+ calculation segfaults, but it is also unnecessary since it doesn't need to be
+ displayed. So we should clip the text.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738242
+
+2014-10-10 17:32:41 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ pbutils: Rename clock-base/seqnum-base to timestamp-offset/seqnum-offset
+ To match how they were renamed elsewhere.
+
+2014-10-10 12:14:17 +0300 Heinrich Fink <hfink@toolsonair.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Use correct property enum value for video-filter property installation
+
+2014-10-08 16:50:52 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: remove FIXME about NV21 support
+ NV21 is already supported so removing FIXME about adding support for it.
+
+2014-10-08 11:26:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: add gradient pattern
+ Makes a gradient between background and foreground color.
+
+2014-10-06 15:17:42 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ video-chroma: improve 4x downsampling coefficients
+
+2014-10-06 22:13:00 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst/audioresample/gstaudioresample.h:
+ audioresample: remove unused variables
+ https://bugzilla.gnome.org/show_bug.cgi?id=738026
+
+2014-10-07 05:50:56 +0900 Danny Song <danny.song.ga@gmail.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Remove leftover #define from 0.10
+ https://bugzilla.gnome.org/show_bug.cgi?id=738018
+
+2014-10-07 12:10:42 +0400 Andrei Sarakeev <sarakusha@gmail.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Only emit the drain signal for the main decode chain, not any subchains
+ https://bugzilla.gnome.org/show_bug.cgi?id=738064
+
+2014-10-06 10:15:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Free factories array when delaying autoplugging due to non-final caps
+
+2014-10-06 10:11:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ videoconverter: Free the converter config in free()
+
+2014-10-02 21:20:48 +0200 Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: unref decode pad after usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=737757
+
+2014-10-04 23:09:19 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Stop storing if we received EOS
+ This was never reset when going from PAUSED->READY and resulted
+ in encoders being not reusable after EOS. They just rejected any
+ buffer because they received EOS in their previous life.
+ The flag wasn't used anywhere except for rejecting buffers after
+ EOS, and this is now handled by GstPad directly.
+
+2014-10-02 00:14:03 +0200 Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+ * ext/vorbis/gstvorbisdeclib.c:
+ vorbisdec: don't reorder streams with channels count greater than eight
+ vorbis_reorder_map is defined for eight channels max. If we have more
+ than eight channels, it's the application which shall define the order.
+ Since we set audio position to none, we just interleave all the channels
+ without any particular reordering.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737742
+
+2014-03-04 16:51:11 +0200 Andres Gomez <agomez@igalia.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Removed setting "iradio-mode" property in the source element
+ The "iradio-mode" property used to have a default FALSE value in HTTP
+ source elements but now it should default to TRUE or just do not exist
+ as a property so it is not really needed to set it any more in
+ uridecodebin.
+ Apart from that this code could've never worked as uridecodebin looks for a
+ string-typed iradio-mode property, but it's a boolean in all sources.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725383
+
+2014-10-02 02:46:58 +1000 Jan Schmidt <jan@centricular.com>
+
+ * docs/design/part-stereo-multiview-video.markdown:
+ design: Add a proposal for handling stereoscopic 3D and multiview
+
+2014-10-01 11:16:30 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: release frame in finish_frame when no output state is configured
+ Otherwise, frame is leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737706
+
+2014-09-25 17:32:32 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-libs/gst/video/video-orc.orc:
+ video-converter: add orc optimized matrix8 function
+ Add an ORC implementation of the matrix8 function.
+ Regenerate video-orc-dist.[ch]
+
+2014-09-29 19:45:22 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audio: Fix up a comment in GstAudioBaseSink
+ Rewrote the comment to not be PulseAudio-specific.
+
+2014-09-27 20:05:38 +0200 Rico Tzschichholz <ricotz@ubuntu.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ video: Make sure to link against libm
+
+2014-09-27 15:58:51 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/xvimage/xvimagepool.c:
+ * sys/xvimage/xvimagepool.h:
+ xvimagesink: get rid of unnecessary private struct for pool
+
+2014-09-27 15:53:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/ximage/ximagepool.c:
+ * sys/ximage/ximagepool.h:
+ ximagesink: get rid of unnecessary private struct for pool
+ This is not exposed as API after all.
+
+2014-09-24 20:38:31 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst-libs/gst/audio/gstaudioiec61937.c:
+ audio: Trivial comment for unhandled MPEG-2 payloading case
+ The spec mentions a version of the MPEG-2 frame with a base frame and
+ extension frame. I don't have IEC 13818-3 to figure out what that is,
+ and don't see any references in search results, so it's a FIXME for now.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736797
+
+2014-09-24 20:11:49 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst-libs/gst/audio/gstaudioiec61937.c:
+ audio: Fixes for MPEG-2 LSF IEC61937 payloading
+ The low sample frequency case for MPEG-2 is <=12kHz (the 32kHz number
+ applies to MPEG-1).
+ https://bugzilla.gnome.org/show_bug.cgi?id=736797
+
+2014-09-17 17:40:04 +0530 Anuj Jaiswal <anuj.jaiswal@samsung.com>
+
+ * gst-libs/gst/audio/gstaudioiec61937.c:
+ audio: correct condition for MPEG case.
+ Signed-off-by: Anuj Jaiswal <anuj.jaiswal@samsung.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=736797
+
+2014-09-26 18:14:11 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video: improve YUV -> RGB conversion
+ Reorganize orc instructions to free up some registers.
+ We can reuse the ORC code to implement the generic AYUV->ARGB matrix.
+
+2014-09-26 16:35:51 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videotestsrc/gstvideotestsrcorc.orc:
+ videotestsrc: storel is better then copyl
+ It is better to use storel to splat the variable into the destination.
+ ORC doesn't know when a variable is last written to so it can't yet optimize
+ away the copy operation.
+
+2014-09-26 15:00:12 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/videoscale/vs_lanczos.c:
+ videoscale: avoid recalculating values
+ Avoid recalculating values used multiple times as base of index. Plus some style
+ fixes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737400
+
+2014-09-26 09:14:51 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_lanczos.c:
+ videoscale: support lanczos method for NV formats
+ Support lanczos scaling method for NV12 and NV21 formats.
+ Scale the 'Y' plane and scale 'NV' plane.
+ Implementation for submethods - int16, int32, float and double
+ https://bugzilla.gnome.org/show_bug.cgi?id=737400
+
+2014-09-25 15:19:21 +0100 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 files
+
+2014-09-24 16:19:30 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video-converter.c:
+ * gst-libs/gst/video/video-converter.h:
+ * gst-libs/gst/video/video.h:
+ * gst/videoconvert/gstvideoconvert.c:
+ * gst/videoconvert/gstvideoconvert.h:
+ * win32/common/libgstvideo.def:
+ video: convertor -> converter
+
+2014-09-24 15:49:42 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video-convertor.c:
+ * gst-libs/gst/video/video-convertor.h:
+ * gst-libs/gst/video/video-orc.orc:
+ * gst-libs/gst/video/video.h:
+ * gst/videoconvert/Makefile.am:
+ * gst/videoconvert/gstcms.c:
+ * gst/videoconvert/gstcms.h:
+ * gst/videoconvert/gstvideoconvert.c:
+ * gst/videoconvert/gstvideoconvert.h:
+ * gst/videoconvert/gstvideoconvertorc-dist.c:
+ * gst/videoconvert/gstvideoconvertorc-dist.h:
+ * gst/videoconvert/gstvideoconvertorc.orc:
+ * gst/videoconvert/videoconvert.h:
+ * tests/check/Makefile.am:
+ * win32/common/libgstvideo.def:
+ video: move videoconvert code to video library
+ Move the conversion code used in videoconvert to the video library
+ and expose a simple but generic API to do arbitrary conversion. It can
+ currently do colorspace conversion but the plan is to add videoscale to
+ it as well.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=732415
+
+2014-09-24 11:04:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video-color.c:
+ * gst-libs/gst/video/video-color.h:
+ * gst/videoconvert/videoconvert.c:
+ * win32/common/libgstvideo.def:
+ video-color: add gst_video_color_matrix_get_Kr_Kb()
+ Move the function to get the color matrix coefficients from
+ videoconvert to the video library.
+
+2014-09-23 14:14:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiosink.c:
+ audiosink: compensate for segment restart with clock's time_offset
+ When playing chained data the audio ringbuffer is released and
+ then acquired again. This makes it reset the segbase/segdone
+ variables, but the next sample will be scheduled to play in
+ the next position (right after the sample from the previous media)
+ and, as the segdone is at 0, the audiosink will wait the duration
+ of this previous media before it can write and play the new data.
+ What happens is this:
+ pointer at 0, write to 698-1564, diff 698, segtotal 20, segsize 1764, base 0
+ it will have to wait the length of 698 samples before being able to write.
+ In a regular sample playback it looks like:
+ pointer at 677, write to 696-1052, diff 19, segtotal 20, segsize 1764, base 0
+ In this case it will write to the next available position and it
+ doesn't need to wait or fill with silence.
+ This solution is borrowed from pulsesink that resets the clock to
+ start again from 0, which makes it reset the time_offset to the time
+ of the last played sample. This is used to correct the place of
+ writing in the ringbuffer to the new start (0 again)
+ https://bugzilla.gnome.org/show_bug.cgi?id=737055
+
+2014-09-21 13:16:43 +0200 Ognyan Tonchev <otonchev@gmail.com>
+
+ * gst-libs/gst/video/gstvideopool.c:
+ videopool: add missing annotation for gst_video_buffer_pool_new()
+ https://bugzilla.gnome.org/show_bug.cgi?id=737072
+
+2014-09-23 23:12:19 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videoscale/vs_4tap.c:
+ videoscale Use stride instead of width in more places
+
+2014-09-19 12:31:49 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Use width instead of stride in buffer offset calculation
+ https://bugzilla.gnome.org/show_bug.cgi?id=736944
+
+2014-09-23 11:56:33 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: reshuffle code in error handling
+ Move the assert to the error handling block at the end of the function so the
+ the logging is still triggered. Reword the logging slightly and add another
+ comment to hint what went wrong.
+ Fixes #737138
+
+2014-09-22 20:15:13 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: log the timestamps if we are unhappy about them
+ When complaining about the DTS!=PTS on keyframes log the actualy timestamps.
+
+2014-09-22 10:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/Makefile.am:
+ tests: add orc test for videoconvert
+
+2014-09-22 10:40:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: Fix format string compiler warning
+ gst-play.c:92:28: error: format string is not a string literal
+ [-Werror,-Wformat-nonliteral]
+ len = g_vasprintf (&str, format, args);
+ ^~~~~~
+
+2014-09-19 14:58:20 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/examples/overlay/gtk-videooverlay.c:
+ example/overlay: Specify minimum gdk version
+ Avoids deprecation warnings (such as for gtk_widget_set_double_buffered()
+ which became deprecated from 3.14)
+
+2014-09-19 18:29:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: add --quiet option to suppress output
+
+2014-09-05 13:49:46 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Do not fail the negotiation if query fails
+ The allocation query failure doesn't mean that the negotiation
+ has failed as the element can allocate buffers itself.
+ Instead, only fail if the pads are flushing and the allocation
+ query failed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735844
+
+2014-09-18 15:45:43 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Added NV support for 4Tap resize
+ https://bugzilla.gnome.org/show_bug.cgi?id=736845
+
+2014-09-18 12:29:37 +0400 Andrei Sarakeev <sarakusha@gmail.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Don't leak input-selector sinkpads
+ https://bugzilla.gnome.org/show_bug.cgi?id=736861
+
+2014-09-18 12:39:48 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Simplify code a bit
+
+2014-09-17 14:34:25 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/encoding/gststreamsplitter.c:
+ streamsplitter: do not leak events when flushing them
+ https://bugzilla.gnome.org/show_bug.cgi?id=736796
+
+2014-09-17 14:18:49 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: do not leak events when flushing them
+ https://bugzilla.gnome.org/show_bug.cgi?id=736796
+
+2014-09-17 14:11:21 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: do not leak events when flushing them
+ https://bugzilla.gnome.org/show_bug.cgi?id=736796
+
+2014-09-17 14:08:17 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: do not leak events when flushing them
+ https://bugzilla.gnome.org/show_bug.cgi?id=736796
+
+2014-09-17 12:17:27 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/libs/audiodecoder.c:
+ audiodecoder: extend flush_events test to check for event leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=736788
+
+2014-09-17 12:17:53 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Don't leak events
+ https://bugzilla.gnome.org/show_bug.cgi?id=736788
+
+2014-09-16 13:32:52 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/audio/gstaudiocdsrc.c:
+ audiocdsrc: do not leak uid after parsing TOC select event
+ https://bugzilla.gnome.org/show_bug.cgi?id=736739
+
+2014-09-17 10:51:59 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: correct the condition for irap flag
+ https://bugzilla.gnome.org/show_bug.cgi?id=736779
+
+2014-09-16 21:42:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Add audio/videoconvert in front of the audio/video-filters
+ audioresample and videoscale is something the application will have to do if
+ required, but we can at least help here by adding the
+ audioconvert/videoconvert elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735748
+
+2014-09-16 01:07:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: Don't ref buffers twice when mapping
+
+2014-09-16 00:41:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/app/gstappsink.h:
+ * gst-libs/gst/app/gstappsrc.h:
+ app: Add FIXME comment for making the instance/class structs private
+
+2014-09-15 21:51:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/app/gstappsrc.h:
+ appsrc: fix recent ABI breakage caused by GstAppSrc structure size increase
+ Also fixes 'make check'.
+ https://bugzilla.gnome.org/show_bug.cgi?id=728379
+
+2014-09-15 16:23:57 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: do not leak pool and allocator in error case
+ https://bugzilla.gnome.org/show_bug.cgi?id=736679
+
+2014-09-12 14:41:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideofilter.c:
+ videofilter: Use new GST_VIDEO_FRAME_MAP_FLAG_NO_REF
+ https://bugzilla.gnome.org/show_bug.cgi?id=736118
+
+2014-09-12 14:39:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ * gst-libs/gst/video/video-frame.h:
+ video-frame: Add GST_VIDEO_FRAME_MAP_FLAG_NO_REF
+ This makes sure that the buffer is not reffed another time when
+ storing it in the GstVideoFrame, keeping it writable if it was
+ writable.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736118
+
+2014-09-12 14:27:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideofilter.c:
+ videofilter: Unref buffers before calling the transform_frame functions
+ GstVideoFrame has another reference, so the buffer looks unwriteable,
+ meaning that we can't attach any metas or anything to it
+ https://bugzilla.gnome.org/show_bug.cgi?id=736118
+
+2014-09-05 09:54:10 -0700 Garg <aksg86@gmail.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: Fix deadlock caused by holding object lock while calling clock functions
+ Issue:
+ During a PAUSED->PLAYING transition when we are rendering an audio buffer in AudioBaseSink
+ we make adjustments to the sink's provided clock i.e. fix clock calibration using the external
+ pipeline clock, within "gst_audio_base_sink_sync_latency function inside gstaudiobasesink.c".
+ For the calibration adjustment we need to get the sink clock time using "gst_audio_clock_get_time".
+ But before calling "gst_audio_clock_get_time" we acquire the Object Lock on the Sink. If sink is
+ a pulsesink, "gst_audio_clock_get_time" internally calls "gst_pulsesink_get_time" which needs to
+ acquire Pulse Audio Main Loop Lock before querying Pulse Audio for its stream time using
+ "pa_stream_get_time". Please see "gst_pulsesink_get_time in pulsesink.c".
+ So the situation here is we have acquired the Object lock on Sink and need PA Main Loop Lock.
+ Now Pulse Audio Main Thread itself might be in the process of posting a stream status
+ message after Paused to Playing transition which in turn acquires the PA Main loop lock and
+ needs the Object Lock on Pulse Sink. This causes a deadlock with the earlier render thread.
+ Fix:
+ Do not acquire the object Lock on Sink before querying the time on PulseSink clock. This is
+ similar to the way we have used get_time at other places in the code. Acquire it after the
+ get_time call. This way PA Main loop will be able to post its stream status message by
+ acquiring the Sink Object lock and will eventually release its Main Loop lock needed for
+ gst_pulsesink_get_time to continue.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736071
+
+2014-09-04 11:56:50 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * tests/examples/app/Makefile.am:
+ * tests/examples/app/appsink-src2.c:
+ appsrc: Add example that shows gst_app_src_push_sample() usage
+
+2014-09-05 11:14:51 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ * win32/common/libgstapp.def:
+ appsrc: Add push_sample() convenience function for easy appsink -> appsrc use
+ https://bugzilla.gnome.org/show_bug.cgi?id=728379
+
+2014-09-11 22:19:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/xvimage/xvcontext.c:
+ * sys/xvimage/xvcontext.h:
+ xvimagesink: only try to set XV_ITURBT_709 port attribute if it exists
+ Don't try to set port attribute that's not advertised by the
+ adaptor. Fixes videotestsrc ! xvimagesink aborting with
+ X Error of failed request: BadMatch (invalid parameter attributes)
+ Major opcode of failed request: 151 (XVideo)
+ Minor opcode of failed request: 13 ()
+ on intel HD4600 graphics with kernel 3.16, xserver 1.15,
+ intel driver 2.21.15.
+
+2014-09-11 16:58:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: protect buffering message handling
+ Use the object lock to avoid concurrent processing which leads
+ to small disasters (assertions or crashes)
+
+2014-09-10 17:24:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ Fix up one-element lists in template caps
+
+2014-09-09 11:37:26 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: ignore timeout in session request header
+ The timeout parameter is only allowed in a session response header
+ but some clients, like Honeywell VMS applications, send it as part
+ of the session request header. Ignore everything from the semicolon
+ to the end of the line when parsing session id.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736267
+
+2014-03-28 13:02:54 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: filter out buffering messages when switching uri
+ When switching URI from about-to-finish, playbin starts decoding the new
+ URI and the queue2 inside uridecodebin starts emitting buffering messages
+ immediately. However, the queue(s) inside playsink still have buffers to
+ play and the pipeline doesn't need to pause for buffering, so we should
+ not send those buffering messages up to the application, otherwise there
+ is an audible glitch caused by pausing the pipeline for a very short time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=727255
+
+2014-07-08 12:37:41 -0400 Kipp Cannon <kipp.cannon@ligo.org>
+
+ * gst/audioresample/resample.c:
+ audioresample: don't skip input samples
+ when downsampling, the output buffer can be filled before all the input
+ samples are consumed. this is correct: when downsampling, several input
+ samples are needed for each output sample, so when only a small number of
+ input samples are available the number of output samples produced can be 0.
+ the resampler, however, was discarding those extra input samples instead of
+ clocking them into its filter history for the next iteration. this patch
+ fixes this by removing the check that the output buffer is full. the code
+ now always loops until all input samples are consumed, and relies on the
+ calling code to have provided a suitably sized location for the output.
+ note that there are already other checks in place in the calling code to
+ ensure that this is the case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732908
+
+2013-01-31 13:49:00 +0100 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: get framerate from previously parsed video info
+
+2013-01-31 13:47:35 +0100 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: do not ask for a bufferpool when checking for composition meta
+
+2014-09-04 15:06:31 +0200 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: schedule reconfigure on source pad when negotiation fails
+ The source pad might be flushing while negotiating, resulting in
+ set_caps or the ALLOCATION query failing. In this case set the
+ reconfigure flag on the source pad so that negotiation is retried on the
+ next buffer.
+
+2013-01-31 15:38:18 +0100 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: just forward the seek event to sink pads like other events
+ https://bugzilla.gnome.org/show_bug.cgi?id=735844
+
+2014-09-04 12:13:45 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: remove unneeded cairo transparence setting
+ he code here:
+ http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/pango/gstbasetextoverlay.c#n1554
+ should make transparent the box that contains the text, I think this code is
+ not correct, it should be:
+ if (overlay->want_shading) {
+ double alpha = overlay->shading_value / 255.0;
+ cairo_paint_with_alpha (cr, alpha);
+ }
+ however I think this code could be removed, we already do a shaded background,
+ why shade the box behind the text with cairo too? only one shading is needed so
+ we must shade with cairo or with methods like these:
+ http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/pango/gstbasetextoverlay.c#n1642
+ not both
+ https://bugzilla.gnome.org/show_bug.cgi?id=736028
+
+2014-09-02 13:10:34 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: Make shading_value a property
+ https://bugzilla.gnome.org/show_bug.cgi?id=735879
+
+2014-09-03 15:23:26 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: GstStructure refcount critical message
+ s3 is not being initialized when run in a loop
+ and the same was being freed, which resulted in the crash
+ https://bugzilla.gnome.org/show_bug.cgi?id=735952
+
+2014-09-02 15:37:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Also include the raw caps in the error message, not just the human readable description
+
+2014-09-02 12:59:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Include codec description for missing plugins in the error message
+ If we had plugins and an error occurred we only include the error message
+ caused by this, otherwise we will include the codec description as generated
+ from the caps.
+ This allows to detect which exact codec was missing instead of getting a
+ generic "no suitable decoders found" error message.
+
+2014-09-01 15:23:27 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/textoverlay.c:
+ tests: textoverlay: add test to reproduce fakesink scenario
+ Adds a new test to textoverlay to make sure it can properly handle
+ elements that have ANY caps but fail to add the overlay meta in
+ the allocation query.
+ This test verifies that textoverlay won't use the caps features even
+ knowing that the overlay meta is accepted when querying the downstream
+ caps because it also needs downstream to confirm by putting the meta
+ in the allocation query.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735800
+
+2014-09-01 12:38:02 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: properly fallback to non-overlay caps
+ When downstream claims to accept the overlay meta but fails to
+ provide it in the allocation query, properly fallback to setting
+ a new caps without the overlay meta as that is not going to be used.
+ Only do this if the original caps doesn't have the overlay already,
+ otherwise there isn't much that can be done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735800
+
+2014-09-01 15:06:51 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: don't set segment.base in pad_submit_packet()
+ Setting segment.base in the segment sent from gst_ogg_demux_handle_page() is
+ enough to ensure that chained oggs are played corretly (see bgo#706569).
+ Tweaking the base in gst_ogg_pad_submit_packet() as well result in delays when
+ playing a file with start != -1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735808
+
+2014-09-01 12:28:24 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ textoverlay: Don't hold any mutexes while calling negotiate
+ It's not done in any other code calling negotiate and will cause deadlocks
+ as it is sending events and queries in the pipeline.
+ Specifically this pipeline was deadlocking:
+ gst-launch-1.0 videotestsrc ! textoverlay ! textoverlay ! fakesink
+
+2014-08-29 14:00:06 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: accumulate base time
+ Base time should be accumulated so non flushing seeks have the expected base.
+ Not accumulating result in segments appearing as "too late" and so are not
+ played by the sink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735509
+
+2014-08-29 19:15:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ textoverlay: remove code that can't be reached
+ If this code could ever be reached, it would leak
+ memory (CID 1231978), but gst_caps_get_features()
+ never returns NULL, so that can't happen.
+
+2014-08-29 18:18:10 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/encoding/gstencodebin.c:
+ encoding: remove assignment that's no longer needed
+ CID 1231980
+
+2014-07-23 21:25:24 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-read.c:
+ riff: Recognize RF64 as RIFF file
+ https://bugzilla.gnome.org/show_bug.cgi?id=735631
+
+2014-08-27 13:45:57 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Protect readsrc, writesrc and controllsrc with a mutex
+ Fixes a crash when controlsrc, readsrc or writesrc are modified from
+ gst_rtsp_source_dispatch_read/write and gst_rtsp_watch_reset at the
+ same time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735569
+
+2014-08-28 17:13:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: setcaps() always returns TRUE and the return value is unused
+ Change it to a void return value. The caps are forwarded afterwards via
+ gst_pad_event_default() and not inside this function.
+ CID 1226477
+
+2014-08-28 17:06:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Fix broken boolean expression
+ We can seek with end_type==NONE and end_type==SET && end_position=-1. The
+ check for end_type!=NONE made the second condition impossible.
+ CID 1226440
+
+2014-08-28 17:00:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Fix broken boolean expression
+ We can seek with end_type==NONE and end_type==SET && end_position=-1. The
+ check for end_type!=NONE made the second condition impossible.
+ CID 1226439
+
+2014-08-25 20:59:40 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ decodebin: Include information from the error messages of tried but failed elements in the missing plugin errors
+
+2014-08-25 16:22:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Initialize local variables for every retry
+
+2014-08-25 15:15:06 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Remove error case that resulted in two error messages
+ We already send one in gst_decode_bin_expose() for this case. Only
+ if we're unable to typefind the caps another error message is needed.
+
+2014-08-24 22:36:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: tighten checks for 'freeform mp3' a little
+ Freeform mp3s typically have bitrates higher than the
+ otherwise max allowed rate. Prevents misdetection of
+ some truetype font files as mp3.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732923
+
+2014-08-25 13:14:36 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Don't ignore ::start/stop return values
+
+2014-08-18 13:04:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-plugins-base.spec.in:
+ spec: add gst-device-monitor-1.0 to RPM .spec file
+ https://bugzilla.gnome.org/show_bug.cgi?id=734944
+
+2014-08-14 16:57:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: only intersect with the filter at the end
+ Otherwise we might change some capsfeatures from ANY to the specific
+ value from the filter and do not filter those out in case the
+ sink doesn't support them
+ https://bugzilla.gnome.org/show_bug.cgi?id=734822
+
+2014-08-15 13:31:53 +0200 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Set 'processing = FALSE' when done discovering SYNC
+ This avoids a race where we would get new tag but we are already
+ prerolled and analyzing results.
+ It is the way it is supposed to be handled as stated in comment:
+ "If preroll is complete, drop these tags - the collected information is
+ possibly already being processed and adding more tags would be racy"
+
+2014-08-14 17:21:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * win32/common/libgstvideo.def:
+ gstvideo: add missing entry to win32 .def
+ gst_video_guess_framerate
+
+2014-08-14 23:53:16 +1000 Jan Schmidt <jan@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add gst_video_guess_framerate() function
+ Takes a nominal frame duration and returns a standard
+ FPS if it matches closely enough (< 0.1%), or else
+ calculates a framerate that'll do.
+
+2014-08-15 01:04:45 +1000 Jan Schmidt <jan@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideometa.h:
+ * gst-libs/gst/video/gstvideoutils.h:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-frame.h:
+ * gst-libs/gst/video/video-overlay-composition.c:
+ video: Various simple docs fixes
+
+2014-08-08 20:01:20 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ videodecoder: Reset last_timestamp_out on new segment
+ Reset last_timestamp_out when applying the output segment
+ change, to avoid decoder confusion over new timestamp timelines when
+ a seamless segment change happens.
+ Move some locks/unlocks to later when they're actually needed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734617
+
+2014-07-14 12:29:50 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: handle group switching for deadend group
+ Gracefully handle switching groups that all pads are deadend.
+ This can happen when quickly switching programs on mpegts as the
+ output is unaligned it can happen that not enough data was accumulated at
+ parsers to generate any buffers, causing the stream to receive EOS before
+ any data can be decoded.
+ To handle this scenario, the _expose function now also gets if there is
+ any next group to be exposed along with the list of endpads. If there are
+ no endpads and there is another group to expose it will switch to this next
+ group and then retry exposing the streams.
+ Also, the requirement to only switch from the chain that has the endpad had
+ to be modified to care for when the drainpad is NULL
+ https://bugzilla.gnome.org/show_bug.cgi?id=733169
+
+2014-07-11 18:51:44 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: consider all deadend pads as drained
+ Otherwise when switching out a group with a deadend pad it will block
+ as it would be waiting for EOS on a deadend that already got one
+ https://bugzilla.gnome.org/show_bug.cgi?id=733169
+
+2014-08-12 13:41:04 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: fix caps negotiation filter
+
+2014-08-13 14:28:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: Make sure to intersect raw caps with our converter caps
+ Otherwise we end up allowing video/x-raw with arbitrary caps features that are
+ not handled by our converters.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734683
+
+2014-08-12 23:18:57 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Don't drain and flush on SEGMENT events.
+ As was done for the base video decoder in commit 695675, don't
+ flush out the decoder on a new SEGMENT event. Segment events
+ may be a new segment, but are also often segment updates for
+ the current segment where the old data should be kept. For new
+ segments, a STREAM_START event will already trigger a drain, but
+ make sure to flush any remaining partial data then as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734666
+
+2014-08-11 10:15:14 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add NV21 support
+ https://bugzilla.gnome.org/show_bug.cgi?id=734650
+
+2014-08-11 18:21:26 +0200 Matthieu Crapet <mcrapet@gmail.com>
+
+ * tests/icles/playback/decodetest.c:
+ * tests/icles/playback/test.c:
+ * tests/icles/playback/test5.c:
+ tests: fix decodebin signal used in icles/playback/ decodetest, test and test5
+ Since release 1.1.4, "new-decoded-pad" no longer exists.
+
+2014-08-08 12:46:47 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ * tests/check/elements/textoverlay.c:
+ basetextoverlay: rework caps negotiation
+ Make textoverlay negotiate caps more correctly.
+ 1) Check what caps we received in the video-sink
+ 2) If it already has the overlay meta -> use it directly
+ 3) If it doesn't, textoverlay try adding the overlay meta and using it,
+ if downstream doesn't support it, just use what is received in the
+ video-sink
+ 4) Check if the allocation query also supports the meta to enable
+ really using it
+ Before it wasn't really doing renegotiation of any kind, just
+ re-checking if it should use the overlay meta or not
+ Also had to update the caps in the test as memory:SystemMemory seems
+ to be required when you use a caps feature otherwise intersection/subset
+ checks will fail.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733916
+
+2014-08-07 17:35:05 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: always intersect with the filter caps
+ Avoids returning values that upstream can't produce
+ https://bugzilla.gnome.org/show_bug.cgi?id=733916
+
+2014-07-30 16:59:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/encoding/gstencodebin.c:
+ * tests/check/elements/encodebin.c:
+ encodebin: delay missing encoder error as passthrough is still possible
+ Set up a fakesink with a pad probe to replace the missing encoder to detect
+ if encoding was really required and only error out in this case. Otherwise
+ just let passthrough branch work.
+ This delays the error posting from the set_state function to when buffers
+ are really flowing. Unit test updated accordingly
+ https://bugzilla.gnome.org/show_bug.cgi?id=650652
+
+2014-08-08 14:08:19 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: Unref pad template caps after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734517
+
+2014-08-11 10:57:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Remove buffering special casing for adaptive streaming demuxers
+ They output smaller buffers now and we should be able to handle the buffering
+ limits like in every other situation now.
+
+2014-08-07 10:44:03 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Don't set decoding timestamps on raw video
+ https://bugzilla.gnome.org/show_bug.cgi?id=733720
+
+2014-08-07 18:10:41 +0300 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: In reverse playback, flush the output queue after decoding each keyframe chain
+ This fixes the reverse playback scenario when upstream is not fully
+ parsing the stream and does not send every keyframe chain separately
+ with the DISCONT flag on the keyframe.
+ To explain this, let's suppose we have this stream:
+ 0 1 2 3 4 5 6 7 8
+ K K K
+ In most circumstances, the upstream parser will chain in the
+ decoder the buffers in the following order:
+ 6 7 8 3 4 5 0 1 2
+ D D D
+ In this case, GstVideoDecoder will flush the parse queue every time
+ it receives discont (D) and we will eventually get in the output queue:
+ (flush here) 8 7 6 (flush here) 5 4 3 (flush here) 2 1 0
+ In case the upstream parser doesn't do this work, though,
+ GstVideoDecoder will receive the whole stream at once and will flush
+ the parse queue afterwards:
+ 0 1 2 3 4 5 6 7 8
+ D
+ During the flush, it will look backwards for keyframes and will
+ decode in this order:
+ 6 7 8 3 4 5 0 1 2
+ This is the same order that it would receive from upstream if
+ upstream was parsing and looking for the keyframes, only that now
+ there is no flushing of the output queue in between keyframes,
+ which will result in the output queue looking like this:
+ 2 1 0 6 5 3 8 7 6
+ This will confuse downstream obviously and will play incorrectly.
+ This patch forces the decoder to flush the output queue every time
+ it picks a new keyframe to decode, so it will end up decoding 6 7 8
+ and then flushing before picking 3 for decoding, so the output will
+ get 8 7 6 before 6 5 3 and the video will play back correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734441
+
+2014-08-10 17:30:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: use pkg-config to detect x11 and xv libs
+ AC_PATH_XTRA macro unnecessarily pulls in libSM and libICE.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731047
+
+2014-08-10 17:27:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/xvimage/xvimageallocator.c:
+ xvimage: fix crash when outputting debug log
+ Can't print a GstMemory via GST_PTR_FORMAT, it will crash
+ inside GObject checking if it's a GObject, and we can't
+ check generically whether it's a derived GstMemory type,
+ as boxed types don't allowe derivation.
+
+2014-08-09 14:24:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * ext/opus/gstopusheader.c:
+ opus: Improve annotation of internal function
+ https://bugzilla.gnome.org/show_bug.cgi?id=734543
+
+2014-08-09 14:14:48 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: Mark caps argument as not being transferred
+ https://bugzilla.gnome.org/show_bug.cgi?id=734540
+
+2014-08-09 14:20:32 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * ext/vorbis/gstvorbisenc.c:
+ vorbisenc: Improve annotation of internal function
+ https://bugzilla.gnome.org/show_bug.cgi?id=734541
+
+2014-08-06 13:41:46 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/elements/appsrc.c:
+ * tests/examples/app/appsink-src.c:
+ * tests/examples/audio/audiomix.c:
+ * tests/examples/audio/volume.c:
+ * tests/examples/dynamic/codec-select.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/snapshot/snapshot.c:
+ * tests/icles/stress-videooverlay.c:
+ * tests/icles/test-textoverlay.c:
+ tests: Add missing unrefs of objects after use
+ Unreffing the objects returned by gst_bin_get_by_name() and
+ gst_pipeline_get_use() were missing in several tests, so add these.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734359
+
+2014-08-06 13:22:56 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Unref peer pad after use in error case
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734350
+
+2014-08-06 10:07:42 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Some minor fixes and cleanup
+
+2014-08-06 09:59:32 -0400 Wang Xin-yu (王昕宇) <comicfans44@gmail.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Make caps set action queued together with buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=729760
+
+2014-08-01 15:00:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Keep a reference to the playsink sinkpads
+ Otherwise playsink might get shut down without us noticing
+ that our pad references are gone now.
+ Probably fixes https://bugzilla.gnome.org/show_bug.cgi?id=733165
+
+2014-07-30 20:53:53 +0300 Mohammed Sameer <msameer@foolab.org>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: don't unset DISCONT flag
+ Unsetting DISCONT flag means we need to copy the buffer. This copy operation
+ mandates that all GstMemory should be copy-able which is not always the case
+ https://bugzilla.gnome.org/show_bug.cgi?id=727409
+
+2014-07-31 18:40:59 +0200 Edward Hervey <edward@collabora.com>
+
+ * Makefile.am:
+ * common:
+ Makefile: Add usage of build-checks step
+ Allows building checks without running them
+
+2014-07-31 16:09:41 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/libs/rtpbasedepayload.c:
+ * tests/check/libs/rtpbasepayload.c:
+ check: Fix include path of rtp checks
+ Fixes make distcheck
+
+2014-07-30 15:23:39 +0200 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ pbutils: discoverer: Always set the pipeline back to NULL after an error
+ Otherwize the pipeline would be in an wrong state and on the next
+ iteration any kind of error could happen
+ Everytime an error happens in a pipeline the application has to set the
+ pipeline back to NULL instead of READY.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733976
+
+2014-07-29 14:20:42 -0300 Thiago Santos <ts.santos@osg.sisa.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: add missing 'time' word to debug message
+ It prints the buffers, bytes and time limits, but 'time' was missing
+ from the string.
+
+2014-07-28 16:56:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Pass through NO_PREROLL state change returns
+ Fixes playback of live pipelines.
+
+2014-07-28 16:55:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Pass through NO_PREROLL state change returns
+ Fixes playback of live pipelines.
+
+2014-07-26 14:52:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: fix 'attempt to unlock mutex that was not locked' in error code path
+ Fixes playbin unit test with latest GLib.
+
+2014-07-08 16:59:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Don't delay set_format
+ This prevent implementing allocation query, as the format need to be
+ known in order to determin the size and number of buffers needed.
+ Note: This may lead to few regressions that will need fixing
+ https://bugzilla.gnome.org/show_bug.cgi?id=732288
+
+2014-07-23 19:51:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Don't unref caps for which we don't own a reference... get one first
+ https://bugzilla.gnome.org/show_bug.cgi?id=733615
+
+2014-07-23 12:36:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Go asynchronously from READY to PAUSED
+ We now add all our elements to uridecodebin *after*
+ GstBin::change_state(READY->PAUSED), so we need to post async-start
+ and async-done messages ourselves if we want to work async.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733495
+
+2014-07-23 12:27:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Go asynchronously from READY to PAUSED
+ We now add all our elements to uridecodebin *after*
+ GstBin::change_state(READY->PAUSED), so we need to post async-start
+ and async-done messages ourselves if we want to work async.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733495
+
+2014-07-21 15:54:05 +0300 Vivia Nikolaidou <n.vivia@gmail.com>
+
+ * tools/gst-discoverer.c:
+ discoverer: Pretty-print topology tags
+ Call the code used in properties for topology tags too.
+ Side-effect achieved: more tags printed, buffers (e.g. images) shortened.
+
+2014-07-21 13:53:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tools/gst-discoverer.c:
+ discoverer: Fix code style a bit
+ if (...)
+ one_line;
+ else if (...) {
+ many_lines;
+ } else
+ one_line;
+ looks a bit confusing.
+
+2014-07-21 13:48:31 +0300 Vivia Nikolaidou <n.vivia@gmail.com>
+
+ * tools/gst-discoverer.c:
+ discoverer: prettier image tag printing
+ Rather than dumping the serialized sample value, the code now
+ prints the number of bytes in the buffer, then the caps in a
+ human-readable format.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733482
+
+2014-07-10 12:39:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Handle CAPS events immediately instead of delaying them
+ https://bugzilla.gnome.org/show_bug.cgi?id=733147
+
+2014-07-11 21:51:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Handle CAPS events immediately instead of delaying them
+ https://bugzilla.gnome.org/show_bug.cgi?id=733147
+
+2014-07-15 17:34:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/playbin.c:
+ playbin: Fix unit test for last change
+ It will successfully asynchronously go to PAUSED now and
+ later fail.
+
+2014-07-15 17:23:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Create new sources after chaining up to the parent class
+ Otherwise we start the new sources already before the parent class
+ got ready to start.
+
+2014-07-15 17:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Create new sources after chaining up to the parent class
+ Otherwise we start the new sources already before the parent class
+ got ready to start.
+
+2014-07-10 16:26:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/playbin-complex.c:
+ playbin-complex: Change template name from %d to the more common %u
+
+2014-07-10 16:24:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Link Parser/Converter directly and already connect to pad-added and other signals before setting elements to PAUSED
+ otherwise we're going to
+ a) start Parser/Converter before they are linked to their capsfilter,
+ breaking their negotiation of a proper stream format
+ b) start demuxers without having connected to their pad-added signals. We
+ miss pads and in the worst case don't link any pads at all
+
+2014-07-10 12:51:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Send sticky events to the new element after setting it to PAUSED
+ ... and if this fails for whatever reason we skip the element and instead
+ try with the next element. This allows us to handle elements that fail
+ when setting caps on them by just skipping to the next alternative element.
+
+2014-07-10 12:50:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Only link elements further after setting them to PAUSED
+ They might fail to go to PAUSED, and when connecting them further
+ we might already expose their srcpads on decodebin if we're unlucky.
+ This prevents us to handle failures going to PAUSED gracefully.
+
+2014-07-10 12:22:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Remove ERROR message filter after we set the element to PAUSED
+ This allows us to catch more errors gracefully and switch to an alternative
+ element instead.
+
+2014-07-10 12:17:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Only continue autoplugging once the pad has final caps
+ If the caps query returned us fixed caps this doesn't mean yet
+ that these caps are actually complete (fields might be missing).
+ It allows to do us some decisions, but the selection of the next
+ element should be delayed as only complete caps allow proper selection
+ of the next element.
+
+2014-07-10 12:03:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Consider the caps after the capsfilter after parsers for autoplugging
+ Otherwise we might try to continue autoplugging e.g. for a specific
+ stream-format although the parser could convert to something else, thus giving
+ us potentially less options for decoders.
+
+2014-07-21 00:17:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ pbutils: fix missing plugin description for missing elements
+ CID: 1226445
+
+2014-07-19 18:04:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.4.0 ===
+
+2014-07-19 17:04:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 1.4.0
+
+2014-07-19 16:27:43 +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
+
+2014-07-18 21:19:03 -0400 Youness Alaoui <kakaroto@kakaroto.homelinux.net>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Fix memory leak with callback notify not being called in dispose
+ https://bugzilla.gnome.org/show_bug.cgi?id=733386
+
+2014-07-19 12:29:56 +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
+
+2014-07-18 16:01:23 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Add example for using encoder presets with profiles
+ https://bugzilla.gnome.org/show_bug.cgi?id=733349
+
+2014-07-18 15:46:05 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Fix typos and old API in docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=733349
+
+2014-07-17 14:36:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix property description string
+ Spotted by Josep Torra.
+
+2014-07-15 16:56:30 +0200 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES
+ https://bugzilla.gnome.org/show_bug.cgi?id=733207
+
+2014-07-12 10:33:30 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Fix filter property getter
+ The switch-case set was incomplete.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733012
+
+=== release 1.3.91 ===
+
+2014-07-11 11:21:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 1.3.91
+
+2014-07-11 11:21:05 +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
+
+2014-07-11 10:13:03 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * configure.ac:
+ * ext/libvisual/plugin.c:
+ * ext/libvisual/visual.c:
+ libvisual: Remove < 0.4 support
+ And remove the version guards that went along with it
+ https://bugzilla.gnome.org/show_bug.cgi?id=733046
+
+2014-07-10 18:17:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ dmabuf: Ensure _get_fd() works even for shared memory
+ Fixes regression introduced by:
+ commit b60888fd4bcacd42bb4e27fa938272d6e72c5c32
+ Author: Michael Olbrich <m.olbrich@pengutronix.de>
+ Date: Tue May 20 11:18:56 2014 +0200
+ dmabuf: share the mapping with shared copies of the memory
+ https://bugzilla.gnome.org/show_bug.cgi?id=730441
+
+2014-07-10 15:52:46 +0100 Philip Withnall <philip.withnall@collabora.co.uk>
+
+ * ext/opus/gstopusheader.c:
+ opus: Fix a double-unref in the Opus header code
+ The headers were never getting reffed when being added to the headers
+ list, which is later unreffed-and-freed by the caller (e.g.
+ gst_opus_parse_parse_frame()).
+ https://bugzilla.gnome.org/show_bug.cgi?id=733013
+
+2014-07-11 08:51:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/vi.po:
+ po: Update translations
+
+2014-07-03 13:46:08 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst-libs/gst/sdp/sdp.h:
+ sdp: add gstmikey.h to sdp.h
+ https://bugzilla.gnome.org/show_bug.cgi?id=732709
+
+2014-07-03 18:32:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/riff/riff-read.c:
+ riff: Print invalid fourcc in error message in hex
+ Previously this was printed as characters which caused later processing
+ of the error message to sometimes warn about non-UTF-8 characters.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732697
+
+2014-06-20 18:02:31 +0200 Gwenole Beauchesne <gwenole.beauchesne@intel.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: parse any source data that is still available.
+ Fix gst_video_decoder_parse_available() to really parse any pending
+ source data that is still available in the adapter. This is a memory
+ optimization to avoid expansion of video packed added to the adapter,
+ but also a fix to EOS condition when the subclass parse() function
+ ultimately only needed to call into gvd_have_frame() and no additional
+ source bytes were consumed, i.e. gvd_add_to_frame() is not called.
+ This situation can occur when decoding H.264 streams in byte-stream/nal
+ mode for instance. A decoder always requires the next NAL unit to be
+ parsed so that to determine picture boundaries. When a new picture is
+ found, no byte is consumed (i.e. gvd_add_to_frame() is not called)
+ but gvd_have_frame() is called (i.e. priv->current_frame is gone).
+ Also make sure to avoid infinite loops caused by incorrect subclass
+ parse() implementations. This can occur when no byte gets consumed
+ and no appropriate indication (GST_VIDEO_DECODER_FLOW_NEED_DATA) is
+ returned.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731974
+ Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
+
+2014-07-02 15:50:23 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/examples/dynamic/codec-select.c:
+ tests: codec-select: fix compilation
+
+2014-07-02 15:49:38 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/sdp/gstmikey.h:
+ mikey: add more Since markers for new methods
+
+2014-07-02 15:38:41 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/sdp/gstmikey.c:
+ * gst-libs/gst/sdp/gstmikey.h:
+ * tests/check/libs/mikey.c:
+ * win32/common/libgstsdp.def:
+ mikey: make message and payload mini-objects
+ Make the MIKEY message and payload objects miniobjects so that they have
+ a GType and are refcounted.
+ We can reuse the dispose method to clear our payload objects.
+ Add some annotations.
+ Implement a copy function for the MIKEY message.
+ Fix the unit test.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732589
+
+2014-07-02 00:21:00 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/examples/dynamic/codec-select.c:
+ tests: codec-select: Plug element name memory leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=732593
+
+2014-07-01 16:14:43 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/sdp/gstmikey.c:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/video-tile.c:
+ docs: Assorted documentation and introspection fixes for new 1.4 API
+ https://bugzilla.gnome.org/show_bug.cgi?id=732595
+
+2014-07-01 16:19:22 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * tests/check/libs/rtspconnection.c:
+ rtspconnection: also allow POST before GET
+ Don't only allow GET and then POST request to setup tunneling over HTTP
+ but also allow POST and then GET.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732459
+
+2014-06-28 17:08:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/libvisual/gstaudiovisualizer.c:
+ * ext/libvisual/gstaudiovisualizer.h:
+ libvisual: Rename get_type() function to prevent conflicts with static linking
+ https://bugzilla.gnome.org/show_bug.cgi?id=728443
+
+=== release 1.3.90 ===
+
+2014-06-28 11:01:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 1.3.90
+
+2014-06-28 10:56:36 +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
+
+2014-06-27 14:24:10 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: fix caps leak
+ Let go the reference to the converter caps after using it
+
+2014-06-27 10:41:55 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/.gitignore:
+ * tools/Makefile.am:
+ * tools/gst-device-monitor-1.0.1:
+ * tools/gst-device-monitor.c:
+ tools: add gst-device-monitor-1.0 utility
+ Just shows devices with basic info and exits. Or will
+ wait for more devices to show up or be removed with
+ the --follow option. It's also possible to pass filters
+ as command line arguments in the form DEVICE_CLASSES
+ or DEVICE_CLASSES:CAPS.
+
+2014-06-26 16:18:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/streamvolume.h:
+ * gst-libs/gst/tag/xmpwriter.h:
+ libs: There is no G_TYPE_CHECK_INTERFACE_TYPE and G_TYPE_CHECK_INTERFACE_CAST
+ Remove the macros that used them, nobody could've used them anyway.
+
+2014-06-26 11:35:43 +0200 Gwenole Beauchesne <gwenole.beauchesne@intel.com>
+
+ * gst-libs/gst/pbutils/codec-utils.c:
+ pbutils: handle more H.264 profiles and levels.
+ Recognize H.264 Level 5.2, as exposed by modern 2160p30+ streams,
+ i.e. commonly known as 4K. Also add initial support for handling
+ Annex.G (SVC) profiles.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732269
+ Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
+
+2014-06-26 04:27:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Bump iso mp4 typefinder to PRIMARY. Add mp4 extension hint.
+ Fixes a problem with at least one file being detected incorrectly as
+ DTS because there's DTS packets early enough in the file.
+
+2014-06-23 01:02:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/rtpbasedepayload.c:
+ tests: fix vararg handling in rtpbasedepayload unit test
+ Makes it pass on 32-bit systems.
+
+2014-06-23 00:33:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/rtpbasepayload.c:
+ tests: fix vararg handling in rtpbasepayload unit test
+ Makes it pass on 32-bit systems.
+
+2014-06-22 20:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: Filter out ANY capsfeatures from the converter caps
+ We can't convert to ANY capsfeatures, they are only there so that we
+ can passthrough whatever downstream can support... but we definitely
+ don't want to return them to upstream.
+
+2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.3 ===
+
+2014-06-22 18:07:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 1.3.3
+
+2014-06-22 17:25:42 +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
+
+2014-06-22 14:23:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/da.po:
+ * po/de.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sr.po:
+ * po/uk.po:
+ po: Update translations
+
+2014-06-20 11:00:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * tests/check/libs/audiodecoder.c:
+ audiodecoder: Don't be too picky about the output frame counter
+ With most decoder libraries, and especially when accessing codecs via
+ OpenMAX or similar APIs, we don't have the ability to properly related
+ the output buffers to a number of input samples. And could e.g. get
+ a fractional number of input buffers decoded at a time.
+ Previously this would in the end lead to an error message and stopped
+ playback. Change it to a warning message instead and try to handle it
+ gracefully. In theory the subclass can now get timestamp tracking
+ wrong if it completely misuses the API, but if on average it behaves
+ correct (and gst-omx and others do) it will continue to work properly.
+ Also add a test for the new behaviour.
+ We don't change it in the encoder yet as that requires more internal logic
+ changes AFAIU and I'm not aware of a case where this was a problem so far.
+
+2014-06-12 12:36:26 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst/tcp/gsttcpserversrc.c:
+ tcpserversrc: close the server socket after accepting a connection
+ g_socket_accept() is only called once for a server socket. So
+ keeping the socket open ist just confusing possible clients.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731566
+
+2014-06-13 10:04:47 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/tcp/gsttcpclientsrc.c:
+ tcpclientsrc: return FLUSHING when select() is canceled
+ https://bugzilla.gnome.org/show_bug.cgi?id=731567
+
+2014-06-12 13:23:29 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst/tcp/gsttcpserversrc.c:
+ tcpserversrc: return FLOW_FLUSHING instead of an error when accept/select is canceled
+ Canceling the accept/select happens when the source is shut down. This is
+ not an error and the GST_FLOW_ERROR causes problems when only part of the
+ pipeline is shut down.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731567
+
+2014-06-12 11:55:59 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/sdp/gstmikey.c:
+ mikey: Fix Wall to NTP conversion
+ We are scaling from a unit in microseconds to a unit in ((1 << 32) per seconds).
+ We therefore scale the microseconds values by:
+ value of a second in the target unit (1 << 32)
+ --------------------------------------------------------------
+ value of a second in the origin format (1 000 000 microsecond)
+
+2014-06-06 12:18:49 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: allow unset seek stop time in push mode
+
+2014-06-11 12:50:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ docs: add streamsynchronizer to documentation
+
+2014-06-11 12:43:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ docs: add playsink element to documentation
+
+2014-06-11 10:53:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ docs: add navigation interface to docs
+
+2014-06-10 12:59:53 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: add send_event handler for flushing
+ Adds a send_event handling for allowing appsrc to flush its internal
+ data, allowing users to flush the pipeline without setting it to null.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724231
+
+2014-06-09 21:05:00 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst/videoscale/vs_fill_borders.c:
+ * gst/videoscale/vs_image.h:
+ videoscale: vs_image: strides are a gsize
+ The strides that are set from the GstVideoInfo structs are
+ a gsize. Using an int can cause overflows when dealing with large
+ enough images
+ https://bugzilla.gnome.org/show_bug.cgi?id=731195
+
+2014-06-09 19:44:56 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/video-info.c:
+ * tests/check/libs/video.c:
+ video: avoid overflows when doing int operations for size
+ size is a gsize, so cast the operands to it to avoid overflows
+ and setting wrong value to the video size.
+ Includes tests.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731195
+
+2014-06-09 10:53:03 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: Remove unneeded check
+ running timestamps are guaranteed to be positive and valid since the
+ GstVideoEncoder base class will clip incoming buffers
+ CID #1139797
+
+2014-06-09 10:38:53 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/vorbis/gstvorbisenc.c:
+ vorbisenc: add missing va_end in variadic function
+ Coverity 1139944
+
+2014-06-06 10:35:31 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * tests/check/libs/videodecoder.c:
+ tests: fix uninitialized variable use in video decoder test
+
+2014-06-05 15:35:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Also catch CODEC_NOT_FOUND errors and delay them until all decodebins are done
+
+2014-06-04 17:00:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Ignore missing-plugin messages unless all decodebins post one
+ When playing RTSP streams there will be one decodebin per stream. If some of
+ them fail because of a missing plugin we should not fail completely but play
+ the supported streams at least.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730868
+
+2014-06-04 14:14:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Do async-done on expose errors too
+
+2014-05-20 12:28:15 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ dmabuf: fix checking mmap flags
+ A simple '&' is not sufficiant. With mmapping_flags == PROT_READ and
+ prot == PROT_READ|PROT_WRITE the check produces the wrong result.
+ Change the check to make sure that prot is a subset of mmapping_flags.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730559
+
+2014-06-03 15:16:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/alsa/gstalsasink.c:
+ alsasink: make gst-ident happy
+
+2014-06-03 15:10:33 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/alsa/gstalsasink.c:
+ alsasink: fix occasional crash intersecting invalid values
+ When a pipeline using alsasink and push mode upstream fails
+ to preroll, the following state will be the case:
+ - A loop upstream will be PAUSED, pushing a first buffer
+ - alsasink will be READY, pending PAUSED, because async
+ On error, the pipeline will switch to NULL. alsasink is in
+ READY, so goes to NULL immediately. It zeroes its cached
+ caps. Meanwhile, the upstream loop can cause a caps query,
+ conccurent with the state change. This will use those cached
+ caps. If the zeroing happens between the NULL test and the
+ dereferencing, GStreamer will critical down in the GstValue
+ code.
+ Since it appears that such a gap between states (PAUSED
+ and pushing upstream, and NULL downstream) is expected, we
+ need to protect the read/write access to the cached caps.
+ This fixes the critical.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=731121
+
+2013-10-14 18:56:55 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * tests/check/libs/videodecoder.c:
+ videodecoder: Keep still meaningfull pending events on FLUSH_STOP
+ Only EOS and segment should be deleted in that case.
+ + Add a testcase
+ https://bugzilla.gnome.org/show_bug.cgi?id=709868
+
+2013-10-14 18:48:08 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * tests/check/libs/audiodecoder.c:
+ audiodecoder: Keep still meaningfull pending events on FLUSH_STOP
+ Only EOS and segment should be deleted in that case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709868
+
+2013-10-14 18:45:10 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * tests/check/libs/videoencoder.c:
+ videoencoder: Keep still meaningfull pending events on FLUSH_STOP
+ Only EOS and segment should be deleted in that case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709868
+
+2013-10-10 18:50:17 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/encoding/gststreamsplitter.c:
+ streamsplitter: Keep still meaningfull pending events on FLUSH_STOP
+ Only EOS and segment should be deleted in that case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709868
+
+2013-10-10 18:48:47 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * tests/check/libs/audioencoder.c:
+ audioencoder: Keep still meaningfull pending events on FLUSH_STOP
+ Only EOS and segment should be deleted in that case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709868
+
+2014-06-02 12:40:27 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggstream.c:
+ oggstream: consider all opus packets as "keyframes"
+ This lets oggdemux determine they are not delta units, and removes
+ spurious per packet warnings about being unable to determine the
+ packet's keyframeness.
+
+2014-05-12 17:13:50 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/sdp/gstmikey.c:
+ mikey: Free MikeyPayload in error cases
+ CID #1212136
+
+2014-03-16 14:27:30 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst/playback/gstdecodebin2.c:
+ * tests/check/elements/decodebin.c:
+ decodebin: aggregate buffering messages
+ Aggregate buffering messages to only post the lower value
+ to avoid setting pipeline to playing while any multiqueue
+ is still buffering.
+ There are 3 scenarios where the entries should be removed from
+ the list:
+ 1) When decodebin is set to READY
+ 2) When an element posts a 100% buffering (already implemented)
+ 3) When a multiqueue is removed from decodebin.
+ For item 3 we don't need to handle it because this should only
+ happen when either 1 is hapenning or when it is playing a
+ chained file, for which number 2 should have happened for the
+ previous stream to finish
+ https://bugzilla.gnome.org/show_bug.cgi?id=726423
+
+2014-05-28 10:23:24 +0100 Philip Withnall <philip.withnall@collabora.co.uk>
+
+ * gst-libs/gst/audio/audio-format.c:
+ audio: Add a missing precondition to gst_audio_format_from_string()
+ https://bugzilla.gnome.org/show_bug.cgi?id=730874
+
+2014-05-26 20:57:30 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/libs/audiodecoder.c:
+ * tests/check/libs/videodecoder.c:
+ tests: videodecoder: audiodecoder: add tests for eos after segment
+ Tests that pushing a buffer after the segment returns EOS
+
+2014-05-26 21:24:07 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: actually return the push result in backwards playback
+ It was always returning _OK regardless of what downstream returned
+
+2014-05-26 12:44:48 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: return EOS when segment is over
+ if a buffer is clipped by being completely out of segment, check if this
+ buffer is after the end of the segment and return EOS upstream
+ https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2014-05-26 12:44:38 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: return EOS when segment is over
+ if a buffer is clipped by being completely out of segment, check if this
+ buffer is after the end of the segment and return EOS upstream
+ https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2014-05-26 11:45:29 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: use new gstutils helper GstFlowCombiner
+ Fixes the handling of GST_FLOW_EOS by using the helper object
+ from gstutils that does the correct combination of flow returns.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2014-05-10 18:32:28 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: Use aux vars to minimize critical region
+ This avoid dead lock between gst_audio_encoder_finish_frame() and
+ gst_opus_enc_get_property().
+ Also, now bytes var is set into protected section.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729882
+
+2014-05-23 19:21:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: play: use cubic volume factor when adjusting volume
+ This is more natural and better-suited for a playback application.
+
+2014-05-21 13:23:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.2 ===
+
+2014-05-21 13:06:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * common:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 1.3.2
+
+2014-05-21 12:01:15 +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
+
+2014-05-21 10:50:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 211fa5f to 1f5d3c3
+
+2014-05-21 10:43:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/libs/video.c:
+ video: And check comparison for real
+
+2014-05-21 10:40:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/libs/video.c:
+ video: Fix broken comparison in unit test
+ libs/video.c:540:50: error: comparison of constant 2 with boolean expression is always false
+ [-Werror,-Wtautological-constant-out-of-range-compare]
+ && !GST_VIDEO_INFO_N_PLANES (&vinfo) > 2) {
+ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ^ ~
+
+2014-05-20 15:59:53 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp-transport: clarify port usage
+ Comment in the docs what the client_port and server_port fields are used
+ for in TCP mode (if the application wants to set those values).
+
+2014-05-20 11:18:56 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ dmabuf: share the mapping with shared copies of the memory
+ With lots of shared memory instances (e.g. created by a RTP payloader) the
+ overhead of duplicating the file descriptor and creating extra mappings is
+ significant. To avoid this, the parent memory maps the whole region and the
+ shared copies just reuse the same mapping.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730441
+
+2014-05-19 13:28:52 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Add read source on write socket.
+ Add a read source on write socket when lost tunnel.
+ To be able to detect when clint closes get channel.
+ This is already done in gst_rtsp_source_dispatch_write but
+ only when the queue is empty.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730368
+
+2014-05-20 09:48:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Always take the playsink lock when adding or removing pad probes
+ Otherwise we might end up inside the callback without having stored
+ the probe id... then try to remove that probe (not!) from the callback
+ and wait forever for the pad to unblock.
+
+2014-05-19 13:57:41 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/alsa/gstalsasink.c:
+ alsasink: pass correct error to g_strerror
+ The error we get is a negated errno.
+ While there, fix a couple typos in messages.
+
+2014-05-19 11:17:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: Free playlist_file string if only printing the version
+
+2014-05-13 14:08:20 +0600 Anuj Jaiswal <anuj.jaiswal@samsung.com>
+
+ * tools/gst-play.c:
+ audio_sink and video_sink leakage fixed
+ https://bugzilla.gnome.org/show_bug.cgi?id=730010
+
+2014-05-13 11:51:55 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Don't use argument for local storage
+ By re-using the uri argument for storing local data, we could end up in
+ a situation where we would free uri ... which would actually be the
+ string passed in argument.
+ Instead explicitely use a local variable. Fixes double-free issues.
+ CID #1212176
+
+2014-05-12 13:18:50 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/video-info.c:
+ video-info: Also check the stride and offset are equal
+ gst_video_info_is_equal() was not checking if stride and offset
+ had changed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729896
+
+2014-05-12 17:17:07 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Free data after removing it from the list
+ While it wouldn't have caused any failures (g_list_remove doesn't dereference
+ the provided pointer), it does make the code cleaner.
+ CID #1212174
+
+2014-05-12 17:15:17 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/sdp/gstmikey.c:
+ mikey: Actually replace payload ...
+ This function is intented to replace the payload, let's actually do that
+ instead of putting back the same (freed) payload
+ CID #1212175
+
+2014-05-12 17:13:50 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/sdp/gstmikey.c:
+ mikey: Free MikeyPayload in error cases
+ CID #1212135
+ CID #1212136
+ CID #1212137
+ CID #1212138
+
+2014-05-10 23:50:44 +0200 Thibault Saunier <tsaunier@gnome.org>
+
+ * ext/pango/gstbasetextoverlay.c:
+ pango: Do not try to add a feature to a caps features ANY
+ It does not makes sense and asserts
+
+2014-05-09 15:32:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: fix leaks in error code paths
+ CID 1212133
+
+2014-05-06 11:12:19 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Reset control_stream.
+ Reset control_stream when gst_rtsp_connection_close.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729632
+
+2014-04-15 14:51:46 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Retry setting configuration with modified config
+ Buffer pool set_config() may return FALSE if requested configuration needed small
+ changes. Reget the config and try setting it again. This ensure we have a configured
+ pool if possible.
+
+2014-05-08 17:10:26 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: use downloadbuffer for download buffering
+ Use the new downloadbuffer element to implement the download buffering
+ feature
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680183
+
+2014-05-06 13:01:32 -0400 Luis de Bethencourt <luis@debethencourt.com>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: push eos event when empty pad data
+ If gst_ogg_mux_queue_pads returns NULL it means we are at EOS, because we get a
+ NULL buffer and this function never sets bestpad.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729315
+
+2014-05-06 08:07:38 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
+
+ * configure.ac:
+ configure: Use X11 detection macro from common
+ https://bugzilla.gnome.org/show_bug.cgi?id=729621
+
+2014-05-06 07:51:11 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/examples/playback/playback-test.c:
+ examples: playback-test: fix crashes when setting buffer-size
+ playbin's buffer-size property takes a gint, not a gint64,
+ so only pass the bits expected to the vararg function, or
+ the terminator might not be found, leading to crashes, esp.
+ with negative numbers.
+ Spotted by Ravi Kiran K N <ravi.kiran@samsung.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=729617
+
+2014-05-06 07:50:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/examples/playback/playback-test.c:
+ examples: fix indentation of playback-test
+
+2014-05-06 08:13:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/examples/playback/playback-test.c:
+ Revert "playback-test: Set buffer-size only for non-negative size"
+ This reverts commit 07a637e2847d56d0f2b0c0ac9095bf37dd324e26.
+
+2014-05-06 11:31:18 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tests/examples/playback/playback-test.c:
+ playback-test: Set buffer-size only for non-negative size
+ https://bugzilla.gnome.org/show_bug.cgi?id=729617
+
+2014-05-05 23:29:44 -0400 Luis de Bethencourt <luis@debethencourt.com>
+
+ * win32/common/libgstpbutils.def:
+ win32: Update defs file
+ commit 622007e7db7e3d32bf8e04e673e057897b646220 added the function
+ gst_discoverer_info_get_missing_elements_installer_details (). It needs to be
+ added to the defs file.
+
+2014-05-04 15:54:54 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
+
+ * configure.ac:
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Link to ws2_32 on Windows
+ Needed for getsockname and setsockopt
+ https://bugzilla.gnome.org/show_bug.cgi?id=729514
+
+2014-05-04 15:54:06 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
+
+ * configure.ac:
+ Make X11 detection more precise
+ Don't be content with just X11/Xlib.h, check for X11/XKBlib.h as well.
+ This prevents false positives (for example, from partial X11 headers
+ installed by tcl/tk).
+ https://bugzilla.gnome.org/show_bug.cgi?id=729513
+
+2014-05-04 15:57:35 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
+
+ * tests/examples/playback/playback-test.c:
+ tests: fix printf format compiler warning in playback test on win32
+ https://bugzilla.gnome.org/show_bug.cgi?id=729515
+
+2014-05-04 18:14:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/.gitignore:
+ Add new unit test binary to .gitignore
+
+2014-01-14 15:39:55 +0100 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ * gst-libs/gst/pbutils/pbutils-private.h:
+ * tools/gst-discoverer.c:
+ discoverer: Add APIs to simply get installer details for missing plugins
+ Currently the API is far from optimal and the user has to work around
+ our badly defined API to simply install missing plugins.
+ API:
+ new:
+ gst_discoverer_info_get_missing_elements_installer_details
+ deprecated:
+ gst_discoverer_info_get_misc
+ gst_discoverer_stream_info_get_misc
+ https://bugzilla.gnome.org/show_bug.cgi?id=720596
+
+2014-05-03 20:48:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+2014-05-03 18:57:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/Makefile.am:
+ textoverlay: Link unit test with the local version of the library, not an installed one
+
+=== release 1.3.1 ===
+
+2014-05-03 17:50:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * 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-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-libs/gst/audio/gstaudiopack-dist.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-plugins-base.doap:
+ * gst/adder/gstadderorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/videoconvert/gstvideoconvertorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.c:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ Release 1.3.1
+
+2014-05-03 17:48:04 +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
+
+2014-05-03 17:22:10 +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
+
+2014-05-02 19:09:59 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ * tests/check/libs/rtpbasepayload.c:
+ rtpbasepayload: Implement reconfigure event & renegotiation without subclass
+ Implement the reconfigure event, also do correct downstream caps negotiation
+ if the subclass doesn't implementy set_caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=725361
+
+2014-05-02 19:09:44 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/libs/rtpbasepayload.c:
+ tests/check/libs/rtpbasepayload.c: Run gst-indent
+ https://bugzilla.gnome.org/show_bug.cgi?id=725361
+
+2014-05-03 10:14:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From bcb1518 to 211fa5f
+
+2014-05-02 18:30:16 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Save the PT after fixating
+
+2014-05-02 19:36:34 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtspdefs: remove outdated comments
+
+2014-05-02 15:09:35 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: avoid underflow in size calculation
+
+2014-05-01 19:31:09 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: do not parse caps for not using it
+ Saving some cpu
+
+2014-01-03 11:06:22 +0100 John Bassett <john.bassett@pexip.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: restrict initial random sequence number to be <= 32767
+ In order to prevent SRTP roll over counter issues the initial sequence
+ number is restricted to <= 32767. This is recommended by RFC 4568 section 6.4.
+
+2014-05-01 15:11:04 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Add some more gobject-introspection annotations for bindings
+ https://bugzilla.gnome.org/show_bug.cgi?id=729123
+
+2014-05-01 13:15:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Don't block on non-serialized events
+ https://bugzilla.gnome.org/show_bug.cgi?id=729321
+
+2014-05-01 13:08:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Don't block on non-serialized events
+ https://bugzilla.gnome.org/show_bug.cgi?id=729321
+
+2014-05-01 13:06:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: Don't block on non-serialized events
+ https://bugzilla.gnome.org/show_bug.cgi?id=729321
+
+2014-05-01 13:05:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Don't block on non-serialized events
+ https://bugzilla.gnome.org/show_bug.cgi?id=729321
+
+2014-04-30 11:06:27 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: check claimed data size against available size
+ Coverity 1208773
+
+2014-04-23 08:06:36 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Empty queue when flush.
+ Empty the watchs queue when calling
+ gst_rtsp_watch_set_flushing with flushing variabel is TRUE.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728772
+
+2014-03-16 16:09:36 +0100 Ognyan Tonchev <otonchev@gmail.com>
+
+ * tests/check/libs/rtspconnection.c:
+ rtspconnection: Add more tests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728907
+
+2014-04-29 10:15:47 -0400 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: fix undefined behaviour of left-shift
+ With a small type for the color values being left-shifted, the result is
+ undefined and it could potentially overflow.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729195
+
+2014-04-29 10:59:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgstsdp.def:
+ win32: fix export files again
+ Revert unintended parts of d8a0927930a87a2eb60d4c98cb3fea8aed911b27
+
+2014-04-29 11:39:18 +0200 Christian Fredrik Kalager Schaller <uraeus@linuxrising.org>
+
+ * gst-plugins-base.spec.in:
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgstsdp.def:
+ Add mikey.h file
+
+2014-04-29 09:58:21 +0200 Haakon Sporsheim <haakon@pexip.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Make caps writable before fixating
+ https://bugzilla.gnome.org/show_bug.cgi?id=729114
+
+2014-04-29 09:54:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdpmessage: Add array length annotation to gst_sdp_message_parse_buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=729123
+
+2014-04-29 08:46:02 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: fix memory leak when gst_rtp_buffer_map fails
+ Make sure rtp->data[3] is set before jumping to error path.
+ https://bugzilla.gnome.org/show_bug.cgi?id=729117
+
+2014-04-28 18:47:06 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tools/gst-play.c:
+ gst-play: add option to supply media files from playlist file
+ https://bugzilla.gnome.org/show_bug.cgi?id=728845
+
+2014-04-27 00:49:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/gio/gstgiobasesink.c:
+ giobasesink: we mustn't change the format of a query response
+ Not even in the DEFAULT case. That's bad 0.10 behaviour, no caller
+ is ever going to check the format of the response.
+
+2014-04-27 00:25:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/playback/gstplay-enum.c:
+ playbin: add nick for soft colorbalance play flag to fix gst-inspect
+ Fix gst-inspect-1.0 playbin criticals when printing the
+ flags, which was caused by a missing nick name for one
+ of the flags.
+
+2014-04-26 23:26:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisenc.c:
+ * ext/vorbis/gstvorbisparse.c:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtphdrext.c:
+ * gst-libs/gst/rtp/gstrtppayloads.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ * gst-libs/gst/sdp/gstmikey.c:
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultihandlesink.c:
+ * gst/tcp/gstmultioutputsink.c:
+ * gst/tcp/gstmultisocketsink.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ docs: remove outdated and pointless 'Last reviewed' lines from docs
+ They are very confusing for people, and more often than not
+ also just not very accurate. Seeing 'last reviewed: 2005' in
+ your docs is not very confidence-inspiring. Let's just remove
+ those comments.
+
+2014-04-25 17:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/gio/gstgiobasesink.c:
+ giobasesink: Implement handling of the SEEKING query
+
+2014-04-25 11:30:37 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Plug caps leaks
+ We were returning in various places without unreffing the caps, and
+ we were also leaking (overwriting) the caps we got from _get_current_caps()
+ Spotted by Haakon Sporsheim in #gstreamer
+
+2014-04-22 18:28:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioresample/resample.c:
+ audioresample: Don't left-shift into the sign bit, instead use unsigned integers
+
+2014-04-22 00:21:01 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: avoid adding empty strings
+ Fixes assertion with some jpeg files
+
+2014-04-21 15:35:32 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tools/gst-play.c:
+ play: Improve pipeline states
+ First set the pipeline to the PAUSED state to check if we are dealing
+ with a live pipeline or not. Then move to the desired state.
+ If we don't do this, it is possible that we receive a BUFFERING message
+ before we know that the pipeline is live and we would set the pipeline
+ to PAUSED and deadlock.
+
+2014-04-21 15:33:10 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * tools/gst-play.c:
+ play: Update buffering state for live pipelines
+ Update the buffering variable, even for live pipelines so that we don't
+ print \n for each buffering message.
+
+2014-04-16 19:53:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ videoframe: Initialise GstVideoFrame to zeroes if mapping fails
+ This should allow for more meaningful errors. Dereferencing NULL
+ is more useful information than dereferencing a random address
+ happened to be on the stack.
+
+2014-04-16 11:43:40 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ exiftag: catch buffer mapping failure
+ Might be what caused:
+ Coverity 1139734
+
+2014-04-15 19:17:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/audioresample.c:
+ audioresample: Fix memory leaks in test
+
+2014-04-15 19:16:44 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/resample.c:
+ audioresample: Fix up indention
+
+2014-04-15 19:16:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioresample/resample_sse.h:
+ audioresample: Fix out of bounds memory accesses
+
+2014-04-15 13:57:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ pango: Make static caps actually static to fix a memory leak
+
+2014-04-15 13:54:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/videotestsrc.c:
+ videotestsrc: Fix memory leak in test
+
+2014-04-15 13:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/encodebin.c:
+ encodebin: Fix memory leak in test
+
+2014-04-15 13:48:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Free preset name in finalize
+
+2014-04-15 13:39:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: Clear Ogg streams before initing them
+ They might've been inited before, in which case we leak
+ memory when initing them again without clearing.
+
+2014-04-15 13:03:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/audioconvert.c:
+ audioconvert: Fix leaks in unit test
+
+2014-04-15 11:55:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/libs/videodecoder.c:
+ * tests/check/libs/videoencoder.c:
+ videoencoder/decoder: Fix memory leaks in the tests
+
+2014-04-15 11:53:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/libs/audiodecoder.c:
+ audiodecoder: Actually allocate enough memory for 64 bits, not just 32 bits
+ Also fix a memory leak.
+
+2014-04-15 11:43:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/libs/audioencoder.c:
+ audioencoder: Fix memory leaks in unit test
+
+2014-04-15 10:29:12 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/libs/rtp.c:
+ rtp: Fix GBytes memory leak in test
+
+2014-04-12 07:10:36 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasedepay: add stats property
+ Add a stats property that holds a structure with all the current
+ values of the depayloader.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=646577
+
+2014-04-12 06:43:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: update docs
+
+2014-04-12 06:27:36 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: add current timestamp and seqnum offset to stats
+ Expose the current timestamp and seqnum offset in the stats
+ See https://bugzilla.gnome.org/show_bug.cgi?id=646577
+
+2014-04-11 10:24:10 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttextrender.h:
+ textrender: push segment event after caps event
+ Fixes warning "Sticky event misordering, got 'segment' before 'caps'".
+
+2014-04-10 16:08:29 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggstream.c:
+ oggstream: use G_GUINT64_CONSTANT instead of ll suffix
+ Thanks slomo for pointing out it's not standard.
+
+2014-04-10 15:55:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * sys/xvimage/xvcontext.c:
+ xvimage: remove dead code
+ matching_attr can not be NULL here, we've tested that away a few
+ lines beforehand.
+ Coverity 1139655
+
+2014-04-10 15:51:05 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: bail out on unsupported caps
+ This avoids using uninitialized data (and properly rejects caps).
+ Coverity 1139898
+
+2014-04-10 15:16:03 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: remove pointless checks for data being NULL
+ It was already checked in an early out, and as it's only
+ incremented for at most the size of the passed buffer, it
+ can only become NULL in an address wraparound.
+ While there, don't cast away const on a pointer.
+ Coverity 1139845
+
+2014-04-10 13:34:58 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: consider "no demuxer" case to not have dynamic pads
+ This fixes a possible NULL dereference.
+ Coverity 1195146
+
+2014-04-10 13:28:30 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: guard against gst_pad_get_peer returning NULL
+ If it does, the pad may be leaked if it's a request pad, though.
+ Coverity 1139799
+
+2014-04-10 13:26:42 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/encoding/gstencodebin.c:
+ encodebin: guard against pathological NULL dereference
+ Coverity 1139798
+
+2014-04-10 12:32:24 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioresample/resample.c:
+ audioresample: reject 0 denominator when creating resampler
+ Coverity 1195140, 1195139, 1195138
+
+2014-04-10 12:14:48 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/video/video-overlay-composition.c:
+ video-overlay-composition: guard against NULL pointer dereference on error
+ If gst_video_overlay_rectangle_apply_global_alpha is called with
+ a rectangle with unsuitable alpha, expanding the alpha plane will
+ fail, and thus lead to dereferencing a NULL src pointer. It's not
+ certain this will happen in practice, as the function is static
+ and callers might ensure suitable alpha before calling, but there
+ is no apparent explicit such check.
+ Add prologue asserts for proper alpha to explicitely prevent this.
+ Coverity 1139707
+
+2014-04-10 12:10:47 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/video/gstvideometa.c:
+ videometa: fix texture_type memcpy size
+ Coverity 1139589, 1139588
+
+2014-04-10 11:19:26 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdpmessage: fix multi statement macros
+ Wasn't playing nice with an if statement below.
+ Coverity 1139767
+
+2014-04-10 11:14:25 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiocdsrc.c:
+ audiocdsrc: guard aginst overflow
+ An audio CD may contain about a tenth of the samples 32 bit can
+ represent, so it doesn't seem likely this will be hit in practice.
+ Coverity 1139805
+
+2014-04-10 12:30:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: descriptions: default to systemstream=false for partial video/mpeg caps
+ Assume systemstream=false for video/mpeg caps where that field
+ is missing.
+
+2014-04-10 10:57:53 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: avoid possible sample count overflow
+ At 48 kHz, 2<<31 samples is reached before 13 hours so it
+ sounds plausible this would be hit.
+ Coverity 1139800, 1139801
+
+2014-04-10 10:45:21 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: fix comparison to unset timestamp
+ Also rejects negative timestamps that aren't GST_CLOCK_TIME_NONE.
+ Coverity 1139797
+
+2014-04-10 10:33:46 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggstream.c:
+ oggstream: fix a few left shifts operations on 32 bits cast to 64 bits
+ This should not cause any actual bug since Theora and Daala have
+ a maximum shift of 31, and a packet duration of 2^31 seems very
+ implausible. But it fixes:
+ Coverity 1139804, 1139803, 1139802
+
+2014-04-10 10:29:34 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggstream.c:
+ oggstream: remove NULL test after dereference
+ And add NULLness asserts at top of function. The only call
+ to this passes local variable pointers, so non NULL.
+ Coverity 206375
+
+2014-04-10 10:25:46 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: test for failure to return tag
+ It should really not happen unless the tag list it corrupt,
+ but the API returns a failure code so we may as well use it.
+ Coverity 1139595
+
+2014-04-10 10:22:43 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: do not dereference NULL pad in warning message
+ Coverity 1197695
+
+2014-04-10 09:18:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-event.c:
+ video-event: Update the running times in the force-keyunit events from the pad offsets
+
+2014-04-09 16:03:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: In adaptive streaming mode, only have a fixed buffer limit for the non-buffering multiqueue
+
+2014-04-09 11:02:00 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusheader.c:
+ opus: add missing va_end in variadic function
+ Coverity 1139944
+
+2014-04-08 15:43:50 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: guard against address parse errors.
+
+2014-03-25 17:11:34 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/adder/gstadder.c:
+ adder: rework the logic to check if eos has to be sent.
+ Checking the size available was incorrect, and the infos
+ for per-pad EOS are available.
+ Same logic as audiomixer.
+ fixes: https://bugzilla.gnome.org/show_bug.cgi?id=727025
+
+2014-04-08 12:46:21 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: parse channels field from compressed audio caps
+ Also parse channels as an optional field in the caps for compressed
+ audio formats.
+
+2014-04-06 22:26:20 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Consider all caps for overlays, not just the first.
+ Check all supported caps on the overlay video pad, not just the
+ first of (possibly) many.
+
+2014-04-05 13:25:46 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play-1.0.1:
+ tools: update gst-play-1.0 man page
+
+2014-04-02 07:20:43 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: do not deactivate the bufferpool, just unref
+ Videodecoder does late renegotiation, it will wait for the next
+ buffer before renegotiating its caps and bufferpool. It might happen
+ that downstream element switched from passthrough to non-passthrough
+ and sent a reconfigure upstream (that caused this renegotiation).
+ This downstream element will ask the video sink below for the bufferpool
+ with an allocation query and will get the same bufferpool that
+ videodecoder is holding, too.
+ When renegotiating, if videodecoder deactivates its bufferpool it
+ might be deactivating the bufferpool that some element downstream
+ is using and cause the pipeline to fail.
+ https://bugzilla.gnome.org/show_bug.cgi?id=727498
+
+2014-02-24 11:17:05 -0500 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: clip start samples to match clipped start time
+ Clock slaving can clip start time to zero, giving us a shorted
+ duration than we originally got. To keep in sync, we must then
+ discard the samples falling before that zero timestamp.
+ This possibly fixes random distortion caused by constant PA
+ underflows which are never resynced.
+
+2014-04-04 17:36:04 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/sdp/gstmikey.c:
+ * gst-libs/gst/sdp/gstmikey.h:
+ * tests/check/libs/mikey.c:
+ * win32/common/libgstsdp.def:
+ mikey: Fix the KEMAC payload
+ The KEMAC payload actually needs to have subpayloads and the key should
+ go into the KEY_DATA subpayload. Add support for subpayloads and
+ implement the KEY_DATA payload.
+ Add some pointers to the conversion functions that allow us to add
+ encryption and decryption later.
+
+2014-04-04 02:14:50 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Drop reference to any source element in NULL state
+ Drop the reference instead of waiting for either finalize(), or
+ for a new source when reused. Everyone else already forgot about
+ the old source.
+
+2014-04-01 10:38:23 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * win32/common/libgstrtsp.def:
+ rtspconnection: Added gst_rtsp_watch_set_flushing to list.
+ Added gst_rtsp_watch_set_flushing to list in file
+ libgstrtsp.def
+
+2014-03-30 18:26:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Always drain the decoder after a discont group in reverse playback mode
+
+2014-03-30 17:54:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Flush the decoder once per discont group, not once per keyframe
+
+2014-03-30 17:54:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Handle reverse playback with multiple GOPs per discont group properly
+ baseparse will reverse each GOP for us already, so the segment events can
+ be after our keyframe. Make sure to get it and all other relevant sticky
+ events before starting to decode.
+
+2014-03-29 10:23:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Log event types of events that are pushed downstream
+
+2014-03-27 20:15:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: In reverse playback mode we need to finish the subclass after passing all frames to it
+
+2014-03-28 09:32:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtspconnection: add flush method
+ Add a method to set/unset the flushing state that makes _wait_backlog()
+ unlock.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=725898
+
+2014-03-27 16:43:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/ximage/ximagesink.c:
+ ximagesink: only extrapolate alpha mask for 32-bit depth
+ Instead of passing bogus alpha mask values when there's no alpha.
+ https://bugzilla.gnome.org/show_bug.cgi?id=727188
+
+2014-03-25 11:14:51 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/sdp/gstmikey.c:
+ mikey: fix return values of g_return_*
+
+2014-03-25 11:07:34 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ rtsptransport: UDP is also default for SAVP and AVPF
+
+2014-03-20 12:29:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/sdp/gstmikey.c:
+ * gst-libs/gst/sdp/gstmikey.h:
+ docs: add MIKEY docs
+
+2014-03-15 18:46:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/sdp/gstmikey.c:
+ * gst-libs/gst/sdp/gstmikey.h:
+ * tests/check/Makefile.am:
+ * tests/check/libs/mikey.c:
+ * win32/common/libgstsdp.def:
+ mikey: add MIKEY parsing helpers
+ MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption
+ parameters between a sender and a receiver in a secure way.
+ This library implements a subset of the features, enough to implement
+ RFC 4567, using MIKEY in SDP and RTSP.
+
+2014-03-16 17:04:44 +0100 Ognyan Tonchev <otonchev@gmail.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fix minor memory leaks in error handling
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726642
+
+2014-03-16 17:06:02 +0100 Ognyan Tonchev <otonchev@gmail.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fix connection_poll()
+ * Only check for conditions we are interested in.
+ * Makes no sense to specify G_IO_ERR and G_IO_HUP in condition, they
+ will always be reported if they are true.
+ * Do not create timed source if timeout is NULL.
+ * Correctly wait for sources to be dispatched, context_iteration() is
+ not guaranteed to always block even if set to do so.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726641
+
+2014-03-20 09:18:31 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: add pt and ssrc to stats
+
+2014-03-16 08:34:30 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/elements/decodebin.c:
+ * tests/check/elements/decodebin2.c:
+ tests: decodebin: port old decodebin2 test for parser and decoder linking
+ They were in the old decodebin2.c tests file and were never ported.
+ Now we can get rid of decodebin2.c
+
+2014-03-16 17:00:38 +0100 Arun Raghavan <arun@accosted.net>
+
+ * gst/playback/gstplay-enum.c:
+ * gst/playback/gstplay-enum.h:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ * tests/examples/playback/playback-test.c:
+ playback: Add video-/audio-filter properties
+ This provides an audio-filter and video-filter property to allow
+ applications to set filter elements/bins. The idea is that these will
+ e
+ applied if possible -- for non-raw sinks, the filters will be skipped.
+ If the application wishes to force the application of the filters, this
+ can be done by setting the new flag introduced on playsink -
+ GST_PLAY_FLAG_FORCE_FILTERS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=679031
+
+2014-03-16 18:38:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplay-enum.h:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ Revert "playback: Add video-/audio-filter properties"
+ This reverts commit fb8fdedb4f4649aa33700bbc720131c1678df49f.
+
+2014-03-15 16:05:22 +0100 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ * gst/playback/gstplay-enum.h:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playback: Add video-/audio-filter properties
+ This provides an audio-filter and video-filter property to allow
+ applications to set filter elements/bins. The idea is that these will be
+ applied if possible -- for non-raw sinks, the filters will be skipped.
+ If the application wishes to force the application of the filters, this
+ can be done by setting the new flag introduced on playsink -
+ GST_PLAY_FLAG_FORCE_FILTERS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=679031
+
+2014-03-15 20:21:32 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Silence a compiler warning
+ Cast the argument into (const char *) on W32, as winsock2 expects it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=726433
+
+2014-03-15 11:24:23 +0100 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Fix documentation for what the audio chain looks like
+ https://bugzilla.gnome.org/show_bug.cgi?id=679031
+
+2014-03-11 21:58:49 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-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 plugin docs and remove old properties and signals
+ Re-generate .args and .signals file from scratch so that
+ old signals that no longer exist (such as the 'new-decoded-pad'
+ signal on decodebin) no longer show up in the documentation.
+
+2014-03-11 22:15:13 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: set a group-id on the stream-start event
+ Set a default group-id to fix a warning printed by the sink.
+
+2014-03-11 17:39:54 +0100 Christian Fredrik Kalager Schaller <uraeus@linuxrising.org>
+
+ * gst-plugins-base.spec.in:
+ Add new header file
+
+2014-03-06 12:59:08 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggmux: implement vp8 granulepos function
+ Add an extra function to the oggstream map to inform it about
+ the incoming buffers. This way oggmux can keep a count on the
+ vp8 invisible frames and calculate the granulepos correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=722682
+
+2014-03-05 16:34:42 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggmux: create vp8 header data if not provided in caps
+ vp8 stream header shouldn't be assumed to be provided in caps always
+ as this would repeat the same code in all demuxers/encoders. Instead,
+ make oggmux generate them if they are not supplied.
+ https://bugzilla.gnome.org/show_bug.cgi?id=722682
+
+2014-03-06 13:55:17 +0100 Göran Jönsson <goranjn@axis.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: gst_rtsp_watch_wait_backlog
+ New method that wait until there is room in backlog queue.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
+
+2014-03-06 13:50:27 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtspconnection: GstRTSPWatch func for tunnel GET response
+ Add a callback in GstRTSPWatch where the response to HTTP GET for
+ tunneled connections can be modified.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725878
+
+2014-03-06 15:34:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtspdefs: add RFC 4567 headers and status code
+ This new Header and status code is used for SRTP
+
+2014-03-07 17:09:24 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ decodebin: Buffer up to 5 seconds in multiqueue buffering mode
+ 2 seconds might be too small for some container formats, e.g.
+ MPEGTS with some video codec and AAC/ADTS audio with 700ms
+ long buffers. The video branch of multiqueue can run full while
+ the audio branch is completely empty, especially because there
+ are usually more queues downstream on the audio branch.
+
+2014-03-06 22:37:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Keep the number of buffers after an adaptive streaming demuxer lower
+ Usually these buffers are multiple seconds large, and having a maximum
+ of 5 buffers in the multiqueue there can use a lot of memory. Lower
+ this to 2 for adaptive streaming demuxers.
+
+2014-03-06 22:28:46 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Simplify adaptive streaming demuxer code a bit
+
+2014-03-06 17:49:09 +0000 Adrien Schwartzentruber <adrien.schwartzentruber@gmail.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ pango: demote debug WARNING to LOG for variable framerate video input
+ No need why we need to warn about that, it's perfectly allowed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=725837
+
+2014-01-30 15:41:49 +0000 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/textoverlay.c:
+ tests: add textoverlay passthrough with composition feature unit tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=721953
+
+2014-01-23 12:20:05 +0000 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ pango: basetextoverlay: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API
+ https://bugzilla.gnome.org/show_bug.cgi?id=721953
+
+2014-01-23 12:19:13 +0000 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst-libs/gst/video/video-overlay-composition.h:
+ video-overlay-composition: add GST_CAPS_FEATURE_META_GST_VIDEO_OVERLAY_COMPOSITION
+
+2014-03-04 16:51:58 +0200 Andres Gomez <agomez@igalia.com>
+
+ * REQUIREMENTS:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ docs: Removing GnomeVFS left bits
+ gnomevfs was removed time ago but there are still some left bits.
+ https://bugzilla.gnome.org/show_bug.cgi?id=725658
+
+2014-03-05 00:35:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: lower H.263 typefinder max probability
+ The typefinder returns LIKELY for as little as one possible
+ sync and no bad sync (not even taking into account how much
+ data was looked at for that). It's generally just not fit
+ for purpose, so should just not return anything like LIKELY
+ at all ever, even more so since it only recognises one out
+ of ten H263 files, and likes to mis-detect mp3s as H263.
+ https://bugzilla.gnome.org/show_bug.cgi?id=700770
+ https://bugzilla.gnome.org/show_bug.cgi?id=725644
+
+2014-03-02 11:58:58 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * tests/check/libs/rtspconnection.c:
+ rtspconnection: Call closed() when GET is closed in tunneled mode
+ This patch adds read source on the write socket in tunneled
+ mode and we get a callback when client disconnects the GET
+ channel.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725313
+
+2014-03-02 12:58:21 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst-libs/gst/video/video-format.c:
+ videoformat: Remove duplicate/incorrect section
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
+
+2014-03-02 12:54:08 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ * gst-libs/gst/video/video-format.c:
+ docs: Add annotations for return values
+ Rephrase and clarify some return value descriptions
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
+
+2014-03-02 05:06:07 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ docs: Fix argument and annotation typos
+ * colorbalance: Fix misspelled annotation
+ * rtsp: Replace incorrectly documented function argument
+ * sdp: Escape @ character to avoid gtk-doc warning
+ * video-*: Add missing annotation colon
+ * videodecoder/video-color: Fix function argument typos
+ * videoutils: Remove unknown annotation field
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
+
+2014-03-02 05:09:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * .gitignore:
+ .gitignore: Ignore gcov intermediate files
+ https://bugzilla.gnome.org/show_bug.cgi?id=725479
+
+2014-02-28 09:34:31 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From fe1672e to bcb1518
+
+2014-02-20 20:01:30 +0000 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: improve autoplug_query_caps return
+ Makes autoplug_query_caps return
+ downstream_caps + intersect_first(filter_caps, element_caps)
+ https://bugzilla.gnome.org/show_bug.cgi?id=724828
+
+2014-02-26 22:11:01 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 1a07da9 to fe1672e
+
+2014-02-26 11:43:06 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix build with older GLib versions
+ The gio/gnetworking.h header is only available since glib 2.36
+ https://bugzilla.gnome.org/show_bug.cgi?id=725206
+
+2014-02-26 11:45:24 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Add missing include
+ https://bugzilla.gnome.org/show_bug.cgi?id=725206
+
+2014-02-21 14:01:37 +0000 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst/playback/gstplaysinkconvertbin.c:
+ playsinkconvertbin: improve gst_play_sink_convert_bin_getcaps return
+ If we have the peer caps and a caps filter, return peer_caps +
+ intersect_first (filter, converter_caps) instead of
+ intersect_first (filter, peer_caps + converter_caps) and preservers
+ downstream caps preference order.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724893
+
+2014-01-31 00:06:18 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/rtp-basepayloading.c:
+ * tests/check/libs/rtpbasedepayload.c:
+ * tests/check/libs/rtpbasepayload.c:
+ tests: Refactor RTP basepayloading test into pay/depay parts
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723328
+
+2014-01-31 00:19:16 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Let caps event also configure seqnum-offset
+ Previously the sequence number kept track of by GstRTPBasePayload would
+ only be set when going from READY to PAUSED state. This meant that a
+ downstream element that attempted to configure a basepayloader by
+ setting seqnum-offset e.g. in its sinkpad's caps template would have
+ trouble configuring the basepayloader. The reason was that the caps
+ event which arrives with the desired value for seqnum-offset did not
+ arrive at the basepayloader until caps negotiation took place,
+ significantly later than the transition from READY to PAUSED.
+ The result after this patch is that the default value for the
+ seqnum-offset property, or later set values for this property, will take
+ effect when going from READY to PAUSED like before. In addition the an
+ arriving caps event will also affect the basepayloaders configured
+ sequence number as the event arrives.
+
+2014-01-31 00:18:35 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Fix payload type property boundary value
+ The payload type field in an RTP packet header is 7 bits wide, hence the
+ boundary values ought to be 0x00 and 0x7f, not the previously stated
+ values 0x00 and 0x80.
+
+2014-01-31 00:06:30 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasedepayload: Fix typos in comments
+
+2014-02-21 19:28:55 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideopool.c:
+ docs: add GstVideoPool to docs
+
+2014-02-21 09:53:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: If we have a demuxer without dynamic srcpads, just assume no-more-pads
+ Otherwise we will wait until the multiqueue after the demuxer will
+ overrun, which is clearly not needed then.
+
+2014-02-21 09:43:38 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Also make sure to not duplicate an element factory after a group
+ If we are using an adaptive stream demuxer, which outputs a non-container
+ stream, we are putting another multiqueue after the *parser* following
+ the adaptive stream demuxer. We do not want to add another instance of
+ the same parser right after this multiqueue.
+
+2014-02-20 15:38:48 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: During pre-rolling always use the auto-preroll limits on multiqueues
+ Even if we're buffering in the multiqueues.
+
+2014-02-20 15:37:54 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Pass through the seekability information when setting multiqueue limits
+
+2014-02-20 15:36:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: During exposing of pads don't set the multiqueue limits multiple times to different values
+ Instead just set them once in the very end to the correct values.
+
+2014-02-20 15:07:26 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Only enable multiqueue buffering once we're pre-rolled
+ Otherwise we will emit buffering messages not just from the last
+ multiqueue but also from previous multiqueues... confusing the
+ application with different percentages during pre-rolling.
+
+2014-02-20 15:02:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Make sure that we always have a second multiqueue for adaptive streaming demuxers
+ For adaptive streaming demuxer we insert a multiqueue after
+ this demuxer. This multiqueue will get one fragment per buffer.
+ Now for the case where we have a container stream inside these
+ buffers, another demuxer will be plugged and after this second
+ demuxer there will be a second multiqueue. This second multiqueue
+ will get smaller buffers and will be the one emitting buffering
+ messages.
+ If we don't have a container stream inside the fragment buffers,
+ we'll insert a multiqueue below right after the next element after
+ the adaptive streaming demuxer. This is going to be a parser or
+ decoder, and will output smaller buffers.
+
+2014-02-19 10:21:16 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Always use buffering in multiqueue for adaptive streams
+
+2014-02-19 10:06:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Only add a queue2 for buffering for non-adaptive streaming streams
+
+2013-02-06 08:46:58 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: pass on the buffering property for adaptive streams
+ Adaptive streams should download its data inside the demuxer, so
+ we want to use multiqueue's buffering messages to control the
+ pipeline flow and avoid losing sync if download rates are low;
+ https://bugzilla.gnome.org/show_bug.cgi?id=707636
+
+2014-02-21 19:07:59 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/.gitignore:
+ tests: add new unit tests to .gitignore
+
+2014-02-19 13:54:17 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/rtspconnection.c:
+ rtspconnection: New unit test
+ See https://bugzilla.gnome.org/show_bug.cgi?id=724720
+
+2014-02-19 13:53:06 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Remove read child source when POST is disconnected
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724720
+
+2014-02-19 16:10:25 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * win32/common/libgstrtsp.def:
+ defs: update for new rtspconnection symbols
+
+2014-02-19 01:55:50 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: allow file to go until the end in push mode
+ When seeking back to original state after duration seeks, let
+ upstream know that we want the whole file, including the last
+ byte that wasn't requested on the duration seeks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724633
+
+2014-02-19 23:54:59 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: remove unused instance variable event
+ It is never set to anything
+
+2014-02-16 17:39:35 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtspconnection: allow specifying a certificate database
+ Two new functions have been added,
+ gst_rtsp_connection_set_tls_database() and
+ gst_rtsp_connection_get_tls_database(). The certificate database will be
+ used when a certificate can't be verified with the default database.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724393
+
+2014-02-16 23:55:17 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: get rid of superfluous whitespaces
+
+2014-02-18 20:48:57 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * tests/check/elements/encodebin.c:
+ encodebin: simplify tests
+ Also use the profile helper for the ogg profile here.
+
+2014-02-18 13:08:09 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/video-info.c:
+ video: Fix NV12_64Z32 default offset and size
+ This was a regression introduced by f52fd7a68, where we started using
+ the stride to encode the dimensions in tiles. This patch simply updates
+ offset and size calculation as described in the documentation,
+ part-mediatype-video-raw.txt.
+
+2014-02-18 15:02:57 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Keep inputselector around until we release its pads
+ Otherwise there's an interesting race condition when we destroy
+ the inputselector (actually it will be destroyed later when its state
+ change message gets destroyed) and afterwards release its sinkpad.
+ This is the code path when the last channel is removed from the
+ input selector.
+ Gave this warning sometimes, for chained oggs or whenever else
+ we change decode groups:
+ GStreamer-CRITICAL **: Padname '':sink_0 does not belong to element inputselector0 when removing
+
+2014-02-18 10:42:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: never do mixing for 1->1 channel conversions
+ MONO and NONE position are the same, for example, but in
+ general there isn't much to do here for such a conversion.
+ Fixes problem in audioconvert, which would end up using
+ a mixmatrix when converting between different mono format
+ because it thinks MONO positioning is different from
+ unpositioned channels, which is not the case in this
+ special case. The mixmatrix would end up being 0.0 so
+ audioconvert would convert to silence samples.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724509
+
+2014-02-18 10:32:46 +0000 Rafał Mużyło <galtgendo@o2.pl>
+
+ * gst-libs/gst/audio/audio-info.c:
+ audio: map channels=1,channel-mask=0 to MONO instead of NONE
+ Fixes problem in audioconvert, which would end up using
+ a mixmatrix when converting between different mono format
+ because it thinks MONO positioning is different from
+ unpositioned channels, which is not the case in this
+ special case. The mixmatrix would end up being 0.0 so
+ audioconvert would convert to silence samples.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724509
+
+2014-02-16 21:24:29 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * tests/check/elements/encodebin.c:
+ encodebin: refactor tests
+ Add a new test to demo how to get missing plugin message.
+ Split some tests that unneccesarily munge unrelated checks into one test.
+
+2014-02-16 15:32:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Only remove the complete text chain if the text pad goes away
+ If the text pads does not go away we just set the overlay to silent, which
+ allows us to immediately re-enable subs later again. However before this
+ change we also released the streamsynchronizer text pads, which deadlocked
+ because there was still dataflow going on. Just do this only if we remove
+ the complete chain.
+ https://bugzilla.gnome.org/show_bug.cgi?id=683504
+
+2014-02-14 20:16:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/Makefile.am:
+ * tools/gst-play.c:
+ tools: gst-play: add volume control
+
+2014-02-13 16:03:01 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: properly flush when seeking at the beginning
+ Reset all internal status when collect pads forwards a flush-stop
+ from the pads to be able to start the stream again.
+
+2014-02-12 17:34:32 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Don't leak pad references
+
+2014-02-02 23:59:36 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ tests: Don't build disabled plugins' check tests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723492
+
+2014-02-11 16:35:45 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: First try to get the pad's current caps, then query caps
+ The caps query might give us ANY caps while the pad has fixed caps
+ configured currently.
+
+2014-02-10 16:33:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Fix memory leak in autoplugging code
+ We should not leak element factories ideally.
+
+2014-02-10 16:33:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/playbin-complex.c:
+ playbin: Fix memory leak in unit test
+
+2014-02-09 23:17:03 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Remove unused function
+
+2014-02-09 11:28:48 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiosrc.h:
+ audiosrc: Fix typo in docs
+ We read *from* the audio device, not to it.
+
+2014-02-08 20:08:29 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/opus.c:
+ opus: Remove unused variable from unit test
+
+2014-02-08 17:11:54 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/videoscale.c:
+ videoscale: Fix compiler warning in unit test
+ error: implicit conversion from enumeration type
+ 'GstFormat' to different enumeration type 'GstVideoFormat'
+
+2014-02-08 17:11:04 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/videoconvert.c:
+ videoconvert: Fix compiler warning in unit test
+ error: implicit conversion from enumeration type
+ 'GstFormat' to different enumeration type 'GstVideoFormat'
+
+2014-02-08 17:07:15 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/playback/playback-test.c:
+ playback-test: Fix types for comparisons
+ Storing a 64 bit integer in a 32 bit integer and then checking
+ for the error cases might not be ideal.
+ error: comparison of constant -9223372036854775808 with
+ expression of type 'guint' (aka 'unsigned int') is always true
+
+2014-02-08 17:02:27 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/ogg/gstoggmux.h:
+ oggmux: Fix typo in header include guard
+ clang does not like this.
+
+2014-02-08 17:01:38 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/alsa/gstalsaplugin.c:
+ alsa: Make clang happy with our g_strdup_vprintf() wrapper
+
+2014-02-07 15:33:34 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/examples/playback/playback-test.c:
+ playback-test: allow seeking outside of the range
+ For download buffer, allow seeking outside of the already downloaded
+ area.
+
+2014-02-07 02:09:10 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ basetextoverlay: use correct segment for text
+ video time uses the 'segment' and the text time should use
+ the 'text_segment'.
+ If different segments are used for video and text it would
+ lead to out of sync video/subtitles.
+
+2014-02-04 14:31:29 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/libs/rtp.c:
+ check: add some more checks
+ Add header and payload length check in case of CSRCs.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=723196
+
+2014-02-03 02:35:57 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/examples/seek/jsseek.c:
+ jsseek: Add missing HAVE_X check
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723507
+
+2014-02-04 13:55:49 +0100 Eric Trousset <etrousset@awox.com>
+
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: Forward TIME seeks upstream too, maybe upstream can handle that
+ https://bugzilla.gnome.org/show_bug.cgi?id=723597
+
+2014-01-31 23:27:03 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/audio-channels.c:
+ * gst-libs/gst/audio/gstaudiometa.c:
+ docs: doc fixes for audio library
+ Add sections docs for audiometa. Fix sections docs for audiochannels. Remove old
+ mixerutil section.
+
+2014-01-31 13:40:36 +0000 Julien Isorce <julien.isorce@collabora.co.uk>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: ensure having caps when setting the buffer pool config
+ It happens if downstream does not propose a buffer pool.
+ GST_DEBUG=2 gst-launch-1.0 videotestsrc ! fakesink
+ https://bugzilla.gnome.org/show_bug.cgi?id=723271
+
+2014-01-30 21:18:04 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: Support non-ASCII tags
+ By calling setlocale() to get us multi-byte/UTF-8 support.
+ https://bugzilla.gnome.org/show_bug.cgi?id=723164
+
+2014-01-28 14:28:27 +0100 Bastien Nocera <hadess@hadess.net>
+
+ * tools/gst-discoverer.c:
+ gst-discoverer: Support non-ASCII tags
+ By calling setlocale() to get us multi-byte/UTF-8 support.
+ https://bugzilla.gnome.org/show_bug.cgi?id=723164
+
+2014-01-30 10:43:48 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From d48bed3 to 1a07da9
+
+2014-01-29 13:58:07 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst/encoding/gststreamsplitter.c:
+ streamsplitter: push pending events before eos
+ Push any pending events downstream before pushing eos
+
+2014-01-29 12:33:21 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/audioencoder.c:
+ tests: audioencoder: add tests analogous to the videoencoder ones
+
+2014-01-29 12:32:16 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: push pending events and tags before EOS
+ if there are tags or events pending and an EOS is received, push those
+ events and tags before the EOS.
+
+2014-01-28 15:25:05 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/libs/videoencoder.c:
+ tests: videoencoder: check that tags are pushed before eos
+ Check that if a new tag event is received right before eos it
+ is pushed before the eos
+
+2014-01-28 15:30:35 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: push tags and events before eos
+ if any tags or events are pending, push them before pushing eos
+
+2014-01-28 15:06:39 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/videoencoder.c:
+ tests: videoencoder: basic videoencoder base class test
+ Adds a single test for video encoding
+
+2013-11-26 01:13:45 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Do cosmetic changes to rtptime calculations
+ * Change running time type to guint64
+ * Use GST_CLOCK_TIME_NONE() to check for invalid timestamps
+ * Name variables so ns-based and hz-based timestamps are evident
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
+
+2014-01-28 00:40:38 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Expose running-time of payloaded stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=719415
+
+2014-01-22 17:47:02 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Improve documentation for perfect-rtptime
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
+
+2014-01-16 16:58:43 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Fix typos in documentation for properties
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
+
+2014-01-28 00:19:07 +1100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ decodebin: make it possible to register multiple handlers for autoplug-select
+ Change the way autoplug-select is accumulated so that it's possible to have
+ multiple handlers. The handlers keep getting called as long as they keep
+ returning GST_AUTOPLUG_SELECT_TRY.
+ One practical example of when this is needed is when hooking into playbin's
+ uridecodebin, which is perhaps not very elegant but the only way to influence
+ which streams playbin autoplugs/exposes.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723096
+
+2014-01-16 21:49:59 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ * tests/check/libs/rtp-basepayloading.c:
+ rtpbasepayload: Add statistics property
+ This property allows for an atomically retrieved set of properties that
+ can e.g. be used to generate RTP-Info headers.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719415
+
+2013-07-26 15:44:28 +0200 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Drop hardcoded list of media suitable for download buffering
+ Discussion on IRC indicated that the main reason for this list was to
+ prevent demuxers that can trigger a lot of seeking from using
+ progressive buffering using queue2 (which due to being seekable triggers
+ that behaviour).
+ However given that upstream can indicate seeks are possible but should
+ be avoided via a scheduling query, this extra whitelisting shouldn't be
+ necessary for well-behaved demuxers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=704933
+
+2014-01-24 12:19:43 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoconvert/gstvideoconvert.c:
+ videoconvert: tweak the scoring algorithm
+ Make a little table of conversions and manually score them. Use this
+ info to define better weights for the scoring algorithm.
+ give separate scores for doing changes and the impact of the change,
+ This allows us to avoid conversion when we can but still allow fairly
+ lossless changes.
+ The old code did not penalize GRAY conversions, PAL conversions were
+ punished too low and depth conversions too high.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722656
+
+2014-01-23 10:45:00 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-chroma.c:
+ video-chroma: don't crash on NULL resamplers
+ Make dummy resamplers for all cases and only execute the horizontal
+ resampler instead of crashing.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=722742
+
+2014-01-21 11:21:56 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: make _get_time more threadsafe
+ We call the _get_time function from the provided clock and we don't lock
+ the sink object for performance reasons. Make sure we only read and
+ check variables once so that they don't change while we are executing
+ the code.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720661
+
+2014-01-20 16:11:04 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioresample/resample.c:
+ audioresample: It's HAVE_EMMINTRIN_H, not HAVE_XMMINTRIN_H for SSE2
+
+2014-01-20 15:44:09 +0100 Antoine Jacoutot <ajacoutot@gnome.org>
+
+ * gst/audioresample/resample.c:
+ audioresample: Fix build on x86 if emmintrin.h is available but can't be used
+ On i386, EMMINTRIN is defined but not usable without SSE so check for
+ __SSE__ and __SSE2__ as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=670690
+
+2014-01-20 10:30:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Initialize Qt variables
+
+2014-01-20 09:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ * tests/examples/overlay/Makefile.am:
+ * tests/examples/overlay/qt-videooverlay.cpp:
+ examples: Port Qt examples to Qt5
+
+2014-01-18 19:22:12 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Fix G726 caps creation
+ https://bugzilla.gnome.org/show_bug.cgi?id=720995
+
+2014-01-18 00:18:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: minor docs fix
+ Can use a custom main context as well if needed.
+
+2014-01-18 13:54:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * win32/common/libgstvideo.def:
+ videodecoder: Add API to get the currently pending frame size for parsing
+ https://bugzilla.gnome.org/show_bug.cgi?id=719890
+
+2014-01-18 21:20:51 +0900 Wonchul Lee <chul0812@gmail.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Remove unnecessary assignment
+ Remove duplicated assignment
+ https://bugzilla.gnome.org/show_bug.cgi?id=722491
+
+2014-01-18 13:31:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Insert decoders without GstAVElement information between the other decoders
+ Otherwise they would be preferred over all decoders independent
+ of their ranks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=722316
+
+2014-01-18 13:12:16 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Only put parsers and sinks first, not all non-decoders
+ https://bugzilla.gnome.org/show_bug.cgi?id=722316
+
+2014-01-17 11:08:32 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/libs/videodecoder.c:
+ tests: videodecoder: plug a few leaks
+ Remove leaks of caps and events references
+
+2014-01-17 10:17:29 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: plug leak when frames are released on subclass stop
+ They end up stored in the 'pending_events' list and should be
+ freed after calling stop
+
+2014-01-17 15:10:42 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tools/gst-play.c:
+ gst-play: Handle CLOCK_LOST message
+ It is necessary for playbin gapless playback when switching
+ between audio-only and video-only files for example.
+
+2014-01-16 16:32:34 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/encoding/gststreamsplitter.c:
+ streamsplitter: handle ACCEPT_CAPS query correctly
+ We can accept a caps when one of the downstream peers can accept the
+ caps. This is not the same as checking a subset of the getcaps
+ result because parsers might accept broader caps than what their getcaps
+ function returns (See https://bugzilla.gnome.org/show_bug.cgi?id=677401).
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722330
+
+2014-01-14 13:02:28 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: add another test for negotiation with gap event
+ Check that even if the subclass doesn't call set_output_format, the base
+ class should use upstream provided caps to fill the output caps that is
+ pushed before the gap event is forwarded, otherwise it ends again fixating
+ the rate and channels to 1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=722144
+
+2014-01-14 13:05:54 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: copy rate and channels from input before fixating output caps
+ For default caps generation when handling gap events that are sent
+ before any buffer, try to use caps that are closer to what upstream
+ provided to avoid fixating rate or channels to 1 as default.
+ So there are the steps:
+ 1) Try to set rate, channels and channel-mask from upstream if provided
+ 2) Fixate the rate and channels to the default rate and channels from
+ audio lib
+ 3) Fixate the caps just to be sure everything is fixed
+ 4) If no channel-mask was provided and channels > 2, use a default
+ channel-mask (taken from audioconvert code)
+ https://bugzilla.gnome.org/show_bug.cgi?id=722144
+
+2014-01-14 23:07:34 +0100 Holger Kaelberer <hk@getslash.de>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: don't recreate xvcontext
+ A xvcontext can be created early in gst_xvimagesink_set_window_handle().
+ In this case don't recreate, i.e. overwrite it in gst_xvimagesink_open().
+ Otherwise XEvents won't be handled in the xevent listener thread.
+ Fixes a regression when setting the window handle on the sink in
+ the very beginning before changing its state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=715138
+
+2014-01-14 12:05:46 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix broken seeking reading the whole file
+ A change in gst_ogg_demux_do_seek caused oggdemux to wait for
+ a page for each of the streams, including a skeleton stream if
+ one was present. Since Skeleton only has header pages, that
+ was never going to end well.
+ Also, the code was skipping CMML streams when looking for pages,
+ so would also have broken on CMML streams.
+ Thus, we change the code to disregard Skeleton streams, as well
+ as discontinuous streams (such as CMML and Kate). While it may
+ be desirable to consider Kate streams too (in order to avoid
+ losing a subtitle starting near the seek point), this may be
+ a performance drag when seeking where no subtitles are. Maybe
+ one could add a "give up" threshold for such discontinuous
+ streams, so we'd get any page if there is one, but do not end
+ up reading preposterous amounts of data otherwise.
+ In any case, it is important that the code that determines
+ the amount of streams to look pages for remains consistent with
+ the "early out" conditions of the code that actually parses
+ the incoming pages, lest we never decrease the pending counter
+ to zero.
+ This fixes seeking on a file with a skeleton track reading all
+ the file on each seek.
+ https://bugzilla.gnome.org/show_bug.cgi?id=719615
+
+2014-01-13 15:14:14 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: use an adaptive chunksize for performance reasons
+ Ogg data is read chunk by chunk, and the chunk size used was
+ originally taken from libvorbisfile. However, this value leads
+ to poor performance when used on an Ogg file with large pages
+ (Ogg pages can be close to 64 KB).
+ We can't just use a larger chunk size, since this will decrease
+ performance on small page streams, so we use an adaptive scheme
+ where the chunk size is twice the largest page size we've seen
+ so far in the stream. For "typical" Ogg/Vorbis, this gives us
+ almost the same chunk size (a bit lower), and this lets us get
+ better performance on streams with large pages.
+
+2014-01-13 20:47:02 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: avoid parsing caps event if it is not used
+ Saves some cpu
+
+2014-01-13 20:44:23 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: make sure caps is set before forwarding gap event
+ Before trying to generate a default fixated caps when handling a gap
+ event, make sure that the same strategy that is used when handling
+ a buffer has been attempted. Otherwise audiodecoder will ignore
+ upstream caps settings such as rate and channels and will likely
+ end with a caps with channels=1 and rate=1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=722144
+
+2014-01-13 19:40:49 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: check that negotiation works buffers and gaps
+ Adds 2 tests to verify that output caps are the expected value, reusing
+ input structure values for both buffers and gaps
+ https://bugzilla.gnome.org/show_bug.cgi?id=722144
+
+2014-01-13 16:33:11 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/audiodecoder.c:
+ tests: audiodecoder: add basic playback test for audio decoder
+ Simple test that just check that audio decoding works as expected
+ https://bugzilla.gnome.org/show_bug.cgi?id=722144
+
+2014-01-14 13:17:26 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/videooverlay.c:
+ videoverlay: Don't mention gconf elements and add a sentence about playbin/playsink
+ playbin/playsink now implement the video overlay interface
+
+2014-01-13 16:28:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * win32/common/libgstvideo.def:
+ win32: add new API to .def file
+
+2014-01-13 16:29:00 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: only copy chroma_site when known
+ Only overwrite the chroma-site if we have a valid value in the reference
+ format.
+
+2014-01-13 16:20:55 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoconvert/gstvideoconvertorc.orc:
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: don't interpolate chroma in I420 -> RGB
+ Don't try to interpolate the chroma samples, the used algorithm only
+ works for horizontal cositing. Let's switch to a faster and safer
+ version until we handle chroma siting correctly in the fastpaths.
+
+2014-01-13 12:16:01 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/gstvideoutils.c:
+ videoutils: add some debug
+
+2014-01-08 19:43:01 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ doc: Add new sections introduce for tile format
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2014-01-08 19:42:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ video: Generate types for tile enumeration
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2014-01-08 19:41:56 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/design/part-mediatype-video-raw.txt:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-frame.c:
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-tile.h:
+ video: Don't use extra plane and componenent for tile format
+ Instead of using extra plane, we encode the number of tiles in x and y in the stride of
+ each planes (i.e. y_tiles << 16 | x_tiles) and introduce tile_mode, tile_width and
+ tile_height into GstVideoFormatInfo structure.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2014-01-03 22:36:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/design/part-mediatype-video-raw.txt:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ * tests/check/elements/videoscale.c:
+ video: rename NV12T -> NV12_64Z32
+ Is a bit more descriptive and allows us to add more tiled types
+ later.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2014-01-03 22:29:09 +0100 Nicolas Dufresne <nicolas.dufresne at collabora.co.uk>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: scale vertical tiles based on subsampling
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2014-01-03 22:18:08 +0100 Nicolas Dufresne <nicolas.dufresne at collabora.co.uk>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: fix tiled pixel stride
+ Pixel stride is per component, not per plane. We get the tile mode from
+ the pixelstride of the TILE component.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-26 17:40:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.h:
+ format: improve docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-25 16:22:32 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/elements/videoscale.c:
+ tests: fix videoscale test for NV12T
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-25 16:06:43 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-frame.c:
+ video-format: fix off-by-one for tiled coordinates
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-25 15:22:24 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-tile.h:
+ video-tile: improve docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-25 14:57:30 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: use shifts when possible
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-25 14:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: fix copy of tiled formats
+ Add code to copy tiled planes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-25 14:11:57 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-tile.c:
+ * gst-libs/gst/video/video-tile.h:
+ video-tile: add tile mode and helper functions
+ Move the tile helper functions to their own file. Make it possible to
+ make other tiling modes later.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-20 21:27:46 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/design/part-mediatype-video-raw.txt:
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ video: add NV12T support
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2013-12-19 16:11:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-format.h:
+ Add tiled color format support
+ https://bugzilla.gnome.org/show_bug.cgi?id=707361
+
+2014-01-13 15:32:23 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Fix typo in the docs
+
+2014-01-11 01:14:19 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/libs/videodecoder.c:
+ tests: videodecoder: check that segment events are not dropped
+ Adds a test that simulates a scenario where the first buffers after
+ a segment can't be decoded and the decoder asks for those frames
+ to be released. The videodecoder base class should make sure that
+ the events attached to those first buffers are pushed even if the
+ buffers aren't going to be.
+ https://bugzilla.gnome.org/show_bug.cgi?id=721835
+
+2014-01-11 01:24:44 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: do not lose events when dropping frames
+ Events must be persisted after a frame is dropped to avoid
+ losing obligatory information for the stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=721835
+
+2014-01-08 11:29:29 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/libs/videodecoder.c:
+ tests: videodecoder: add test for reverse playback
+ Checks that buffers are pushed backwards in reverse playback
+ https://bugzilla.gnome.org/show_bug.cgi?id=721666
+
+2014-01-06 20:53:15 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: use new segment earlier for reverse playback
+ For reverse playback, the segment event will only be pushed when
+ the first buffer is actually pushed. But for decoding frames and storing
+ those into the list to be pushed the output_segment.rate value is used
+ to determine if it is forward or reverse playback.
+ In case a previous segment event (or none) is in use it will mistakenly
+ think it is doing forward playback and push the buffers immediatelly and
+ try to clip buffers based on an old segment (or an uninitialized one, leading
+ to an assertion)
+ This patch fixes this by copying the segment earlier if on reverse playback
+ https://bugzilla.gnome.org/show_bug.cgi?id=721666
+
+2014-01-10 14:24:12 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: fix unit test breaking on duration query
+ The new switch caused breaks to not break of the main switch
+ anymore, causing fall through.
+
+2014-01-10 15:06:23 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videoconvert/gstvideoconvertorc-dist.c:
+ * gst/videoconvert/gstvideoconvertorc-dist.h:
+ videoconvert: Update disted orc files once again
+
+2014-01-10 11:17:38 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: add dot file dumping for pipeline graph debugging
+
+2014-01-10 11:17:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pango/gstbasetextoverlay.c:
+ textoverlay: don't leak GAP events
+
+2014-01-10 09:53:21 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: do not set TIME duration when asked for another format
+ This fixes asserts in pipelines such as:
+ gst-launch-1.0 videotestsrc num-buffers=1000 ! x264enc ! h264parse ! \
+ matroskamux name=mux ! filesink location=test.mkv
+
+2014-01-10 09:21:08 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videoconvert/gstvideoconvertorc-dist.c:
+ * gst/videoconvert/gstvideoconvertorc-dist.h:
+ videoconvert: Update disted orc files
+
+2014-01-09 18:12:00 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoconvert/gstvideoconvertorc.orc:
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: rework YUV->RGB fastpaths
+ Rework the orc code to be around 10% faster and support arbitrary matrices.
+ Pass the matrix parameters to the YUV->RGB functions to make them work
+ for all matrices. This enables more and faster fastpath conversions.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=721701
+
+2014-01-09 18:08:41 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoconvert/gstvideoconvertorc.orc:
+ videoconvert: fix I420 to BGRA fast-path some more
+ Calculate alpha value differently so that we can avoid running out
+ of registers.
+
+2014-01-08 16:20:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videoconvert/gstvideoconvertorc.orc:
+ videoconvert: remove unused code
+
+2014-01-03 15:24:29 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add G726 ADPCM support
+ https://bugzilla.gnome.org/show_bug.cgi?id=720995
+
+2014-01-07 22:04:20 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/libs/videodecoder.c:
+ tests: videodecoder: add check for serialization of events
+ Tests that events are properly serialized with buffers, also checks
+ that the usual events are sent (stream start, caps, segment and eos).
+
+2014-01-07 16:28:18 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/videodecoder.c:
+ tests: videodecoder: add simple playback test
+ Add a simple playback test that makes sure that video decoder pushes
+ buffers in the same order it receives and that it respects the
+ set timestamps and durations
+
+2014-01-07 15:01:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * win32/common/libgstrtsp.def:
+ defs: update for new symbols
+
+2014-01-07 14:46:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ rtsptransport: calculate default lower transport
+ Add an internal method to calculate the default lower transport whan it
+ is missing.
+
+2014-01-07 14:31:09 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsptransport: add method to get media-type from transport
+ Add a method to make a media-type from the transport. Deprecate the old
+ method that only used the mode.
+ Based on patch from Aleix Conchillo Flaqué <aleix@oblong.com>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720219
+
+2014-01-07 11:51:01 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsptransport: add GType for Profile
+ See https://bugzilla.gnome.org/show_bug.cgi?id=720696
+
+2014-01-05 23:35:52 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add support of BWF RF64 a 64bit wav variant
+ Detect and describe the RF64 Broadcast Wave Format.
+ Fixes #519220
+
+2014-01-05 21:39:52 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/riff/riff-read.c:
+ * gst-libs/gst/riff/riff-read.h:
+ * win32/common/libgstriff.def:
+ riff: remove new parse_ncdt api again
+ This chunk is avi specific, no need to expose this as public api.
+
+2014-01-04 22:30:17 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * win32/common/libgstriff.def:
+ win32: export new riff api
+
+2014-01-04 21:54:10 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/riff/riff-read.c:
+ riff: fix indentation messup from previous commit
+
+2014-01-04 21:31:07 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-read.c:
+ * gst-libs/gst/riff/riff-read.h:
+ riff: add support for nikon tags
+ Nikon cameras store metadata in a custom format. Add parsing of the chunk and
+ extract some initial data.
+ API: gst_riff_parse_ncdt()
+ Fixes #636143
+
+2014-01-03 02:18:20 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobasesrc: Avoid unnecessary configuration
+ Port a change from audiobasesink from def07410, to ignore setcaps
+ when the caps don't actually change, and avoid a reconfiguration
+ and reset of the ringbuffer in that case.
+
+2013-11-15 14:17:03 +0000 William Grant <wgrant@ubuntu.com>
+
+ * configure.ac:
+ configure: Prevent the NEON check in configure from passing under aarch64.
+ The test verifies that the NEON C intrinsics work, but the rest of the
+ codebase uses lots of direct ARMv7 NEON assembly. The same intrinsics
+ work in A64, but the assembly is slightly different.
+ Prevent the check from passing so that we don't use this where it won't
+ work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=712367
+
+2013-12-31 10:17:55 +0100 Stéphane Cerveau <scerveau@gmail.com>
+
+ * gst-libs/gst/riff/riff-ids.h:
+ riff: Add id3 tag
+ Add id3 tag for wavparse
+ https://bugzilla.gnome.org/show_bug.cgi?id=721241
+
+2013-12-31 09:37:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/icles/test-effect-switch.c:
+ Revert "test-effect-switch: Change one of the pad blocks to and idle probe"
+ This reverts commit 40fe5dcc84ff2cc7dbe0112d7830a33fd764d4e1.
+ Using an idle probe here is not ideal because we'll send an EOS event
+ from the application thread... which might block for quite some time.
+ Go back to a block probe.
+
+2013-12-30 19:48:29 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Always set pixel-aspect-ratio and interlace-mode in the fixed caps
+ Otherwise our caps will not be compatible with elements that require a
+ 1/1 pixel-aspect-ratio or progressive video.
+ https://bugzilla.gnome.org/show_bug.cgi?id=721103
+
+2013-12-30 19:40:29 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/icles/test-effect-switch.c:
+ test-effect-switch: Don't put two format fields into the first capsfilter
+
+2013-12-30 19:12:53 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/icles/test-effect-switch.c:
+ test-effect-switch: Change one of the pad blocks to and idle probe
+ Just because we can.
+
+2013-12-30 17:30:15 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Add missing break statement
+ And do a minor cleanup
+ COVERITY CID 1139753
+
+2013-12-30 14:30:23 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/riff/riff-ids.h:
+ riff: add two chunk-ids for samples instruments
+ Wav files can have 'smpl' and 'inst' chunks.
+
+2013-12-30 13:46:34 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: Fix array read
+ nbchannels ranges from 1 to 8, therefore use '- 1' to get the proper
+ array value.
+
+2013-12-30 13:33:00 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Remove useless assignement
+ Was already set before
+
+2013-12-26 17:47:46 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ gstrtpbasepayload: use the session's suggested ssrc after a collision, if the session provides one
+ Conflicts:
+ gst-libs/gst/rtp/gstrtpbasepayload.c
+
+2013-12-10 15:19:14 +0000 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstrawcaps.h:
+ playback: add ANY caps features to default audio/video raw caps
+ Allows elements using audio/video caps features to be used by playbin.
+
+2013-12-30 10:53:24 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-info.c:
+ * gst-libs/gst/video/video-info.c:
+ audio/video-info: Properly initialize the info structures in set_format()
+ And don't assume in other code that set_format() preserves any fields at
+ all. These assumptions were already made here for fields that were changed
+ by set_format().
+
+2013-12-30 10:14:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-info.c:
+ * gst-libs/gst/video/video-info.c:
+ audio/video-info: Initialize the complete struct to 0 in the beginning
+ Instead of only initializing some parts in some code paths. Also
+ makes it easier to use the reserved bits of the structs later.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720810
+
+2013-12-27 14:29:46 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: increase max payload size to 4000 bytes
+ 1275 is the maximum size of a frame, but the encoder may return
+ up to 3 frames, and we need a few extra bytes for TOC, etc. We
+ use 4000, which is a bit more, and suggested in the libopus docs.
+
+2013-12-20 19:48:06 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobasesrc: Bunch of cosmetic/grammar fixes
+
+2013-12-20 18:58:43 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobasesrc: Retarget FIXME to 2.0
+ Properly fixing this one would break API.
+
+2013-12-20 18:54:39 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ * gst-libs/gst/audio/gstaudiocdsrc.c:
+ * gst-libs/gst/audio/gstaudiodecoder.h:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiobase*: Drop trailing withespaces
+
+2013-12-20 18:53:13 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobasesrc: Break some too long lines
+
+2013-12-20 18:41:59 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobasesrc: Add FIXME for times in NSECONDS
+ Timebase is in nanoseconds pretty much everywhere else
+
+2013-12-26 23:21:45 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Choose a default initial caps before sending GAP
+ If there are no caps from the audio decoder when handling a GAP
+ event - as when one is received right at the start on a DVD without
+ initial audio - then choose any default caps for downstream and
+ then send the GAP, so the audio sink has a configured format in
+ which to start the ringbuffer.
+ Also, make the audio sink reject a GAP without caps with a clearer
+ error message.
+ Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=603921
+
+2013-12-26 17:41:00 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsptransport: add more profiles
+ Add support for Feedback profiles
+
+2013-12-25 10:45:11 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: fix plane copy for index plane
+ Move the code to handle the index plane in the _copy_plane.
+
+2013-12-24 01:20:25 +0000 Lionel Landwerlin <llandwerlin@gmail.com>
+
+ * gst-libs/gst/video/colorbalance.c:
+ colorbalance: add missing annotation for list_channels()
+ https://bugzilla.gnome.org/show_bug.cgi?id=720999
+
+2013-12-23 14:54:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videoconvert/gstvideoconvertorc.orc:
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: Fix I420 to BGRA fast-path alpha setting
+ This fast-path was adding 128 to every component including
+ alpha while it should only be done for all components except
+ alpha. This caused wrong alpha values to be generated.
+ Also remove the high-quality I420 to BGRA fast-path as it needs
+ the same fix, which causes an additional instruction, which causes
+ orc to emit more than 96 variables, which then just crashes.
+ This can only be fixed in orc by breaking ABI and allowing more
+ variables.
+
+2013-12-22 22:33:26 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From dbedaa0 to d48bed3
+
+2013-12-22 21:56:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * po/Makevars:
+ po: set gettext domain in Makevars so we don't have to patch the generated Makefile.in.in
+ https://bugzilla.gnome.org/show_bug.cgi?id=705455
+
+2013-12-22 22:07:43 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/.gitignore:
+ tests: make git ignore new test binary
+
+2013-12-20 18:06:25 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Always reset last_align
+ Should be done for all the reset_sync() cases. Not
+ only for the READY to PAUSED one.
+
+2013-12-20 18:02:42 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Reset last_align to 0, not -1
+ This is the expected behavior in READY -> PAUSED
+
+2013-12-20 17:58:43 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Always reset avg_skew on _reset
+ Only case in which it wasn't (READY to PAUSED) should
+ have had this value reseted too.
+
+2013-12-20 17:10:44 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Retarget FIXME to 2.0
+ Properly fixing this one would break API
+
+2013-12-20 15:13:54 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Factor out reset sync routine
+
+2013-12-20 01:06:33 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Drop dead _sink_async_play() code
+
+2013-12-20 01:03:14 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Break some too long lines
+
+2013-12-20 00:09:22 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Cosmetics, grammar/spelling
+ - Drop repeated 'yet' from debug msg
+ - Drop repeated 'to' from param desc
+ - Some spelling
+
+2013-12-20 08:41:45 -0500 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/audio/audio-info.c:
+ * gst-libs/gst/video/video-info.c:
+ audio/video: Initialize all {audio|video}info fields
+ Fixes "Unitialized Scalar Variable" issues reported by Coverity.
+ Has the added advantage of detecting whether somebody *does* use those
+ fields (ending up with a invalid address).
+ https://bugzilla.gnome.org/show_bug.cgi?id=720810
+
+2013-12-19 17:41:31 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ gstaudiobasesink: Refactor alignment computation for clarity
+
+2013-12-18 15:52:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/subparse.c:
+ subparse: Add unit test for LRC subtitles
+
+2013-12-18 15:24:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: Add support for parsing LRC subtitles
+ https://bugzilla.gnome.org/show_bug.cgi?id=678590
+
+2013-12-18 15:07:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ subparse: Add typefinder for LRC subtitles
+
+2013-12-10 13:54:28 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ sdp: parse encryption key field
+ * gst-libs/gst/sdp/gstsdpmessage.c: parse encryption key field (k).
+ https://bugzilla.gnome.org/show_bug.cgi?id=720215
+
+2013-12-17 18:04:33 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ * tests/check/libs/pbutils.c:
+ pbutils: add typefinder and descriptions for audio/x-xi
+ xi files can be read by libsndfile.
+
+2013-12-17 18:03:40 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ descriptions: longer version of two audio codec descriptions
+
+2013-12-17 17:25:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-format.h:
+ video-format: Document usage of GST_VIDEO_FORMAT_ENCODED
+ This must only ever be used in caps in combination with a non-system
+ memory GstCapsFeatures, and where it does not make sense to specify
+ any of the other video formats. Examples of this would be in gst-vaapi.
+
+2013-12-17 17:23:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ Revert "video: specify/restrict usage of GST_VIDEO_FORMAT_ENCODED"
+ This reverts commit 5fcdabd907ca45595b64131bbae0ea963e259a7c.
+ Instead of making it impossible to use the ENCODED format we should
+ just document that it must not be used for capsfeature-less caps.
+ Also this commit broke API/ABI.
+
+2013-12-17 17:09:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Release the allocator on hard resets
+
+2013-12-16 15:53:41 +0000 Julien Isorce <julien.isorce@collabora.co.uk>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: release buffer pool and allocator on full reset
+ It allows to release the buffer pool sooner (i.e. when going
+ to GST_STATE_READY). Previously it was released in finalize.
+ Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=720389
+
+2013-12-15 21:01:42 -0800 Todd Agulnick <todd@agulnick.com>
+
+ * gst-libs/gst/audio/audio-format.c:
+ * sys/xvimage/xvimagesink.c:
+ Some compiler warning fixes to satisfy XCode compiler
+ https://bugzilla.gnome.org/show_bug.cgi?id=720513
+
+2013-12-16 11:35:12 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Read image-type from the GstSample info struct
+ But for backwards compatibility keep reading it from the caps and only
+ use the info struct if the caps don't contain the image-type.
+
+2013-12-13 14:36:41 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: gst_video_decoder_release_frame() is available since 1.2.2
+
+2013-12-13 10:06:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: play: allow parse-launch strings for audio and video sink
+
+2013-12-12 13:42:59 +0100 Julien Isorce <julien.isorce@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: change SSRC on GstRTPCollision event
+ Change our SSRC and update the caps when we receive a GstRTPCollision
+ event from downstream.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711560
+
+2013-12-12 13:06:30 +0100 Julien Isorce <julien.isorce@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: implement src_event function
+ Add a srcpad event handler and call the src_event vmethod.
+
+2013-12-11 16:49:35 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/video/video-format.h:
+ * gst-libs/gst/video/video-info.c:
+ video: specify/restrict usage of GST_VIDEO_FORMAT_ENCODED
+ GST_VIDEO_FORMAT_ENCODED was added to support *extracting* video-related
+ information (like width, height, framerate,...) from caps.
+ It is __NOT__ intended to be used as a format field on video/x-raw caps.
+
+2013-12-10 00:13:55 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/libs/rtp-basepayloading.c:
+ tests: Add test for rtpbasepayload/-depayload
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720162
+
+2013-12-10 00:56:07 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * tests/check/libs/rtp.c:
+ rtpbuffer: Allow subbuffering of empty buffers
+ See https://bugzilla.gnome.org/show_bug.cgi?id=720162
+
+2013-12-09 16:34:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/convertframe.c:
+ convertframe: Fix indention
+
+2013-12-09 16:33:40 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/gstvideoencoder.h:
+ videoencoder: Add sink_query() src_query() virtual functions
+ Based on the videodecoder change by Nicolas Dufresne and applied
+ here for consistency.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720103
+
+2013-11-27 16:39:52 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ videodecoder: Add sink_query() src_query() virtual
+ https://bugzilla.gnome.org/show_bug.cgi?id=720103
+
+2013-12-09 13:55:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play-kb.c:
+ tools: play: fix compiler warning on windows
+
+2013-12-06 19:27:04 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst-libs/gst/video/gstvideoutils.h:
+ videocodecframe: Correct function name in doc
+
+2013-12-06 16:23:46 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideoencoder.h:
+ videoencoder: Remove gst_video_encoder_set/get_discont
+ They've never existed outside the header file.
+
+2013-12-04 01:08:13 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * docs/design/Makefile.am:
+ docs: add missing files for distribution
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720015
+
+2013-12-05 16:17:22 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: handle the RESYNC flag
+ Also resync when a buffer with the RESYNC flag is seen.
+
+2013-12-05 14:39:57 +0000 Julien Isorce <julien.isorce@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audiodec/enc: clear reconfigure flag if negotiate succeeds
+ So that it avoids to send an allocation query twice.
+ One from an early call to gst_audio_encoder_negotiate from a
+ subclass, then one from gst_audio_encoder_allocate_output_buffer.
+ Which means that previously gst_audio_encoder_negotiate was not
+ clearing the GST_PAD_FLAG_NEED_RECONFIGURE even on success.
+ Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=719684
+
+2013-12-05 14:31:25 +0000 Julien Isorce <julien.isorce@collabora.co.uk>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videodec/enc: clear reconfigure flag if negotiate succeeds
+ So that it avoids to send an allocation query twice.
+ One from an early call to gst_video_encoder_negotiate from a
+ subclass, then one from gst_video_encoder_allocate_output_frame.
+ Which means that previously gst_video_encoder_negotiate was not
+ clearing the GST_PAD_FLAG_NEED_RECONFIGURE even on success.
+ Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=719684
+
+2013-12-05 12:04:59 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: Require caps to be set before any data processing
+
+2013-12-05 11:39:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/theora/gsttheoradec.c:
+ theoradec: Use new gst_video_decoder_set_needs_format() API
+
+2013-12-05 11:37:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Use FALSE instead of 0
+
+2013-12-05 11:34:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * win32/common/libgstvideo.def:
+ videodecoder: Add API to allow subclasses to specify that they needs caps before any buffers
+
+2013-12-05 11:25:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Return not-negotiated if we don't have caps when the first buffer arrives
+ Otherwise things like filesrc ! jpegenc ! fakesink just crash with
+ a segmentation fault because subclasses expect caps to be there.
+
+2013-12-04 19:24:08 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: no fallback to segment start for reverse playback
+ See https://bugzilla.gnome.org/show_bug.cgi?id=709965
+
+2013-12-05 00:27:14 +0900 Justin Joy <justin.joy.9to5@gmail.com>
+
+ * gst-libs/gst/video/convertframe.c:
+ convertframe: Fix trivial memory leak in debug statement
+ gst_element_get_name() requires the caller to g_free() the return value
+ https://bugzilla.gnome.org/show_bug.cgi?id=719850
+
+2013-12-02 20:35:04 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: use segment start as fallback ts if no other available
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709965
+
+2013-12-01 12:37:52 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * win32/common/libgstvideo.def:
+ videodecoder: add new API to docs and defs
+
+2013-11-26 20:50:33 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ videodecoder: make _release_frame external API
+ ... so subclasses can release a frame all the way (also from frame list)
+ without having to pass through _finish_frame or _drop_frame.
+ The latter may not be applicable, or may or may not have already
+ been called for the frame in question.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=693772
+
+2013-11-26 20:51:58 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: fix spelling error in debug message
+
+2013-11-29 17:30:09 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: copy sticky events
+
+2013-11-29 17:26:13 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: copy sticky events
+
+2013-11-29 13:32:55 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/theora/gsttheoraparse.c:
+ theoraparse: Fix event handling
+ Send CAPS event before any SEGMENT events or any other events
+ that must come in order after the CAPS event.
+
+2013-11-29 09:04:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: gst-play: quit on Q or Esc key
+
+2013-11-28 16:22:01 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/tcp/gsttcpserversink.c:
+ tcp: fix compilation with MSVC
+ error C2440 at line 165 of gsttcpserversink.c
+ type cast error: cannot convert from GSocket* to GstMultiSinkHandle
+
+2013-11-28 11:25:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: activate ghost pad before targetting
+ Activate the decodebin2 pad before setting the target. This makes sure
+ that the events are copied.
+
+2013-11-21 22:54:42 +1100 Matthew Waters <ystreet00@gmail.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/gstvideometa.h:
+ videometa: add GstVideoGLTextureUploadMeta buffer pool option
+ allows configuration of whether GstVideoGLTextureUploadMeta is
+ added to buffers resulting from a buffer pool. This is sperate
+ to the caps feature in that an element may want to add the upload
+ meta itself rather than allowing the buffer pool to.
+ https://bugzilla.gnome.org/show_bug.cgi?id=712798
+
+2013-11-26 12:29:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: error out if no frames are decoded before eos
+ Raise an error in case no frames are decoded before EOS and we
+ have input, meaning that data was received but it was somehow invalid.
+ Based on the videodecoder change, merged here for consistency.
+ https://bugzilla.gnome.org/show_bug.cgi?id=711094
+
+2013-11-26 12:20:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Allow using -1 for infinite tolerated errors
+ Allows using -1 to make audiodecoder never post an error message
+ after decoding errors.
+ Based on the videodecoder change, merged here for consistency.
+ https://bugzilla.gnome.org/show_bug.cgi?id=711094
+
+2013-11-26 12:03:24 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Fix visualizations if no visualization plugin was set
+ https://bugzilla.gnome.org/show_bug.cgi?id=712280
+
+2013-10-29 14:40:23 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: error out if no frames are decoded before eos
+ Raise an error in case no frames are decoded before EOS and we
+ have input, meaning that data was received but it was somehow invalid.
+ https://bugzilla.gnome.org/show_bug.cgi?id=711094
+
+2013-10-29 14:11:51 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: allow using -1 for infinite tolerated errors
+ Allows using -1 to make videodecoder never post an error message
+ after decoding errors.
+ https://bugzilla.gnome.org/show_bug.cgi?id=711094
+
+2013-11-24 14:38:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play-kb.h:
+ * tools/gst-play.c:
+ tools: play: implement seeking via console in interactive mode
+ Arrow left and right to seek back of forward.
+
+2013-11-24 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: play: fix endless loop on unhandled keys
+ When debugging output is not enabled.
+
+2013-11-24 13:49:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: play: add keyboard controls for next/previous item in list
+ Make the '>' and '<' keys skip to the next or previous item in
+ the playlist.
+
+2013-11-24 01:08:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/Makefile.am:
+ * tools/gst-play-kb.c:
+ * tools/gst-play-kb.h:
+ * tools/gst-play.c:
+ tools: play: add --interactive switch and basic keyboard handling
+ Only pause/play with spacebar for now.
+
+2013-11-23 11:25:28 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add typefinder for OpenEXR
+
+2013-11-21 21:33:59 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: avoid descending output timestamps
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712796
+
+2013-11-22 21:00:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tools/gst-play.c:
+ tools: play: add --shuffle command line option
+
+2013-11-21 16:34:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/subparse.c:
+ tests: add unit test for samiparser issue
+ https://bugzilla.gnome.org/show_bug.cgi?id=712805
+
+2013-11-21 22:04:46 +0900 Jihyun Cho <jihyun.jo@gmail.com>
+
+ * gst/subparse/samiparse.c:
+ subparse: fix null pointer access in sami parser
+ https://bugzilla.gnome.org/show_bug.cgi?id=712805
+
+2013-11-21 15:19:47 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/subparse/gstssaparse.c:
+ * gst/subparse/gstsubparse.c:
+ subparse: g_memmove() is deprecated
+ Just use plain memmove(), g_memmove() is deprecated in
+ recent GLib versions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=712811
+
+2013-11-18 19:27:14 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/input-selector-test.c:
+ tests: fix input-selector-test
+ Update for pad template name changes.
+
+2013-11-18 16:03:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/appsrc.c:
+ tests: fix appsrc test with latest GLib version
+ With the latest GLib, g_source_remove() complains about not finding
+ the timeout source with the given ID here, since it was already
+ destroyed by returning FALSE from the timeout callback. Also return
+ FALSE from the bus watches when we don't want to be called any more.
+
+2013-11-16 13:06:37 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/pango/gstbasetextoverlay.c:
+ * ext/theora/gsttheoraparse.c:
+ * gst/app/gstapp.c:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/gio/gstgiosink.c:
+ * gst/gio/gstgiosrc.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultihandlesink.c:
+ * gst/tcp/gstmultioutputsink.c:
+ * gst/tcp/gstmultisocketsink.c:
+ * gst/videorate/gstvideorate.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ docs: remove old 0.10 Since markers
+ They're just confusing.
+
+2013-11-16 12:29:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ * gst-libs/gst/rtsp/gstrtsprange.h:
+ docs: cosmetic since marker fixes
+
+2013-11-16 15:24:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: also set output buffer DTS
+
+2013-11-14 01:53:31 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Fix identification of some MPEG files
+ Make sure we begin by peeking at MPEG2_MAX_PROBE_LENGTH
+ bytes.
+ Fixes:
+ https://bugzilla.gnome.org/show_bug.cgi?id=678011
+
+2013-11-13 20:12:48 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: Fix gst_rtp_buffer_ext_timestamp() with clang 5 on iOS/ARM
+ The bitwise NOT operator is not defined on signed integers.
+ Thanks to Wim Taymans for finding the cause.
+ https://bugzilla.gnome.org/show_bug.cgi?id=711819
+
+2013-11-12 18:58:43 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/streamsynchronizer.c:
+ tests: fix race in streamsynchronizer test
+ Wait for thread to exit before starting to free the
+ to_push list, otherwise thread might check the final
+ to_push->next node only after we've freed it already.
+
+2013-11-11 14:10:53 +0200 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: try to negotiate the buffer pool even though there is no o/p format
+ We could have allocation query before caps event and even without caps inside
+ the query. In such cases , the downstream can return a bufferpool object with
+ out actually configuring it. This feature is helpful to negotiate the bufferpool
+ with out knowing the output video format. For eg: some hardware accelerated
+ decoders can interpret the o/p video format only after it finishes the decoding
+ of one buffer at least.
+ https://bugzilla.gnome.org/show_bug.cgi?id=687183
+
+2013-11-07 15:03:34 +0000 Tom Greenwood <tcdgreenwood@hotmail.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Fix deadlock that may occur when multiple threads access appsrc at once
+ https://bugzilla.gnome.org/show_bug.cgi?id=711550
+
+2013-11-04 09:55:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: accumulate buffers in adapter
+ Accumulate buffers in an adapter instead of appending them because append causes
+ a lot of memcpys.
+ Keep track of the last tagsize and accumulate enough data before attempting to
+ parse more data.
+ This patch implements a minimal amount of changes in order to not change the
+ behaviour. We should really rewrite the tag handling and trimming using
+ the adapter API instead of merging and trimming into a buffer.
+
+2013-11-06 12:16:31 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/adder.c:
+ adder: Free consistency checker instance in test_live_seeking test
+
+2013-11-06 12:01:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/adder.c:
+ adder: Release some request pads properly in the unit test
+
+2013-11-05 11:18:01 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 865aa20 to dbedaa0
+
+2013-11-04 11:34:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * tools/gst-discoverer.c:
+ discoverer: fix build after last commit
+ Add a forward declaration for my_g_string_append_printf that specifies
+ G_GNUC_PRINTF. Turn off indent on it as it drives gst-indent crazy.
+
+2013-11-04 11:17:30 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * tools/gst-discoverer.c:
+ discoverer: fix -Wformat-nonliteral warning
+
+2013-11-03 15:57:54 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/libs/audio.c:
+ audio: Add unit test for filling memory with silence samples
+
+2013-11-03 12:23:12 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/gstaudiopack-dist.c:
+ * gst-libs/gst/audio/gstaudiopack-dist.h:
+ audio: Update ORC dist files
+
+2013-11-03 12:22:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-format.c:
+ * gst-libs/gst/audio/gstaudiopack.orc:
+ audio-format: Use ORC for filling memory with silence samples
+
+2013-11-01 17:02:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * win32/common/libgstrtsp.def:
+ rtspconnection: Add new API to the docs and .def file
+
+2013-11-01 16:43:56 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtspconnection: Fix indention in header
+
+2013-11-01 07:25:01 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtspconnection: allow setting tls certificate validation
+ Added new functions gst_rtsp_connection_set_tls_validation_flags() to
+ allow setting the TLS certificate validation flags when establishing a
+ TLS connection.
+ A getter is also available, gst_rtsp_connection_get_tls_validation_flags().
+ https://bugzilla.gnome.org/show_bug.cgi?id=711231
+
+2013-11-01 14:22:13 +0000 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: fix duplicate 'const' declaration warnings
+ https://bugzilla.gnome.org/show_bug.cgi?id=711258
+
+2013-10-16 16:46:05 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/playback/gstrawcaps.h:
+ playback: Add subpicture/x-dvb as raw caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=710325
+
+2013-10-28 12:36:04 +0100 Antonio Ospite <ospite@studenti.unina.it>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: fix adding borders when NV12 is used
+ When the frame buffer is NV12 the borders are not added at all, fix that
+ and fill them to black.
+ https://bugzilla.gnome.org/show_bug.cgi?id=711003
+
+2013-10-23 16:43:32 +0100 Matthieu Bouron <matthieu.bouron@gmail.com>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: remove unneeded guint comparaison
+ https://bugzilla.gnome.org/show_bug.cgi?id=710760
+
+2013-10-14 18:45:16 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: also filter 'framed' field when looking for same streams
+ Fixes extra streams for some mp4 files containing aac audio.
+
+2013-10-08 21:57:11 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix copy'n'paste in comment
+
+2013-10-10 15:56:32 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: Do nothing when flushing the encoder when no caps were set
+ In case we receive a flush event before having our caps set, we will
+ end up trying to create a theora encoder even though we are not ready.
+ Avoid that situation making sure we are initialized before accepting to
+ be flushed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709858
+
+2013-10-11 21:51:00 +0200 Stephan Sundermann <stephansundermann@gmail.com>
+
+ * gst-libs/gst/video/navigation.c:
+ navigation: Add missing out parameter annotations to GstNavigation
+ https://bugzilla.gnome.org/show_bug.cgi?id=709938
+
+2013-10-10 14:09:19 +0100 Julien Isorce <julien.isorce@collabora.co.uk>
+
+ * tests/examples/overlay/qtgv-videooverlay.cpp:
+ examples/overlay: handle the case when xvimagesink is not found
+ So that ximagesink can have a chance to be found.
+ In qtgv-videooverlay.
+
+2013-10-10 14:01:44 +0100 Julien Isorce <julien.isorce@collabora.co.uk>
+
+ * tests/examples/overlay/gtk-videooverlay.c:
+ * tests/examples/overlay/qt-videooverlay.cpp:
+ examples/overlay: unref sink only when found
+ In gtk-videooverlay and qt-videooverlay examples.
+
+2013-10-07 14:52:00 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst/encoding/gstencodebin.c:
+ encodebin: Handle changes in encoding_profile::restriction during playback
+ There are cases where we want to change the restrictions caps during
+ playback, handle that in encodebin.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709588
+
+2013-10-08 17:07:02 +0200 Takashi Iwai <tiwai@suse.de>
+
+ * ext/alsa/gstalsa.c:
+ * ext/alsa/gstalsa.h:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsa: Add channel map API support
+ The initial support for the new ALSA chmap API.
+ Just translate the current chmap to GstAudioChannelPosition during the
+ setup. No function to specify the channel map manually yet, so still
+ impossible to assign any non-standard positions or to configure in a
+ different order even if the hardware allows.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709755
+
+2013-10-08 16:02:46 +0200 Takashi Iwai <tiwai@suse.de>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: Don't clear need_reorder flag too early
+ gst_audio_ring_buffer_set_channel_positions() checks whether the given
+ positions are identical with the current setup and returns
+ immediately if so. But it also clears need_reorder flag before this
+ comparison, thus this flag might be wrongly cleared if the function is
+ called twice with the same channel positions.
+ Move the flag clearance after the check.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709754
+
+2013-10-08 16:13:58 -0300 Thiago Santos <ts.santos@partner.samsung.com>
+
+ * tests/check/elements/videotestsrc.c:
+ videotestsrc: improve test for backwards playback
+ Improve test by checking that timestamps are decreasing
+
+2013-10-08 16:10:54 -0300 Thiago Santos <ts.santos@partner.samsung.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * tests/check/elements/videotestsrc.c:
+ videotestsrc: implement duration query
+ Add duration query to videotestsrc, it can answer this query when
+ the num-buffers property is set.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709646
+
+2013-06-07 16:32:23 -0400 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * tests/check/elements/videotestsrc.c:
+ tests: test videotestsrc in reverse playback
+ https://bugzilla.gnome.org/show_bug.cgi?id=701813
+
+2013-10-08 00:08:34 -0300 Thiago Santos <ts.santos@partner.samsung.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ videotestsrc: implement reverse playback
+ Decrement the n_frames counter when doing reverse playback to
+ have timestamps and offsets reducing instead of increasing
+ https://bugzilla.gnome.org/show_bug.cgi?id=701813
+
+2013-10-08 09:13:50 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: don't overflow in bytes<->time conversion
+ fps_n and _d values can be large and this can overflow a uint. Also fix
+ copy'n'paste mistake in comments.
+
+2013-10-07 22:52:27 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: filter 'parsed' field when checking for same caps
+ We're checking the caps to see if we got more caps details after a parser got
+ plugged. This will also have a flipped 'parsed' field. If the field was already
+ present before the parse the match will fail. Add a function that will do the
+ check while excluding this field.
+
+2013-10-07 22:51:46 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: don't shadow local variables
+
+2013-10-07 22:51:04 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: early return when we have no streams
+
+2013-10-07 22:49:52 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: also log stream-id
+
+2013-10-07 18:53:18 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: fix quark-mismatch for toc and stream-id
+ Seems like a copy'n'paste from 15ee41df.
+
+2013-10-05 21:01:53 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: report depth for video
+ This was returning 0 in all cases. Use the data from GstVideoFormatInfo instead.
+
+2013-10-04 13:57:51 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: Map buffer as READWRITE if the buffer and memory is writable
+ and only use the input buffer as temporary buffer in that case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709408
+
+2013-09-30 21:46:10 +0200 Hans Månsson <hansm@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Connect to proxy if specified
+ Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708880
+
+2013-10-03 19:52:58 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * tools/gst-discoverer.c:
+ discoverer: extract helper to print common stream info
+ Save some lnes of code by using a helper for common stream info.
+
+2013-10-02 11:27:41 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: extract some common code
+ Extract code to make a GstDiscovererInfo. Extracts code that sets StreamInfo.
+
+2013-10-02 15:02:44 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaysink.c:
+ playsink: If the visualisation is changing and reconfiguration is pending, do it all during reconfiguration
+ Otherwise we will have two pad blocks that want to use the same mutex
+ and block each other via the streamlock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=709210
+
+2013-10-02 13:06:03 +0200 Edward Hervey <edward@collabora.com>
+
+ * win32/common/libgstpbutils.def:
+ win32: Update defs file
+
+2013-10-02 12:26:59 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/codec-utils.c:
+ * gst-libs/gst/pbutils/codec-utils.h:
+ * win32/common/libgstpbutils.def:
+ pbutils: Add codec-utility funtions to support H265
+ https://bugzilla.gnome.org/show_bug.cgi?id=708921
+
+2013-10-01 23:17:06 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ descriptions: Add description for H.265
+
+2013-09-24 15:51:46 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add typefind function for H265
+ https://bugzilla.gnome.org/show_bug.cgi?id=708680
+
+2013-09-24 16:47:52 -0700 Thiago Santos <ts.santos@partner.samsung.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: make sure elements are in null before disposing
+ If a pipeline fails to preroll, it might happen that the sinks are
+ put into READY state from playbin's sink activation, but they are never
+ set to playsink, so they aren't being managed by a GstBin and will keep
+ their READY state until they are unreffed, leading to a warning.
+ Prevent this by always forcing them to NULL when deactivating a group
+ https://bugzilla.gnome.org/show_bug.cgi?id=708789
+
+2013-09-28 13:19:02 +0200 Johannes Dewender <gnome@JonnyJD.net>
+
+ * gst-libs/gst/audio/gstaudiocdsrc.c:
+ audiocdsrc: Don't consider trailing data tracks for MusicBrainz disc id calculation
+ MusicBrainz removes trailing data tracks from releases on the server
+ and also for the calculation of the MusicBrainz Disc ID.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708991
+
+2013-09-23 11:35:43 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: check if acquired in set_timestamp
+ Also use GST_OBJECT_LOCK when accessing object data in set_timestamp.
+ https://bugzilla.gnome.org/show_bug.cgi?id=702230
+
+2013-09-15 21:48:43 +0200 MathieuDuponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst/adder/gstadder.c:
+ adder: Don't take channel mask in consideration in mono or stereo
+ This could cause negotiation to fail.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708633
+
+2013-09-27 22:41:28 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: clip buffer before pushing it
+ https://bugzilla.gnome.org/show_bug.cgi?id=708953
+
+2013-09-27 22:40:28 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst-libs/gst/audio/audio.c:
+ audio: change buffer timestamp when clipping even if data hasn't been trimmed
+ https://bugzilla.gnome.org/show_bug.cgi?id=708952
+
+2013-09-27 22:53:43 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add entry for text/x-raw
+ https://bugzilla.gnome.org/show_bug.cgi?id=708954
+
+2013-09-25 19:29:24 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add MPEG 2 AAC description
+ https://bugzilla.gnome.org/show_bug.cgi?id=708773
+
+2013-09-25 15:17:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiobasesink.c:
+ audiobasesink: do big correction for large drift
+ If we are using skew slaving and we drift more than twice the allowed amount, do
+ a big correction to get back on track more quickly.
+
+2013-09-24 18:28:57 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 6b03ba7 to 865aa20
+
+2013-09-24 16:26:37 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Unset input/output_stream after freeing the GIOStream
+ watch->input_stream and watch->output_stream are owned by the GIOStream
+ and should be unset after freeing the stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708689
+
+2013-09-24 15:05:21 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ configure: Actually use 1.3.0.1 as version to make configure happy
+
+2013-09-24 15:00:20 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.2.0 ===
+
+2013-09-24 14:16:22 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 1.2.0
+
+2013-09-24 14:14:18 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * 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
+
+2013-09-24 12:47:26 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Make sure to cache context types we did not store yet
+ https://bugzilla.gnome.org/show_bug.cgi?id=708668
+
+2013-09-24 12:10:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Only create writesrc when it is actually needed
+ Creating a GSource and not attaching it to a context will cause
+ a leak of it's child sources. That is why we create writesrc right
+ before attaching it to a context.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708667
+
+2013-09-22 22:55:33 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst/adder/gstadder.c:
+ adder: send pending segment out before checking for EOS
+ Otherwise there would be cases where it would not send its segment
+ out when the first collected after getting it would already yield EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708590
+
+2013-09-19 17:25:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/video-frame.c:
+ video-frame: copy offsets from metadata
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708606
+
+2013-09-21 15:17:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: fix docs
+
+2013-09-20 16:16:42 +0200 Edward Hervey <edward@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From b613661 to 6b03ba7
+
+2013-09-19 18:42:49 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 74a6857 to b613661
+
+2013-09-19 17:34:46 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 098c0d7 to 74a6857
+
+2013-09-19 16:33:29 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ dmabuf: Fix compilation if no mmap is available
+ Also #ifdef some more code paths that don't make sense without mmap.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708372
+
+2013-09-19 12:58:53 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Switch to playing to handle live URI
+ Fixes discovery on dvb://
+
+2013-09-19 11:34:54 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.1.90 ===
+
+2013-09-19 10:49:58 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/libgstallocators.def:
+ Release 1.1.90
+
+2013-09-19 10:13:32 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * 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
+
+2013-09-18 20:42:55 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: If there is no PTS at all, assume it starts from the segment start
+ This is to make the multifilesrc ! pngdec case work
+ https://bugzilla.gnome.org/show_bug.cgi?id=688043
+
+2013-09-19 09:44:47 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * 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
+
+2013-09-18 22:05:36 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Implement context caching for sinks that are not in playsink yet
+
+2013-09-18 18:21:54 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: No need to set the GstContext on the sink before activating it
+ This is all handled by the GstBin code now.
+
+2013-09-04 20:21:54 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ * gst-libs/gst/allocators/gstdmabuf.h:
+ dmabuf: Make it not a singleton
+ Makes it easier to track how many users there are
+ Also make it possible to create a dmabuf struct on systems without mmap,
+ it just won't be possible to map it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707793
+
+2013-09-13 16:01:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: check for valid payload type
+ The payload type can't be between 72 and 76 because with the marker bit set,
+ this could be mistaken for an RTCP packet then. We do a relaxed check and
+ only refuse 72-76 when the marker bit is set. The effect is that when
+ we try to map an RTCP packet as an RTP packet, we will certainly fail.
+
+2013-09-13 09:17:38 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ configure: rely solely on pkg-config to find libogg and libvorbis
+ And get rid of AS_SCRUB_INCLUDES
+ https://bugzilla.gnome.org/show_bug.cgi?id=707658
+
+2013-09-12 12:23:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_4tap.c:
+ videoscale: fix 4tap for RGB15 and RGB16
+ Fix component ordering, it's wrong in both the scanline and merge
+ function so it cancels eachother out and isn't really a except for
+ loss of precision of the green component.
+ Fix calculation of the filter weight
+
+2013-09-10 17:02:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_scanline.c:
+ videoscale: optimize merge for RGB15 and RGB16
+
+2013-09-10 16:55:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_4tap.c:
+ videoscale: remove redundant MAX
+ The checks above make it inpossible for the value to be smaller than
+ what we check against with the MAX call.
+
+2013-09-12 09:42:36 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder/decoder: Mark pads as requiring reconfiguration again if negotiation fails
+ Otherwise we might end up in non-optimal configuration, especially
+ when a flush happened during reconfiguration.
+
+2013-09-12 09:35:00 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videodecoder/videoencoder: Mark pads as requiring reconfiguration again if negotiation fails
+ Otherwise we might end up in non-optimal configuration, especially
+ when a flush happened during reconfiguration.
+
+2013-09-10 21:44:33 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for TechSmith Screen Capture 2
+ https://bugzilla.gnome.org/show_bug.cgi?id=707878
+
+2013-09-10 21:44:21 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for TechSmith Screen Capture 2
+ https://bugzilla.gnome.org/show_bug.cgi?id=707878
+
+2013-09-06 15:36:12 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: check for full eos after a pad goes eos in push mode
+ After a pad is on EOS, verify if all pads are EOS and return
+ upstream, avoiding keeping the buffer flow without having more
+ data to push
+
+2013-09-06 15:56:39 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: properly handle stop position at seeks in push mode
+ Store the seek stop and seqnum and properly restore them when
+ receiving the corresponding Segment from upstream. Also fixes
+ seqnum for converted seek events.
+
+2013-09-10 16:16:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_4tap.c:
+ videoscale: fix RGB15 masks
+
+2013-09-10 16:06:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_scanline.c:
+ videoscale: simplify YUYV and UYVY linear scaling
+ Simplify the code and make it handle odd width
+
+2013-09-10 16:05:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_scanline.c:
+ videoscale: small cleanups
+ Use BLEND macro
+ Fix NV12 corner case
+
+2013-09-10 16:03:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_scanline.c:
+ videoscale: fix RGB15 masks
+
+2013-09-10 12:18:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_scanline.c:
+ videoscale: simplify nearest scaling
+ Round the accumulator to avoid later checks
+ Remove some bound checks that would never trigger
+ Fix odd width scaling
+
+2013-09-10 11:31:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoscale/vs_image.c:
+ videoscale: pick nearest line in scaling
+ Use rounding to pick the nearest line instead of rounding down.
+
+2013-09-03 17:27:37 +0100 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst-libs/gst/tag/id3v2.c:
+ * gst-libs/gst/tag/tags.c:
+ tag: id3: encapsulate ID3V2 blob frames in GstSample
+ id3mux and id3v2mux expect GST_TAG_ID3V2_FRAME type to be stored in a
+ GstSample and not a buffer, which is also needed because we can't
+ attach extradata/caps to buffers any more. These are private tags
+ no one should be poking at, and also the extra info is missing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707765
+
+2013-09-09 19:26:34 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: fix and improve raw video format description strings
+ Mark terms such as "planar", "packed", and "palettized" as
+ translatable, and re-arrange strings a bit to make them
+ better suited for translation.
+ Also fix bug in yuv descriptions, one plane is packed, more
+ is planar (or semi-planar).
+ https://bugzilla.gnome.org/show_bug.cgi?id=707789
+
+2013-09-09 15:52:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/video/gstvideometa.h:
+ * gst-libs/gst/video/video.h:
+ docs: fix some doc blocks
+
+2013-08-21 23:54:49 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst-libs/gst/video/gstvideofilter.c:
+ videofilter: implement transform_meta virtual method.
+ If tags of the meta only contain "video", let it be copied.
+
+2013-08-21 23:56:15 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/audio/gstaudiometa.c:
+ * gst-libs/gst/video/gstvideometa.c:
+ * gst-libs/gst/video/video.h:
+ video/audio: #define metadata strings.
+ For instance "orientation" becomes GST_VIDEO_ORIENTATION_METADATA.
+
+2013-09-07 19:14:50 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tools/gst-play.c:
+ tools: play: set playbin to NULL state on error to flush messages
+ Just flushing the bus doesn't work here for some reason, so set
+ playbin to NULL state, which seems to clear all error state and
+ makes sure we do play the next playable song and don't pick up
+ 'ghost' error messages from previous files on the bus.
+
+2013-09-06 23:17:44 +0200 Loïc Minier <lool@dooz.org>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playback: fix docs of convert-sample action signal
+ convert-sample returns a GstSample, not a GstBuffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707660
+
+2013-09-06 13:28:00 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ video: fix build without orc or older or versions
+ ./.libs/libgstvideo-1.0.so: undefined reference to `video_orc_unpack_NV24'
+ ./.libs/libgstvideo-1.0.so: undefined reference to `video_orc_pack_NV24'
+
+2013-09-06 12:44:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: disable fastpath for odd width on some formats
+
+2013-09-06 12:43:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/video-format.c:
+ * gst-libs/gst/video/video-orc.orc:
+ video-format: fix NV24 pack/unpack function
+ We can't reuse the NV12 functions, we need to make new ones.
+
+2013-09-06 12:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: handle odd width in more pack/unpack functions
+
+2013-09-05 18:33:28 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: minor pack_YVYU optimisation
+ Re-use already calculated line offset.
+
+2013-09-05 17:46:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: flush pending lines on odd height
+
+2013-09-05 17:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: add additional width/height constraints
+ Some of the fastpath function can only work with aligned widht/height
+ so make sure we check this as well when choosing a fastpath.
+ Add fastpath for I420/YV12 -> BGRx
+
+2013-09-05 17:06:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: fix chroma offsets
+
+2013-09-05 16:25:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: don't convert too much with odd width
+
+2013-09-05 16:15:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: fix unpack functions for odd formats
+
+2013-09-05 15:02:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: clean up pack/unpack functions
+
+2013-09-05 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: handle odd width in various pack functions
+
+2013-09-05 12:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: don't overrun the arrays on UYVP
+
+2013-09-05 11:05:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: handle lines in one go
+ Handle odd heights in 1 go when no vertical subsampling is used.
+
+2013-09-05 11:04:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: fix height round down
+
+2013-09-04 17:34:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: also allocate temp lines in fastpath
+ Some of the fastpath functions need tmplines, so make sure we allocate some in
+ the fastpath too.
+ This avoids SEGFAULTs with odd heights.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=663248
+
+2013-09-04 17:21:23 +0200 Christian Fredrik Kalager Schaller <uraeus@linuxrising.org>
+
+ * gst-plugins-base.spec.in:
+ Update specfile with latest changes
+
+2013-09-04 15:07:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: add more fastpaths
+ Also reuse the I420 code for YV12 because it can handle the swapped UV fields
+ just fine.
+
+2013-06-10 16:06:21 +0100 Alex Ashley <alex.ashley@youview.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Added "dash" and "avc3" fourCC codes to qt_type_find.
+ This commit adds detection of the "dash" and "avc3" compatible brands
+ in qt_type_find.
+ Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
+ structure for fragmented MP4 called "avc3". The principal difference
+ between AVC1 and AVC3 is the location of the codec initialisation
+ data (e.g. SPS, PPS). In AVC1 this data is placed in the initial MOOV
+ box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data goes in
+ the first sample of every fragment (i.e. the first sample in each mdat
+ box). The principal reason for avc3 is to make it easier for client
+ implementations, because it removes the requirement to insert the
+ SPS+PPS in to the decoder pipeline every time there is a representation
+ change.
+ https://bugzilla.gnome.org/show_bug.cgi?id=702004
+
+2013-08-31 01:05:40 +0200 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES.in
+ https://bugzilla.gnome.org/show_bug.cgi?id=707158
+
+2013-09-03 17:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: only chroma subsample when needed
+
+2013-09-03 15:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: fix handling of chroma resample
+ Increase the number of temporary lines that we need, it is possible that the
+ up and downsampling offsets are out of phase and that we need to keep some
+ extra lines around. Also copy the unhandled output lines for the next round
+ instead of overwriting them.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706823
+
+2013-09-03 15:41:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: improve debug
+
+2013-09-03 00:47:18 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Check for invalid stop position before calculating a duration from it
+ https://bugzilla.gnome.org/show_bug.cgi?id=707332
+
+2013-08-29 11:17:15 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ Require orc >= 0.4.18
+ Which contains important bug-fixes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=698520
+
+2013-08-30 15:19:32 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for MSS1 and MSS2 windows media formats
+
+2013-08-30 13:51:47 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Provide correct media type for MSS1 and MSS2
+ Windows Media Video Screen (WMV Screen) are video formats that
+ specilise in screencast content. This provides a correct media type
+ for them instead of just video/x-asf-unknown.
+
+2013-08-28 13:26:38 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.1.4 ===
+
+2013-08-28 12:41:42 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 1.1.4
+
+2013-08-28 12:31:23 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * 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
+
+2013-08-27 15:03:54 +0200 Andoni Morales Alastruey <ylatuya@gmail.com>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: fix forwarding of GstForceKeyUnit events
+ Use the frame id to match the output forced keyframe with
+ the event that forced it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=706885
+
+2013-08-26 11:44:06 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * ext/vorbis/gstvorbisenc.c:
+ * ext/vorbis/gstvorbisenc.h:
+ vorbisenc: remove unused variables
+
+2013-08-26 11:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: do additional packet checks
+ Check the packet size and avoid crashing on malformed packets.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=655727
+
+2013-08-26 11:46:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: improve bye parsing
+ It is an error to ask for a non-existing BYE SSRC, the caller should
+ check the SSRC count first.
+
+2013-08-23 18:06:36 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst-libs/gst/allocators/gstdmabuf.c:
+ dmabuf: fix mmap counting
+ A successful gst_dmabuf_mem_map must always increment the mmap count.
+ Otherwise the first gst_dmabuf_mem_unmap will unmap the memory and all
+ other user will access unmapped memory.
+ https://bugzilla.gnome.org/show_bug.cgi?id=706680
+
+2013-08-26 08:08:32 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * ext/vorbis/gstvorbisenc.c:
+ vorbisenc: implement flushing
+
+2013-08-25 10:25:43 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Make sure to return TRUE if the same caps are set again
+
+2013-08-23 19:47:57 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: improve fixate_format function readability even more
+ Do the flags comparisons only once and re-use the result.
+
+2013-08-23 19:41:32 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: simplify fixate_format function some more
+ If we have no output format yet, any format will do. The
+ !out_info condition existed in every path, so just split
+ it our for clarity. KISS.
+
+2013-08-23 19:05:41 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: make fixate function more readable
+ Use some variables to replace accessor macros to make code
+ a little bit mor readable.
+
+2013-08-23 18:52:44 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: remove unnecessary deep nesting in fixate function
+ Makes it easier to read and removes two levels of indentation.
+
+2013-08-23 19:20:03 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Only set the caps when they actually changed
+
+2013-08-23 19:17:16 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: Simplify pushing of pending events during negotiation
+ And also don't send the same caps twice.
+
+2013-08-23 19:10:48 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Fix last commit and simplify code a lot
+
+2013-08-23 18:51:59 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: If we have to lose precision, try to lose as less precision as possible
+ https://bugzilla.gnome.org/show_bug.cgi?id=706624
+
+2013-08-23 16:59:30 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Fix previous commit
+ (sorry)
+
+2013-08-23 15:22:43 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videocoder: Don't push out identical caps
+ This avoids triggering plenty of extra code/methods/overhead downstream when
+ we can just quickly check whenever we want to set caps whether they are
+ identical or not
+ https://bugzilla.gnome.org/show_bug.cgi?id=706600
+
+2013-08-23 15:22:05 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Don't push out identical caps
+ This avoids triggering plenty of extra code/methods/overhead downstream when
+ we can just quickly check whenever we want to set caps whether they are
+ identical or not
+ https://bugzilla.gnome.org/show_bug.cgi?id=706600
+
+2013-08-22 17:33:45 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Update segment.base with the chain's start time too
+ Fixes playback of chained ogg files.
+ https://bugzilla.gnome.org/show_bug.cgi?id=706569
+
+2013-08-22 14:18:29 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add typefinder for video/x-pva
+ https://bugzilla.gnome.org/show_bug.cgi?id=158719
+
+2013-08-21 16:02:00 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tools/gst-play.c:
+ gst-play: move current playlist index along in about-to-finish
+
+2013-08-21 15:39:30 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tools/gst-play.c:
+ gst-play: add --gapless mode
+ so we can test about-to-finish.
+
+2013-08-21 12:34:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbasedepayload.c:
+ rtpbasedepayload: mark DISCONT on buffer in all cases
+ Always mark discont on the input buffer when we detect a seqnum
+ discont and not only when we previously marked ourselves DISCONT.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706422
+
+2013-08-21 11:20:28 +0100 Rico Tzschichholz <ricotz@ubuntu.com>
+
+ * gst-libs/gst/video/gstvideometa.h:
+ videometa: fix syntax error
+
+2013-08-14 16:20:45 +0100 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst-libs/gst/tag/gstid3tag.c:
+ tag: id3: handle publisher, interpreted-by and musical-key tags
+ https://bugzilla.gnome.org/show_bug.cgi?id=705999
+
+2013-08-15 11:03:47 +0100 Matthieu Bouron <matthieu.bouron@collabora.com>
+
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ tag: add musical-key tag
+ https://bugzilla.gnome.org/show_bug.cgi?id=705999
+
+2013-08-19 10:39:19 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ * gst-libs/gst/pbutils/pbutils-private.h:
+ Revert "pbutils: allow describing unfixed caps if they share the same media type"
+ This reverts commit 065f1603b0f1d2adc8477bf1f3ebe2b154885d89.
+ This is not considered the correct solution, see:
+ https://bugzilla.gnome.org/show_bug.cgi?id=703378
+
+2013-08-16 13:22:33 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: improved and extended typefinder for module music formats
+ introduced new caps: audio/x-mod, modtype : { xm, okt, mod, ptm, ... }
+ https://bugzilla.gnome.org/show_bug.cgi?id=706061
+
+2013-07-15 16:13:11 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c:
+ rtpbaseaudiopayload: Avoid copying the data
+
+2013-08-17 16:58:06 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tests/icles/playback/test6.c:
+ tests: fix uridecodebin signal used in playback test6
+ "new-decoded-pad" no longer exists.
+
+2013-08-17 16:53:30 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tools/Makefile.am:
+ * tools/gst-play-1.0.1:
+ tools: add man page for new gst-play-1.0 utility
+ https://bugzilla.gnome.org/show_bug.cgi?id=553520
+
+2013-08-14 17:04:19 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/gst-i18n-app.h:
+ * tools/.gitignore:
+ * tools/Makefile.am:
+ * tools/gst-play.c:
+ tools: add simple command-line gst-play utility for testing purposes
+ Differs from a plain gst-launch-1.0 playbin uri=... pipeline in that
+ it can take multiple arguments and as such allows testing of things
+ like gapless playback, switching between different formats and the
+ like. Very minimal at this point, we'll probably want to add
+ interactive controls and more options at some point.
+ https://bugzilla.gnome.org/show_bug.cgi?id=553520
+
+2013-08-16 13:59:35 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: fix direct includes
+ https://bugzilla.gnome.org/show_bug.cgi?id=695889
+
+2013-08-16 13:55:33 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/pbutils/missing-plugins.h:
+ pbutils: fix direct includes
+ https://bugzilla.gnome.org/show_bug.cgi?id=695889
+
+2013-08-16 13:47:31 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * gst-libs/gst/video/gstvideoutils.h:
+ * gst-libs/gst/video/video-chroma.h:
+ * gst-libs/gst/video/video-frame.h:
+ video: make direct includes work again
+ Not nice to break people's code if we can avoid it. Could
+ add a warning in the next cycle, and then require single
+ includes in the cycle after.
+ https://bugzilla.gnome.org/show_bug.cgi?id=695889
+
+2013-08-16 13:06:58 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/audio/audio-channels.h:
+ * gst-libs/gst/audio/audio-format.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/gstaudiometa.h:
+ * gst-libs/gst/audio/gstaudioringbuffer.h:
+ * gst-libs/gst/audio/gstaudiosink.h:
+ * gst-libs/gst/audio/gstaudiosrc.h:
+ audio: make direct includes work again
+ Not nice to break people's code if we can avoid it. Could
+ add a warning in the next cycle, and then require single
+ includes in the cycle after.
+ https://bugzilla.gnome.org/show_bug.cgi?id=695889
+
+2013-08-16 14:12:32 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tests/icles/test-header-compile:
+ tests: add test-header-compile script
+ https://bugzilla.gnome.org/show_bug.cgi?id=695889
+
+2013-08-16 12:12:05 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ Revert "typefind: improved and extended typefinder for module music formats"
+ This reverts commit 4c79f35c7abc78bf4d325a8cd2059e8832ea0b34.
+ It causes some MP4 files to be detected as mod files.
+
+2013-08-13 23:18:34 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: improved and extended typefinder for module music formats
+ introduced new caps: audio/x-mod, modtype : { xm, okt, mod, ptm, ... }
+ https://bugzilla.gnome.org/show_bug.cgi?id=706061
+
+2013-08-15 14:15:05 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Don't reset too much if we're resetting because of a soft-flush
+ Fixes reverse playback with Ogg/Theora.
+
+2013-08-15 13:15:05 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ theora: Use new video codec base classes' flush vfunc
+
+2013-08-15 12:45:35 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Don't reset decoder on segment events
+ Either there was a flush before that resets everything anyway,
+ or resetting would make us lose information we might need if
+ it's just a segment update.
+
+2013-08-15 12:44:56 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/gstvideoencoder.h:
+ video{en,de}coder: Add new flush vfunc as a replacement for reset
+
+2013-08-14 16:55:55 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideodecoder.h:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ * gst-libs/gst/video/gstvideoencoder.h:
+ video{en,de}coder: Revert to old ::reset() behaviour and deprecate it
+
+2013-08-15 16:12:45 +0800 Jie Yang <yang.jie@intel.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: ADTS/AAC, find more aac sync to select correctly
+ https://bugzilla.gnome.org/show_bug.cgi?id=691462
+
+2013-08-14 15:43:23 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Don't set sink to NULL if a new one is set while the old one is still in use
+
+2013-08-14 11:43:50 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/gio/gstgiobasesrc.c:
+ gio: fix printf format compiler warning
+
+2013-08-13 20:39:15 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: document that "finished" and "discovered" signals are only emitted in async mode
+ https://bugzilla.gnome.org/show_bug.cgi?id=660195
+
+2013-08-13 17:39:34 +0200 Edward Hervey <edward@collabora.com>
+
+ * tests/check/elements/.gitignore:
+ check: Update .gitignore
+
+2013-08-13 17:39:25 +0200 Edward Hervey <edward@collabora.com>
+
+ * .gitignore:
+ .gitignore: Ignore files from automake test-driver
+
+2013-08-13 13:43:32 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * tests/check/elements/playbin-complex.c:
+ playbin-complex: Set fakesink as audio-sink to not use a real audio device
+ https://bugzilla.gnome.org/show_bug.cgi?id=705157
+
+2013-08-12 13:47:38 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add typefind function for WebP image format
+ https://bugzilla.gnome.org/show_bug.cgi?id=705826
+
+2013-08-04 01:01:25 +1000 Jonathan Matthew <jonathan@d14n.org>
+
+ * gst/gio/gstgiobasesrc.c:
+ gio: make better use of the cached buffer
+ When playing mp3 files from a smb server, we get 64k read requests
+ that mostly overlap. Without using the cache to partially satisfy
+ these, we send these requests straight to the server, resulting in
+ a lot more network traffic than necessary.
+ https://bugzilla.gnome.org/show_bug.cgi?id=705415
+
+2013-07-25 20:47:02 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Clear taglist on reception of a STREAM_START event
+ https://bugzilla.gnome.org/show_bug.cgi?id=705109
+
+2013-07-30 17:37:43 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Clear taglist on reception of a STREAM_START event
+ https://bugzilla.gnome.org/show_bug.cgi?id=705109
+
+2013-08-08 12:11:07 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Set proxy flags on the pads and use default event handler for simplicity
+ https://bugzilla.gnome.org//show_bug.cgi?id=705555
+
+2013-08-06 15:42:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: calculate FB packet length correctly
+
+2013-08-06 15:11:05 +0200 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/adder/gstadder.c:
+ adder: Raw buffers DTS should always be CLOCK_TIME_NONE
+
+2013-08-05 16:14:22 +0200 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/adder/gstadder.c:
+ adder: set DTS and PTS, sync on DTS
+
+2013-08-02 20:08:29 +0200 Arnaud Vrac <avrac@freebox.fr>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: check for tags on the right combiner instance
+ The get-tags actions are not working in all cases, because the track
+ number is used to resolve the stream combiner instead of the stream
+ type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=705369
+
+2013-08-02 16:57:43 -0700 David Schleef <ds@schleef.org>
+
+ * tests/check/Makefile.am:
+ tests: move orc removal to distclean
+
+2013-08-02 14:33:24 -0700 David Schleef <ds@schleef.org>
+
+ * configure.ac:
+ configure: create dir tests/check/orc
+ This is required now that subdir-objects is used, since automake
+ expects to create a .deps directory inside.
+
+2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
+
+ * configure.ac:
+ build: add subdir-objects to AM_INIT_AUTOMAKE
+ Fixes warnings with automake 1.14
+ https://bugzilla.gnome.org/show_bug.cgi?id=705350
+
+2013-08-02 11:00:06 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Demote ERROR statement back to DEBUG
+ It crawled in with david's latest commit
+
+2013-08-02 08:22:59 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Revert previous commit
+ The 'hard' argument of reset changed signification after the latest
+ start/stop/reset refactoring.
+
+2013-08-01 16:01:30 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Pass on 'hard' argument from _flush to _reset
+ When most of the code was moved from _flush() to _reset() the 'hard'
+ argument was no longer propagated.
+
+2013-07-31 11:26:58 -0700 David Schleef <ds@schleef.org>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: Add pinwheel and spokes patterns
+
+2013-07-30 15:58:26 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: private/teletext -> application/x-teletext
+
+2013-07-29 19:41:43 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * po/LINGUAS:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sl.po:
+ * po/sr.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ po: update translations
+
+2013-07-26 15:29:05 +0200 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Prevent seeks when _SCHEDULING_FLAG_SEQUENTIAL is set
+ Don't go into pull mode when the upstream scheduling flags indicate
+ seeks should be avoided by setting GST_SCHEDULING_FLAG_SEQUENTIAL.
+ https://bugzilla.gnome.org/show_bug.cgi?id=704929
+
+2013-07-29 14:47:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ Back to development
+
=== release 1.1.3 ===
-2013-07-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2013-07-29 13:37:00 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.1.3
+ * 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-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/libgstpbutils.def:
+ * win32/common/video-enumtypes.c:
+ Release 1.1.3
+
+2013-07-29 13:36:51 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
2013-07-29 12:11:38 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* tests/check/Makefile.am:
* tests/check/elements/playbin-complex.c:
- * tests/check/elements/playbin-compressed.c:
playbin: Rename compressed unit test to complex
It's not really about compressed streams anymore, but also
about stream switching and stream combiners.
Automatic update of common submodule
From 5edcd85 to 098c0d7
+2013-05-15 10:18:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/check/elements/opus.c:
+ opus: Fix event handling in unit test
+
2013-05-15 09:26:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/audio-info.c:
* gst-libs/gst/app/Makefile.am:
app: Don't use $(GST_PLUGIN_LIBTOOLFLAGS) for real libraries
+2012-10-24 12:16:39 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ gst: Add better support for static plugins
+
2012-10-24 12:10:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
uridecodebin: remove commented code
This is dead since ~6 years.
+2013-03-27 22:24:03 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Merge SBC decoder and encoder from bluez
+ https://bugzilla.gnome.org/show_bug.cgi?id=690582
+
+2007-08-23 19:12:23 +0000 Marcel Holtmann <marcel@holtmann.org>
+
+ sbc: Add SBC encoder and decoder skeletons for GStreamer
+
2013-03-12 08:10:23 +0100 Stefan Sauer <ensonic@users.sf.net>
* gst/audiotestsrc/gstaudiotestsrc.c:
Decoders that get unparsed input are internally leaking nearly
every incoming buffer. This checks that case.
+2013-02-11 11:06:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: clear the state of the decoder
+ Set the channels and rate back to their default values in _stop because they
+ are used to renegotiate when needed.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=692950
+
2013-02-09 16:50:05 +0000 Tim-Philipp Müller <tim@centricular.net>
* tests/check/elements/streamsynchronizer.c:
Automatic update of common submodule
From a942293 to 2de221c
+2013-01-28 14:12:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix crash when setting "cbr" property when encoder is not running yet
+ https://bugzilla.gnome.org/show_bug.cgi?id=692698
+
2013-01-27 09:45:59 +0530 B.Prathibha <prathibhab@cdac.in>
* tests/check/pipelines/basetime.c:
We need to initialize this variable because we can't be sure that the subclass
will set it.
+2012-12-18 16:56:28 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opus: use appropriate printf format for gsize
+
2012-12-18 15:34:42 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
* ext/vorbis/gstvorbisdec.c:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/app.h:
- * gst-libs/gst/app/gstapp.h:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudio.h:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/gstpbutils.h:
* gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/riff/gstriff.h:
* gst-libs/gst/riff/riff.h:
* gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/rtp/gstrtp.h:
* gst-libs/gst/rtp/rtp.h:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/rtsp.h:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/sdp.h:
* gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/tag/gsttag.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/gstvideo.h:
rtsp: fix GstRTSPMessage g-i annotations for out parameters
https://bugzilla.gnome.org/show_bug.cgi?id=687620
+2012-11-03 20:38:00 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * ext/opus/gstopus.c:
+ * ext/opus/gstopuscommon.c:
+ * ext/opus/gstopuscommon.h:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ * ext/opus/gstopusheader.c:
+ * ext/opus/gstopusheader.h:
+ * tests/check/elements/opus.c:
+ Fix FSF address
+ https://bugzilla.gnome.org/show_bug.cgi?id=687520
+
2012-11-03 23:05:09 +0000 Tim-Philipp Müller <tim@centricular.net>
* COPYING:
* win32/common/config.h:
Back to feature development
+2012-10-24 23:40:20 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: fixed buffer unmapping bug
+ When the decoder received a NULL buffer, it tried to
+ unmap a not mapped buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=686829
+
=== release 1.0.2 ===
2012-10-25 00:54:24 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/audio/gstaudiocdsrc.h:
audiocdsrc: mention TOCs in docs
+2012-10-17 17:34:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ Use gst_element_class_set_static_metadata()
+ where possible. Avoids some string copies. Also re-indent
+ some stuff. Also some indent fixes here and there.
+
2012-10-17 16:54:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/theora/gsttheoradec.c:
* ext/ogg/gstoggmux.c:
oggmux: send stream-start event
+2012-09-20 18:42:50 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/opus/gstopus.c:
+ opusenc: Rank as Primary
+
2012-09-22 16:07:35 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
* tests/check/libs/xmpwriter.c:
replace gst_tag_list_free with gst_tag_list_unref
+2012-09-14 17:08:49 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ replace gst_element_class_set_details_simple with gst_element_class_set_metadata
+
2012-09-14 17:02:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/theora/gsttheoradec.c:
* tests/check/elements/videotestsrc.c:
tests: port to the new GLib thread API
+2012-09-12 09:10:35 +0200 Peter Korsgaard <jacmet@sunsite.dk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opus + jpegformat: unbreak non-debug build
+ opus + jpegformat plugin builds fail when gstreamer is configured with
+ --disable-gst-debug as they are checking the GST_DISABLE_DEBUG symbol
+ instead of GST_DISABLE_GST_DEBUG.
+ Signed-off-by: Peter Korsgaard <jacmet@sunsite.dk>
+ https://bugzilla.gnome.org/show_bug.cgi?id=683850
+
2012-09-12 10:12:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/elements/videoscale.c:
video: Add support for 4:2:2 10 bit video.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683838
+2012-09-11 18:02:28 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/elements/opus.c:
+ test: Flush opus encoder between tests
+
+2012-09-11 18:01:58 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/elements/opus.c:
+ test: Flush opus encoder between tests
+
2012-09-11 20:53:16 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/tag/gsttagdemux.c:
... and therefore will never unblock the other streams.
Fixes blocking issue when using playbin suburi feature
+2012-09-11 14:31:49 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opusenc: port to the new GLib thread API
+
2012-09-11 12:53:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/video/video-info.c:
* tools/.gitignore:
* tools/Makefile.am:
* tools/gst-discoverer-1.0.1:
- * tools/gst-discoverer.1.in:
* tools/gst-visualise-m.m:
* tools/gst-visualise.1.in:
tools: remove gst-visualise script
* gst-libs/gst/riff/riff-read.c:
riff: fix build on big endian systems
+2012-08-04 16:31:30 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusheader.c:
+ gst_tag_list_free -> gst_tag_list_unref
+
2012-07-29 00:49:31 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/video/video-orc-dist.c:
* gst-libs/gst/video/video-orc-dist.h:
* gst-libs/gst/video/video-orc.orc:
- * gst-libs/gst/video/videoblendorc-dist.c:
- * gst-libs/gst/video/videoblendorc-dist.h:
- * gst-libs/gst/video/videoblendorc.orc:
orc: rename to video-orc*
2012-07-23 14:23:39 +0200 Robert Swain <robert.swain@collabora.co.uk>
2012-07-16 21:58:23 +0200 Stefan Sauer <ensonic@users.sf.net>
* ext/libvisual/Makefile.am:
- * ext/libvisual/gstaudiobasevisualizer.c:
* ext/libvisual/gstaudiobasevisualizer.h:
* ext/libvisual/gstaudiovisualizer.c:
* ext/libvisual/gstaudiovisualizer.h:
* ext/libvisual/Makefile.am:
* ext/libvisual/gstaudiobasevisualizer.c:
* ext/libvisual/gstaudiobasevisualizer.h:
- * ext/libvisual/gstbaseaudiovisualizer.c:
* ext/libvisual/gstbaseaudiovisualizer.h:
* ext/libvisual/visual.c:
* ext/libvisual/visual.h:
* gst-libs/gst/video/gstvideodecoder.h:
videodecoder: Add GstVideoDecoder::propose_allocation() vfunc
+2012-06-15 10:32:39 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: add missing mutex unlock on error path
+
+2012-06-15 10:24:24 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ * ext/opus/gstopusheader.h:
+ opus: set author to myself, and update copyright notices
+ because as slomo noted, in fact pretty much all the code in there is mine.
+
2012-06-14 23:08:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/examples/playback/playback-test.c:
* gst-libs/gst/video/video.h:
video: add support for premultiplied alpha
+2012-05-29 17:24:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: read gain from the right place in the header
+ It's at byte offset 16, not 14.
+
2012-05-29 17:48:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
When we need to add borders, take the pixel stride into account to move to the
right horizintal offset.
+2012-05-27 23:41:24 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: do not assert on bad header, error out instead
+
2012-05-26 19:56:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/libs/tag.c:
do not currently support) needs it to be specified in bytes. Thanks to
Julien Moutte for pointing this out.
+2012-05-24 22:12:56 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusheader.c:
+ opus: reject major version number above what we grok
+
+2012-05-24 21:58:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusheader.c:
+ opus: bump written version from 0 to 0x01
+ as per the spec update at https://wiki.xiph.org/OggOpus#ID_Header
+
+2012-04-30 14:40:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: fix lost packet handling for FEC/PLC
+ The base audio decoder sends zero size packets, not NULL buffers,
+ to signal dropped packets.
+
2012-05-24 13:43:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaybin2.c:
* gst-libs/gst/video/gstvideodecoder.h:
videodecoder: Change configure_buffer_pool() vfunc to decide_allocation() with same semantics as in basetransform
+2012-04-04 11:51:28 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * ext/opus/gstopusheader.c:
+ opus: Handle GstByteWriter return values
+
2012-04-19 14:41:40 +0200 Stefan Sauer <ensonic@users.sf.net>
* tests/check/pipelines/streamheader.c:
* gst-libs/gst/interfaces/.gitignore:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
- * gst-libs/gst/interfaces/navigation.c:
- * gst-libs/gst/interfaces/navigation.h:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/tuner.h:
* gst-libs/gst/interfaces/tunerchannel.c:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
- Update .po files
-
-2012-04-11 21:45:26 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-
- * gst/tcp/gstmultihandlesink.c:
- tcp: update property documentation to reference correct property
-
-2012-04-11 17:40:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/vorbis/gstvorbisenc.c:
- vorbisenc: fix channel mask
-
-2012-04-11 16:59:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/libs/struct_i386.h:
- tests: remove GstNetAddress
- Really, really remove all mention of GstNetBuffer
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673510
-
-2012-04-02 08:59:58 +0200 Alban Browaeys <prahal@yahoo.com>
-
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/pbutils/Makefile.am:
- * tests/examples/encoding/Makefile.am:
- * tools/Makefile.am:
- libs: Link against internal tag library
-
-2012-04-11 09:57:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * gst-libs/gst/audio/Makefile.am:
- audio: Remove obsolete FIXME 0.11
-
-2012-04-01 22:38:30 +0200 Alban Browaeys <prahal@yahoo.com>
-
- * gst-libs/gst/pbutils/Makefile.am:
- * tests/examples/encoding/Makefile.am:
- pbutils: Link against internal gst video
- Link pbutils and encoding tests against internal version of libgstvideo.
-
-2012-04-10 00:45:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * ext/alsa/gstalsamixerelement.c:
- * ext/alsa/gstalsasink.c:
- * ext/alsa/gstalsasrc.c:
- * ext/cdparanoia/gstcdparanoiasrc.c:
- * ext/libvisual/visual.c:
- * ext/ogg/gstoggaviparse.c:
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggparse.c:
- * ext/ogg/gstogmparse.c:
- * ext/pango/gstclockoverlay.c:
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextrender.c:
- * ext/pango/gsttimeoverlay.c:
- * ext/theora/gsttheoradec.c:
- * ext/theora/gsttheoraenc.c:
- * ext/theora/gsttheoraparse.c:
- * ext/vorbis/gstvorbisdec.c:
- * ext/vorbis/gstvorbisenc.c:
- * ext/vorbis/gstvorbisparse.c:
- * ext/vorbis/gstvorbistag.c:
- * gst/adder/gstadder.c:
- * gst/audioconvert/gstaudioconvert.c:
- * gst/audiorate/gstaudiorate.c:
- * gst/audioresample/gstaudioresample.c:
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/encoding/gstencodebin.c:
- * gst/encoding/gstsmartencoder.c:
- * gst/encoding/gststreamcombiner.c:
- * gst/encoding/gststreamsplitter.c:
- * gst/gdp/gstgdpdepay.c:
- * gst/gdp/gstgdppay.c:
- * gst/gio/gstgiosink.c:
- * gst/gio/gstgiosrc.c:
- * gst/gio/gstgiostreamsink.c:
- * gst/gio/gstgiostreamsrc.c:
- * gst/playback/gstdecodebin2.c:
- * gst/playback/gstplaybin2.c:
- * gst/playback/gstplaysink.c:
- * gst/playback/gstplaysinkaudioconvert.c:
- * gst/playback/gstplaysinkconvertbin.c:
- * gst/playback/gstplaysinkvideoconvert.c:
- * gst/playback/gststreamsynchronizer.c:
- * gst/playback/gstsubtitleoverlay.c:
- * gst/playback/gsturidecodebin.c:
- * gst/subparse/gstssaparse.c:
- * gst/subparse/gstsubparse.c:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gstmultihandlesink.c:
- * gst/tcp/gstmultioutputsink.c:
- * gst/tcp/gstmultisocketsink.c:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
- * gst/videoconvert/gstvideoconvert.c:
- * gst/videorate/gstvideorate.c:
- * gst/videoscale/gstvideoscale.c:
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/volume/gstvolume.c:
- * sys/ximage/ximagesink.c:
- * sys/xvimage/xvimagesink.c:
- Use new gst_element_class_set_static_metadata()
+ Update .po files
+
+2012-04-11 21:45:26 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/tcp/gstmultihandlesink.c:
+ tcp: update property documentation to reference correct property
+
+2012-04-11 17:40:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/vorbis/gstvorbisenc.c:
+ vorbisenc: fix channel mask
+
+2012-04-11 16:59:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/libs/struct_i386.h:
+ tests: remove GstNetAddress
+ Really, really remove all mention of GstNetBuffer
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673510
+
+2012-04-02 08:59:58 +0200 Alban Browaeys <prahal@yahoo.com>
+
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * tests/examples/encoding/Makefile.am:
+ * tools/Makefile.am:
+ libs: Link against internal tag library
+
+2012-04-11 09:57:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Remove obsolete FIXME 0.11
+
+2012-04-01 22:38:30 +0200 Alban Browaeys <prahal@yahoo.com>
+
+ * gst-libs/gst/pbutils/Makefile.am:
+ * tests/examples/encoding/Makefile.am:
+ pbutils: Link against internal gst video
+ Link pbutils and encoding tests against internal version of libgstvideo.
+
+2012-04-10 17:24:05 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * tests/check/elements/opus.c:
+ tests: port some more to 1.0
+
+2012-04-10 17:22:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: tweak caps negotiation
+ ... so as to avoid leaking caps or manipulating NULL caps.
+
+2012-04-10 00:45:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/alsa/gstalsamixerelement.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/libvisual/visual.c:
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstogmparse.c:
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisenc.c:
+ * ext/vorbis/gstvorbisparse.c:
+ * ext/vorbis/gstvorbistag.c:
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/encoding/gstencodebin.c:
+ * gst/encoding/gstsmartencoder.c:
+ * gst/encoding/gststreamcombiner.c:
+ * gst/encoding/gststreamsplitter.c:
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ * gst/gio/gstgiosink.c:
+ * gst/gio/gstgiosrc.c:
+ * gst/gio/gstgiostreamsink.c:
+ * gst/gio/gstgiostreamsrc.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysinkaudioconvert.c:
+ * gst/playback/gstplaysinkconvertbin.c:
+ * gst/playback/gstplaysinkvideoconvert.c:
+ * gst/playback/gststreamsynchronizer.c:
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/subparse/gstssaparse.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultihandlesink.c:
+ * gst/tcp/gstmultioutputsink.c:
+ * gst/tcp/gstmultisocketsink.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/videoconvert/gstvideoconvert.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/volume/gstvolume.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ Use new gst_element_class_set_static_metadata()
+
+2012-04-09 14:39:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update for new translatable strings
-2012-04-09 14:39:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+2012-04-06 14:52:12 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * po/af.po:
- * po/az.po:
- * po/bg.po:
- * po/ca.po:
- * po/cs.po:
- * po/da.po:
- * po/de.po:
- * po/el.po:
- * po/en_GB.po:
- * po/eo.po:
- * po/es.po:
- * po/eu.po:
- * po/fi.po:
- * po/fr.po:
- * po/gl.po:
- * po/hu.po:
- * po/id.po:
- * po/it.po:
- * po/ja.po:
- * po/lt.po:
- * po/lv.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/pl.po:
- * po/pt_BR.po:
- * po/ro.po:
- * po/ru.po:
- * po/sk.po:
- * po/sl.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/tr.po:
- * po/uk.po:
- * po/vi.po:
- * po/zh_CN.po:
- po: update for new translatable strings
+ Merge remote-tracking branch 'origin/0.10'
+ Conflicts:
+ gst/h264parse/gsth264parse.c
+ gst/videoparsers/gsth264parse.c
2012-04-06 10:54:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin.c:
playback: Remove gstdecodebin.c, which is nowaday unused anyway
+2012-04-05 17:15:11 -0400 Thibault Saunier <thibault.saunier@collabora.com>
+
+ Merge remote-tracking branch 'origin/0.10'
+
2012-04-05 18:42:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 7fda524 to 464fe15
+2012-04-05 18:02:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/opus/gstopus.c:
+ gst: Update for GST_PLUGIN_DEFINE() API changes
+
2012-04-05 15:11:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/alsa/gstalsaplugin.c:
* win32/common/config.h:
gst: Update versioning
+2012-04-04 14:41:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ gst: Update versioning
+
+2012-04-04 12:06:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/0.10'
+
2012-04-04 09:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst/videoconvert/gstvideoconvert.c:
videoconvert: plug caps leak
+2012-04-02 15:31:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/0.10'
+ Conflicts:
+ gst/mpegtsdemux/tsdemux.c
+
2012-04-02 14:23:16 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.h:
* tests/examples/app/appsrc-stream2.c:
update for buffer api change
+2012-03-30 17:09:34 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fixup merge
+
2012-03-30 16:56:45 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* tests/check/elements/appsrc.c:
* gst-libs/gst/audio/gstaudiodecoder.h:
audiodecoder: Rename ::event() to ::sink_event() and add ::src_event()
+2012-03-30 12:22:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ ext: Update for GstAudioEncoder API changes
+
2012-03-30 12:13:40 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst-libs/gst/tag/gstexiftag.c:
Which is telling more about what this actually does and is more
consistent with the video base classes.
+2012-03-29 18:04:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-bad
+
2012-03-29 17:41:55 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* tests/check/libs/libsabi.c:
* docs/design/draft-hw-acceleration.txt:
design: First go at hardware-acceleration design doc
+2012-03-29 17:41:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/0.10'
+ Conflicts:
+ NEWS
+ RELEASE
+ common
+ configure.ac
+ docs/libs/gst-plugins-bad-libs-sections.txt
+ docs/plugins/gst-plugins-bad-plugins.args
+ docs/plugins/gst-plugins-bad-plugins.hierarchy
+ docs/plugins/gst-plugins-bad-plugins.interfaces
+ docs/plugins/inspect/plugin-adpcmdec.xml
+ docs/plugins/inspect/plugin-adpcmenc.xml
+ docs/plugins/inspect/plugin-assrender.xml
+ docs/plugins/inspect/plugin-audiovisualizers.xml
+ docs/plugins/inspect/plugin-autoconvert.xml
+ docs/plugins/inspect/plugin-bayer.xml
+ docs/plugins/inspect/plugin-bz2.xml
+ docs/plugins/inspect/plugin-camerabin2.xml
+ docs/plugins/inspect/plugin-celt.xml
+ docs/plugins/inspect/plugin-dataurisrc.xml
+ docs/plugins/inspect/plugin-debugutilsbad.xml
+ docs/plugins/inspect/plugin-dtmf.xml
+ docs/plugins/inspect/plugin-dtsdec.xml
+ docs/plugins/inspect/plugin-dvbsuboverlay.xml
+ docs/plugins/inspect/plugin-dvdspu.xml
+ docs/plugins/inspect/plugin-faac.xml
+ docs/plugins/inspect/plugin-faad.xml
+ docs/plugins/inspect/plugin-gsm.xml
+ docs/plugins/inspect/plugin-h264parse.xml
+ docs/plugins/inspect/plugin-mms.xml
+ docs/plugins/inspect/plugin-modplug.xml
+ docs/plugins/inspect/plugin-mpeg2enc.xml
+ docs/plugins/inspect/plugin-mpegdemux2.xml
+ docs/plugins/inspect/plugin-mpegtsdemux.xml
+ docs/plugins/inspect/plugin-mpegvideoparse.xml
+ docs/plugins/inspect/plugin-mplex.xml
+ docs/plugins/inspect/plugin-pcapparse.xml
+ docs/plugins/inspect/plugin-rawparse.xml
+ docs/plugins/inspect/plugin-rtpmux.xml
+ docs/plugins/inspect/plugin-rtpvp8.xml
+ docs/plugins/inspect/plugin-scaletempo.xml
+ docs/plugins/inspect/plugin-schro.xml
+ docs/plugins/inspect/plugin-sdp.xml
+ docs/plugins/inspect/plugin-segmentclip.xml
+ docs/plugins/inspect/plugin-shm.xml
+ docs/plugins/inspect/plugin-videomaxrate.xml
+ docs/plugins/inspect/plugin-videoparsersbad.xml
+ docs/plugins/inspect/plugin-vp8.xml
+ docs/plugins/inspect/plugin-y4mdec.xml
+ ext/celt/gstceltdec.c
+ ext/dts/gstdtsdec.c
+ ext/modplug/gstmodplug.cc
+ ext/opus/gstopusenc.c
+ gst-libs/gst/video/gstbasevideocodec.c
+ gst-libs/gst/video/gstbasevideocodec.h
+ gst-libs/gst/video/gstbasevideodecoder.c
+ gst-libs/gst/video/gstbasevideodecoder.h
+ gst-libs/gst/video/gstbasevideoencoder.c
+ gst-libs/gst/video/gstbasevideoencoder.h
+ gst/adpcmdec/Makefile.am
+ gst/audiovisualizers/gstbaseaudiovisualizer.c
+ gst/h264parse/gsth264parse.c
+ gst/mpegdemux/mpegtsparse.c
+ gst/mpegtsdemux/mpegtsbase.c
+ gst/mpegtsdemux/mpegtspacketizer.c
+ gst/mpegtsdemux/mpegtsparse.c
+ gst/mpegtsdemux/tsdemux.c
+ gst/mpegtsdemux/tsdemux.h
+ gst/mxf/mxfdemux.c
+ gst/rawparse/gstaudioparse.c
+ gst/videoparsers/gsth263parse.c
+ gst/videoparsers/gsth264parse.c
+ sys/d3dvideosink/d3dvideosink.c
+ sys/decklink/gstdecklinksink.cpp
+ sys/dvb/gstdvbsrc.c
+ sys/shm/gstshmsrc.c
+ sys/vdpau/h264/gstvdph264dec.c
+ sys/vdpau/mpeg/gstvdpmpegdec.c
+ tests/examples/opencv/gst_element_print_properties.c
+ win32/common/config.h
+
2012-03-29 17:14:48 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbasepayload.c:
* gst/gdp/gstgdppay.c:
update for buffer changes
+2012-03-27 15:13:24 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/opus/gstopus.c:
+ opus: Rank rtp pay/depay
+ This way they can be auto-plugged.
+
2012-03-27 18:16:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/tag/gsttagmux.c:
* tests/check/elements/decodebin2.c:
tests: update for caps api changes
+2012-03-12 17:06:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: fix for caps api change
+
2012-03-12 16:39:14 +0200 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
* configure.ac:
buffers. Users of the bufferpool should do this manually based on the results of
the allocation query.
+2012-03-08 11:32:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/opus.c:
+ tests: fix more caps
+
2012-03-08 10:59:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/elements/videoscale.c:
Simply intersect the format with the supported formats to make the code deal
with lists of formats.
+2012-03-07 17:14:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/opus/gstopuscommon.c:
+ * ext/opus/gstopuscommon.h:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusheader.c:
+ opus: port to updated 0.11
+
2012-03-07 12:45:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
common: update common module
For new check-norepeat target.
+2012-03-07 12:59:28 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: only request and process 1 frame at a time
+ ... since it is specified in _finish_frame that input buffer may be invalidated
+ after calling it, and is as such not reliably available for further encoding.
+ Also, requesting or allowing several frames is only useful if subclass intends
+ to process these "in 1 run" (as in, 1 output buffer), not for having another
+ (inner) loop in subclass where the baseclass one will do just fine.
+
+2012-03-07 12:55:43 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: configure baseclass requested samples really in samples
+ ... as opposed to bytes.
+
2012-03-07 09:04:18 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* win32/common/libgstaudio.def:
* tests/examples/playback/Makefile.am:
* tests/examples/playback/playback-test.c:
- * tests/examples/playback/seek.c:
playback: Rename file from seek.c to playback-test.c
2012-03-02 11:57:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/playback/Makefile.am:
* tests/examples/playback/seek.c:
* tests/examples/seek/Makefile.am:
- * tests/examples/seek/seek.c:
examples: Move seek example into its own directory
2012-03-02 11:01:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/icles/test-colorkey.c:
Suppress deprecation warnings in selected files, for g_value_array_* mostly
+2012-02-27 13:13:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ audioencoders: chain up to parent event handler
+
2012-02-27 13:08:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
happy.
https://bugzilla.gnome.org/show_bug.cgi?id=670548
+2012-02-21 10:06:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ gst/colorspace/colorspace.c
+
2012-02-21 10:05:20 +0100 David Schleef <ds@schleef.org>
* gst/videoconvert/videoconvert.c:
videoconvert: clamp intermediates when dithering
Port from the colorspace plugin in -bad.
+2012-02-20 16:07:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ ext/opus/gstopusparse.c
+ gst/colorspace/colorspace.c
+
2012-02-20 15:29:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* tests/examples/seek/seek.c:
* win32/common/libgstaudio.def:
defs: update
+2012-02-17 09:01:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2012-02-16 14:33:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ gst/mpegtsdemux/mpegtsbase.c
+ gst/mpegtsdemux/mpegtspacketizer.c
+ gst/mpegtsdemux/tsdemux.c
+ gst/mve/gstmvedemux.c
+
2012-02-16 14:23:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: assert some more that subclass parsed frame has proper len
+2012-02-15 17:14:34 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2012-02-15 13:42:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
tagdemux: fix src query handler
We don't want to blindly forward all queries.
+2012-02-14 11:19:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2012-02-14 10:50:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/elements/decodebin.c:
method to get to the padtemplates. Fixes 'GstTagDemux subclass GstTagDemux
did not set up a {sink,src} pad template' warnings.
+2012-02-10 16:46:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ ext/chromaprint/gstchromaprint.c
+ ext/mpeg2enc/Makefile.am
+ ext/voaacenc/gstvoaacenc.c
+ gst/dvbsuboverlay/gstdvbsuboverlay.c
+ gst/mpegtsdemux/mpegtsbase.c
+ gst/sdp/gstsdpdemux.c
+ gst/videoparsers/gsth264parse.c
+ sys/d3dvideosink/d3dvideosink.c
+ tests/examples/camerabin/gst-camera-perf.c
+ tests/examples/camerabin/gst-camerabin-test.c
+ tests/examples/camerabin2/gst-camerabin2-test.c
+ tests/examples/mxf/mxfdemux-structure.c
+ tests/examples/scaletempo/demo-main.c
+
2012-02-10 15:41:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/elements/videoscale.c:
audioencoder: don't unref caps parameter
Fix refcounting on incomming caps to make sure we don't unref it too much.
+2012-02-03 00:50:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ build: fix CFLAGS order and LIBS order
+ _BAD_CFLAGS should always come first, then GST_PLUGINS_BASE_CFLAGS,
+ then GST_BASE_CFLAGS then GST_CFLAGS. Same for libs: first plugins
+ base libs, then GST_BASE_LIB then GST_LIBS.
+
2012-01-07 23:09:23 -0500 Ryan Lortie <desrt@desrt.ca>
* autogen.sh:
* sys/v4l/v4lsrc_calls.c:
v4l: include the glib compatiblity header for the deprecated mutex API
+2012-01-27 14:49:58 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopusenc.c:
+ plenty: fixup glib deprecations
+
2012-01-27 15:12:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
Merge branch 'master' into 0.11
Automatic update of common submodule
From c463bc0 to 7fda524
+2012-01-25 13:22:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ configure.ac
+ ext/kate/gstkateenc.c
+ gst/colorspace/colorspace.c
+ gst/mpegvideoparse/mpegvideoparse.c
+
2012-01-25 12:50:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/adder/gstadder.c:
* configure.ac:
* docs/plugins/Makefile.am:
* ext/Makefile.am:
- * ext/gio/Makefile.am:
- * ext/gio/gstgio.c:
- * ext/gio/gstgio.h:
- * ext/gio/gstgiobasesink.c:
- * ext/gio/gstgiobasesink.h:
- * ext/gio/gstgiobasesrc.c:
- * ext/gio/gstgiobasesrc.h:
- * ext/gio/gstgiosink.c:
- * ext/gio/gstgiosink.h:
- * ext/gio/gstgiosrc.c:
- * ext/gio/gstgiosrc.h:
- * ext/gio/gstgiostreamsink.c:
- * ext/gio/gstgiostreamsink.h:
- * ext/gio/gstgiostreamsrc.c:
- * ext/gio/gstgiostreamsrc.h:
* gst/gio/Makefile.am:
* gst/gio/gstgio.c:
* gst/gio/gstgio.h:
* docs/plugins/Makefile.am:
* gst/tcp/Makefile.am:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gstmultifdsink.h:
* gst/tcp/gstmultisocketsink.c:
* gst/tcp/gstmultisocketsink.h:
* gst/tcp/gsttcp-marshal.list:
it seems pretty certain it's the right thing to do, but I'll put
this caveat here in case someone checks in the future.
+2012-01-13 00:11:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/master' into 0.11
+
2012-01-12 23:35:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
freed data with chained and normal files, both with gst-launch
playbin2 and Totem.
+2012-01-11 13:32:36 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * tests/check/elements/opus.c:
+ tests: fix buffer leaks in opus tests
+
2012-01-11 12:52:17 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Don't unref the parent in the event function
+2012-01-10 15:50:37 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ gst/mpegtsdemux/tsdemux.c
+ gst/videoparsers/gsth264parse.c
+ tests/check/elements/camerabin2.c
+
+2012-01-10 13:38:50 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix slist leak
+
+2012-01-10 13:38:42 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix caps leak
+
2012-01-10 13:15:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
Merge branch 'master' into 0.11
gst/playback/gstsubtitleoverlay.c
tests/check/libs/tag.c
+2011-12-30 11:49:27 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/master' into 0.11
+ Conflicts:
+ tests/examples/camerabin2/Makefile.am
+
+2011-12-30 11:41:17 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/master' into 0.11-premerge
+ Conflicts:
+ docs/libs/Makefile.am
+ ext/kate/gstkatetiger.c
+ ext/opus/gstopusdec.c
+ ext/xvid/gstxvidenc.c
+ gst-libs/gst/basecamerabinsrc/Makefile.am
+ gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
+ gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
+ gst-libs/gst/video/gstbasevideocodec.c
+ gst-libs/gst/video/gstbasevideocodec.h
+ gst-libs/gst/video/gstbasevideodecoder.c
+ gst-libs/gst/video/gstbasevideoencoder.c
+ gst/asfmux/gstasfmux.c
+ gst/audiovisualizers/gstwavescope.c
+ gst/camerabin2/gstcamerabin2.c
+ gst/debugutils/gstcompare.c
+ gst/frei0r/gstfrei0rmixer.c
+ gst/mpegpsmux/mpegpsmux.c
+ gst/mpegtsmux/mpegtsmux.c
+ gst/mxf/mxfmux.c
+ gst/videomeasure/gstvideomeasure_ssim.c
+ gst/videoparsers/gsth264parse.c
+ gst/videoparsers/gstmpeg4videoparse.c
+
2011-12-28 16:25:37 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* tests/check/libs/video.c:
2011-12-20 10:08:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/design/design-audiosinks.txt:
- * docs/design/draft-media-types.txt:
* docs/design/part-interlaced-video.txt:
* docs/design/part-mediatype-video-raw.txt:
* docs/design/part-playbin.txt:
- * docs/design/part-playbin2.txt:
docs: small update to design docs
2011-12-19 23:41:25 +0100 Stefan Sauer <ensonic@users.sf.net>
A more robust way would be to find a good place to reinject the
headers when a seek fails, but I can't seem to get this to work.
+2011-12-15 16:42:20 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opus: fix bad merge (stray unmap, undeclared var)
+
2011-12-15 11:01:01 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst-libs/gst/tag/gstexiftag.c:
* po/sr.po:
po: update translations
+2011-12-09 17:25:41 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusheader.c:
+ opusenc: add upstream negotiation for multistream ability
+ This will help elements that cannot deal with multistream,
+ such as the RTP payloader.
+ The caps now do not include a "streams" field anymore, but
+ a "multistream" boolean, since we have no real use for knowing
+ the exact amount of streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
2011-12-09 19:21:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtsp/gstrtsptransport.c:
* tests/check/libs/video.c:
tests: disable composition tests in video unit test for now
+2011-12-07 15:13:11 -0200 Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopus.c:
+ Adding opus RTP payloader/depayloader element
+ Adding OPUS RTP module based on the current draft:
+ http://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.txt
+ https://bugzilla.gnome.org/show_bug.cgi?id=664817
+
+2011-12-08 19:47:55 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusheader.c:
+ opus: include streams count in caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
+2011-12-08 18:45:27 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopuscommon.c:
+ * ext/opus/gstopuscommon.h:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ * ext/opus/gstopusheader.c:
+ * ext/opus/gstopusheader.h:
+ opus: properly create channel mapping tables
+ There are two of them, unintuitively enough; the one passed
+ to the encoder should not be the one that gets written to the
+ file. The former maps the input to an ordering which puts
+ paired channels first, while the latter moves the channels
+ to Vorbis order. So add code to calculate both, and we now
+ have properly paired channels where appropriate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
2011-12-09 15:03:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.h:
caps and not random caps, and it's hard to imagine a situation
where someone would want to rely on the previous behaviour.
+2011-12-07 00:06:11 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: header cleanup
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
+2011-12-07 00:06:11 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: Truncate caps first
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
+2011-11-28 19:47:34 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: default to stereo 48000 Hz if possible when no headers seen
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
2011-12-06 21:57:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/videorate/gstvideorate.c:
* sys/xvimage/xvimagesink.c:
update for basesink event handler changes
+2011-11-28 19:38:34 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: guard against decoding 0 samples
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
2011-12-02 11:10:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
Merge remote-tracking branch 'origin/master' into 0.11
* gst-libs/gst/audio/streamvolume.h:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
- * gst-libs/gst/interfaces/mixer.c:
- * gst-libs/gst/interfaces/mixer.h:
- * gst-libs/gst/interfaces/mixeroptions.c:
- * gst-libs/gst/interfaces/mixeroptions.h:
- * gst-libs/gst/interfaces/mixertrack.c:
- * gst-libs/gst/interfaces/mixertrack.h:
- * gst-libs/gst/interfaces/streamvolume.c:
- * gst-libs/gst/interfaces/streamvolume.h:
* gst/playback/Makefile.am:
* gst/playback/gstplaybin2.c:
* gst/volume/gstvolume.c:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/interfaces/colorbalance.c:
- * gst-libs/gst/interfaces/colorbalance.h:
- * gst-libs/gst/interfaces/colorbalancechannel.c:
- * gst-libs/gst/interfaces/colorbalancechannel.h:
- * gst-libs/gst/interfaces/videoorientation.c:
- * gst-libs/gst/interfaces/videoorientation.h:
- * gst-libs/gst/interfaces/videooverlay.c:
- * gst-libs/gst/interfaces/videooverlay.h:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/colorbalance.c:
* gst-libs/gst/video/colorbalance.h:
* gst-libs/gst/video/video.h:
libgstvideo: Add force key unit events
+2011-11-28 23:20:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/master' into 0.11
+
+2011-11-28 23:20:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ Merge commit '7521b597f4dc49d8d168f368f0e7ebaf98a72156' into 0.11
+
+2011-11-28 23:20:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ Merge commit '26d6add9457f00ce8ec13844368466f0e3816e5d' into 0.11
+ Conflicts:
+ ext/rtmp/gstrtmpsink.c
+
2011-11-28 21:25:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
Merge remote-tracking branch 'origin/master' into 0.11
various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
+2011-11-28 13:08:27 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ various: fix pad template ref leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=662664
+
2011-09-07 16:04:14 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/theora/gsttheoradec.c:
If highres-timestamp is 0, try lowres and if that fails fallback to system clock
timestamps.
+2011-11-27 23:33:45 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/master' into 0.11
+
2011-11-27 20:14:08 +0100 Matej Knopp <matej.knopp@gmail.com>
* gst/playback/gsturidecodebin.c:
uridecodebin: fix debug message printf format compiler warning
https://bugzilla.gnome.org/show_bug.cgi?id=662607
+2011-11-26 15:37:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/master' into 0.11
+ Conflicts:
+ ext/opus/gstopusdec.c
+ ext/opus/gstopusparse.c
+ gst-libs/gst/video/gstbasevideodecoder.c
+ gst-libs/gst/video/gstbasevideodecoder.h
+
2011-11-26 12:12:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
Merge remote-tracking branch 'origin/master' into 0.11
oggmux: set collectpads2 not to wait on sparse streams
https://bugzilla.gnome.org/show_bug.cgi?id=663174
+2011-11-25 11:41:19 -0200 Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opusenc: Fixing "Unused var" compiling error for opus codec
+ https://bugzilla.gnome.org/show_bug.cgi?id=664815
+
+2011-11-25 14:00:18 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusheader.c:
+ opusenc: only use mono streams for > 2 channels
+ I'm getting odd results with packing streams into stereo
+ streams, and using only mono streams is enough in all cases.
+
+2011-11-25 12:47:42 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopuscommon.c:
+ * ext/opus/gstopuscommon.h:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opus: add some more debug information about channel mapping
+
+2011-11-25 12:40:31 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: do not cause the decoder to apply the channel mapping again
+ Since we already reorder channels, we do not want to write that
+ reordering in the header, or the decoder will do it again.
+
+2011-11-25 12:39:20 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: fix bogus assertion
+
2011-11-25 15:35:39 +0100 Josep Torra <n770galaxy@gmail.com>
* gst/playback/gstplaysinkconvertbin.c:
* gst/playback/gstplaybin2.c:
docs: mention explicitly that playbin2 signals are emitted from a streaming thread
+2011-11-25 12:48:58 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/master' into 0.11
+ Conflicts:
+ ext/faac/gstfaac.c
+ ext/opus/gstopusdec.c
+ ext/opus/gstopusenc.c
+ gst/audiovisualizers/gstspacescope.c
+ gst/colorspace/colorspace.c
+
2011-11-25 11:11:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
This new property will force the output framerate to
a specific value and can be changed during playback.
+2011-11-24 13:38:59 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusheader.c:
+ opus: pre-skip and output gain are little endian, remove reminder note
+
+2011-11-24 13:29:56 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopuscommon.c:
+ * ext/opus/gstopuscommon.h:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ * ext/opus/gstopusheader.c:
+ * ext/opus/gstopusheader.h:
+ opus: multichannel support
+
+2011-11-23 17:49:58 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opus: switch to multistream API
+ It's very similar to the basic API, and is a superset ot it,
+ which will allow encoding and decoding more than 2 channels.
+
+2011-11-23 17:32:03 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: shuffle supported sample rates to favor 48000
+
+2011-11-23 16:36:54 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opusenc: remove useless setup field
+
2011-11-24 12:38:54 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
* ext/vorbis/gstvorbisenc.c:
vorbisenc: do not accept 256 channels, 255 is the max vorbis supports
+2011-11-23 13:22:12 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: implement replay gain
+ It would ideally be better to leave this to a rgvolume element,
+ but we don't control the pipeline. So do it by default, and allow
+ disabling it via a property, so the correct volume should always
+ be output.
+
+2011-11-23 11:58:54 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: add in-band FEC support
+ This allows reconstruction of lost packets if FEC info is included
+ in the next packet, at the cost of extra latency. Since we do not
+ know if the stream has FEC (and this can change at runtime), we
+ always incur the latency, even if we never lose any frame, or see
+ any FEC information. Off by default.
+
2011-11-23 11:10:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggstream.c:
ogg: fix compilation
+2011-11-23 11:08:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ ext/opus/gstopusdec.c
+ ext/opus/gstopusenc.c
+ ext/opus/gstopusparse.c
+ gst/audiovisualizers/gstwavescope.c
+ gst/filter/Makefile.am
+ gst/filter/gstfilter.c
+ gst/filter/gstiir.c
+ gst/playondemand/gstplayondemand.c
+
2011-11-23 10:50:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
Conflicts:
ext/ogg/gstoggmux.c
+2011-11-22 20:27:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: mark properties changeable at runtime with GST_PARAM_MUTABLE_PLAYING
+
+2011-11-22 18:33:17 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * tests/check/elements/opus.c:
+ opus: add test
+
+2011-11-22 17:04:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opusenc: allow setting most properties at PLAYING time
+ Opus allows these to be changed during encoding, transparently
+ to the decoder.
+
+2011-11-22 16:14:06 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: bound the bitrate to more sensible values
+ Go from the bounds mentioned in the spec, and allow some more
+ variation.
+ In particular, don't allow silly low bitrates, and allow reaching
+ the maximum useful bitrate.
+
+2011-11-22 15:33:20 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opusenc: fix crash on pathological parameters
+ Asking for 1 bit/s would select a 0 byte buffer, leading
+ to a crash. Buffer size is now controlled by a max-payload-size
+ property, which can't be less than 2.
+
2011-11-22 13:29:10 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
oggstream: extract opus comments if available
+2011-11-21 17:48:54 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusheader.c:
+ * ext/opus/gstopusheader.h:
+ opus: move header magic testing to gstopusheader
+
+2011-11-21 17:01:49 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: skip pre-skip samples
+
+2011-11-21 12:50:22 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: read pre-skip from first header if available
+
2011-11-22 13:15:33 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggstream.c:
xvimagebufferpool: Use the default ::free_buffer() implementation
Which does exactly the same thing
+2011-11-21 12:02:28 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: reset tagsetter interface on stop
+
+2011-11-21 11:44:01 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: handle NULL packets (used for PLC)
+
+2011-11-21 11:28:10 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: light cleanup
+
+2011-11-20 09:58:06 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: do not push header buffers
+ Opus headers appear only when muxed in Ogg, so only place them
+ on the caps, where oggmux will find them, but other elements will
+ be blithely unaware of them.
+
+2011-11-20 09:52:46 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusheader.c:
+ * ext/opus/gstopusheader.h:
+ opus: make opusparse set headers on caps
+ Header-on-caps code moved to a new shared location to avoid
+ duplicating the code.
+
2011-11-19 16:06:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggstream.c:
ogg: add opus support
+2011-11-19 15:58:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix terminating NUL being written in signature
+
+2011-11-16 19:40:20 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: make frame-size an enum
+ It only supports a set number of specific values (including
+ a non integer one).
+
+2011-11-16 19:22:44 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: the encoder might not make use of all the bytes
+
2011-11-18 17:58:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/gio/gstgiosrc.c:
* gst-libs/gst/audio/gstaudiobasesink.c:
fix for scheduling mode rename
+2011-11-17 17:32:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ ext/celt/gstceltdec.c
+ ext/opus/gstopusdec.c
+ ext/opus/gstopusdec.h
+ ext/opus/gstopusenc.c
+ ext/opus/gstopusenc.h
+ ext/opus/gstopusparse.c
+
2011-11-17 17:07:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
* gst/adder/gstadder.c:
collectpads: port API changes
+2011-11-16 18:49:03 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: do not include variable fields in caps
+ Those can vary from one packet to the next, so have no reason
+ to be in the caps.
+
+2011-11-16 18:43:53 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix constrained-vbr property name typo
+
+2011-11-16 18:35:29 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: let the base class handle all timing
+
2011-11-16 19:00:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* ext/vorbis/gstvorbisenc.c:
... which ensures nothing subsequently tries to slip past _chain
and into a possibly improperly setup subclass.
+2011-11-15 19:53:33 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopus.c:
+ opusparse: add opusparse element
+ A very simple element that parses Opus streams from the ad hoc
+ framing used by the Opus test vectors.
+
+2011-11-16 17:24:20 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: allow negotiation of rate/channels with downstream
+ Since an opus stream may be decoded to any (sensible) rate,
+ and either stereo or mono, we try to accomodate downstream.
+
+2011-11-16 17:05:17 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: rewrite logic
+ Parameters such as frame size, etc, are variable. Pretty much
+ everything can change within a stream, so be prepared about it,
+ and do not cache parameters in the decoder.
+
+2011-11-16 16:56:43 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opus: port to base audio encoder/decoder
+
2011-11-15 13:29:31 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst/subparse/gstsubparse.c:
add parent to query function
+2011-11-16 13:26:35 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: allow negotiation of rate/channels with downstream
+ Since an opus stream may be decoded to any (sensible) rate,
+ and either stereo or mono, we try to accomodate downstream.
+
+2011-11-16 01:14:32 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: rewrite logic
+ Parameters such as frame size, etc, are variable. Pretty much
+ everything can change within a stream, so be prepared about it,
+ and do not cache parameters in the decoder.
+
+2011-11-15 23:00:32 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ opusdec: remove buffer pool, buffers are not constant size
+
+2011-11-15 19:53:33 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopus.c:
+ opusparse: add opusparse element
+ A very simple element that parses Opus streams from the ad hoc
+ framing used by the Opus test vectors.
+
2011-11-16 12:37:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/libvisual/visual.c:
Use the _check_reconfigure method instead of checking flags.
Don't need to ref the parent anymore, core does that.
+2011-11-15 17:49:48 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix pointer mismatch in memcpy on drain
+
2011-11-15 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
+2011-11-14 13:41:58 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opus: port to encoder/decoder base classes
+
2011-11-14 12:45:31 +0100 Robert Swain <robert.swain@gmail.com>
* gst-libs/gst/audio/gstaudiodecoder.c:
* tests/check/Makefile.am:
* tests/check/libs/.gitignore:
* tests/check/libs/audiocdsrc.c:
- * tests/check/libs/cddabasesrc.c:
* tests/check/libs/gstlibscpp.cc:
* tests/check/libs/libsabi.c:
* tests/check/libs/struct_arm.h:
* gst-libs/gst/audio/gstaudiocdsrc.c:
* gst-libs/gst/audio/gstaudiocdsrc.h:
* gst-libs/gst/cdda/Makefile.am:
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- * gst-libs/gst/cdda/gstcddabasesrc.h:
* gst-plugins-base.spec.in:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
Indent
Add padding
+2011-11-11 17:46:41 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ opus: port to 0.11
+
2011-11-11 18:23:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/fft/gstfftf32.c:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/rtp/gstbasertppayload.c:
- * gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/rtp/gstrtpbaseaudiopayload.c:
* gst-libs/gst/rtp/gstrtpbaseaudiopayload.h:
* gst-libs/gst/rtp/gstrtpbasedepayload.c:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
rename baseaudio* -> audiobase*
2011-11-11 11:52:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioringbuffer.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.c:
- * gst-libs/gst/audio/gstringbuffer.h:
rename files to match contained objects
2011-11-11 11:21:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
remove bogus files
They got somehow commited in 7012e88090e69339c60a4eb9449f7a7e39ca6aa3
+2011-11-11 10:39:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-11-10 23:02:35 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst/volume/gstvolume.c:
* tests/icles/audio-trickplay.c:
controller: port controller api changes
+2011-11-10 18:34:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-11-10 18:32:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/libvisual/visual.c:
* tests/check/libs/gstlibscpp.cc:
tests: fix build after removal of base64 lib
+2011-11-10 17:13:40 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix bandwidth property type mismatch
+
2011-11-10 17:52:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/video/gstvideosink.h:
pbutils: Fix introspection annotations
Fixes #663689
+2011-11-10 12:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-11-10 11:42:10 +0100 Edward Hervey <edward@collabora.com>
* tests/check/libs/struct_arm.h:
* gst/playback/gstsubtitleoverlay.c:
upates for new ACCEPT_CAPS query
+2011-11-09 12:24:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-09 12:19:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ gst/colorspace/colorspace.c
+
2011-11-09 12:11:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
gst/playback/gstplaysinkvideoconvert.c
gst/playback/gstplaysinkvideoconvert.h
+2011-10-05 18:25:58 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix latency query
+ This makes live 'audiosrc ! opusenc ! opusdec ! audiosink' pipelines
+ actually work without all audio being dumped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=660999
+
+2011-10-05 15:47:06 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: use debug level for debug info, not error
+ https://bugzilla.gnome.org/show_bug.cgi?id=660999
+
+2011-09-29 14:22:33 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: fix calculation of filler data size
+ https://bugzilla.gnome.org/show_bug.cgi?id=660469
+
2011-05-02 13:05:28 +0300 Felipe Contreras <felipe.contreras@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
Some found by Havard Graff.
Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
+2011-11-07 10:02:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-11-04 22:00:43 +0100 Stefan Sauer <ensonic@users.sf.net>
* gst/adder/gstadder.c:
* ext/ogg/gstoggdemux.c:
oggdemux: fix somtimes pad
+2011-11-04 11:01:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-11-04 10:48:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggmux.c:
* gst-libs/gst/video/video.h:
video: Add convenience macros for accessing GstVideoInfo flags
+2011-11-02 10:31:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-10-31 02:39:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/netbuffer/gstnetbuffer.c:
* ext/theora/gsttheoradec.c:
* gst-libs/gst/video/Makefile.am:
- * gst-libs/gst/video/gstmetavideo.c:
- * gst-libs/gst/video/gstmetavideo.h:
* gst-libs/gst/video/gstvideometa.c:
* gst-libs/gst/video/gstvideometa.h:
* gst-libs/gst/video/gstvideopool.h:
Update for pad API changes
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
+2011-10-31 14:51:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/master' into 0.11
+
2011-10-31 14:26:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
* gst/typefind/gsttypefindfunctions.c:
fix compilation
+2011-10-27 16:13:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-10-27 15:44:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
* tests/examples/audio/volume.c:
* tests/examples/volume/.gitignore:
* tests/examples/volume/Makefile.am:
- * tests/examples/volume/volume.c:
volume: move volume example to audio
2011-10-27 09:42:36 +0200 Stefan Sauer <ensonic@users.sf.net>
baseaudiosink: fix unused variable compiler warning if debugging in core is disabled
https://bugzilla.gnome.org/show_bug.cgi?id=660150
+2011-10-18 14:32:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-10-18 13:00:29 +0200 René Stadler <rene.stadler@collabora.co.uk>
* gst/playback/gstsubtitleoverlay.c:
* gst-libs/gst/audio/audio.c:
audio: Indent and doc fixes
+2011-10-16 15:28:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-10-13 08:53:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: handle empty input by discarding
+2011-10-08 11:17:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-10-08 11:05:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/vorbis/gstvorbisdec.c:
and falling back to a prefix check if nothing was found.
https://bugzilla.gnome.org/show_bug.cgi?id=657261
+2011-10-06 14:05:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-10-04 21:17:37 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst/encoding/gstencodebin.c:
The video-sink property allows manual specification via g_object_set ()
of the video sink element to be used.
+2011-10-04 13:29:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-10-03 15:20:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
a similar way to add other streams (eg, subtitles).
https://bugzilla.gnome.org/show_bug.cgi?id=642878
+2011-10-03 11:24:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-09-28 14:57:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: fix decoding
+ A simple ... opusenc ! opusdec ... pipeline now works.
+ https://bugzilla.gnome.org/show_bug.cgi?id=660364
+
+2011-09-28 14:56:18 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: moan if we get an unexpected amount of data
+ https://bugzilla.gnome.org/show_bug.cgi?id=660364
+
+2011-09-28 14:22:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opus: properly setup caps and init state from caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=660364
+
+2011-09-28 13:25:21 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusenc.c:
+ opusenc: use the same frame size setup as the opus test code
+ https://bugzilla.gnome.org/show_bug.cgi?id=660364
+
+2011-09-28 13:24:52 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ opusdec: opus supports a select set of sampling rates
+ https://bugzilla.gnome.org/show_bug.cgi?id=660364
+
+2011-09-28 13:24:21 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusenc.c:
+ opus: make it build against current, and remove cruft
+ https://bugzilla.gnome.org/show_bug.cgi?id=660364
+
2011-09-27 00:26:29 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/alsa/gstalsasrc.c:
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: minor docs fix
+2011-09-26 22:31:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-09-26 21:11:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioencoder.c:
audio: update audio format enums to match changes in 0.11
And add new audio format info stuff to docs.
+2011-09-06 16:13:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-09-06 15:40:02 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
* gst-libs/gst/audio/gstaudiodecoder.h:
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
- * gst-libs/gst/audio/gstbaseaudiodecoder.c:
- * gst-libs/gst/audio/gstbaseaudiodecoder.h:
- * gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
* win32/common/libgstaudio.def:
audio: rename GstBaseAudioDecoder/Encoder to GstAudioDecoder/Encoder
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: use base class fill method
+2011-08-25 12:49:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ ext/resindvd/rsnwrappedbuffer.c
+
2011-08-24 17:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggmux.c:
audioresample: fix build without orc
https://bugzilla.gnome.org/show_bug.cgi?id=656781
+2011-08-17 19:01:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
2011-08-17 17:24:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* tests/examples/overlay/.gitignore:
* tests/examples/overlay/Makefile.am:
* tests/examples/overlay/gtk-videooverlay.c:
- * tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/overlay/qt-videooverlay.cpp:
- * tests/examples/overlay/qt-xoverlay.cpp:
* tests/examples/overlay/qtgv-videooverlay.cpp:
* tests/examples/overlay/qtgv-videooverlay.h:
- * tests/examples/overlay/qtgv-xoverlay.cpp:
- * tests/examples/overlay/qtgv-xoverlay.h:
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
* tests/icles/.gitignore:
* tests/icles/Makefile.am:
* tests/icles/stress-videooverlay.c:
- * tests/icles/stress-xoverlay.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-videooverlay.c:
- * tests/icles/test-xoverlay.c:
tests: update for GstXOverlay => GstVideoOverlay
2011-08-08 10:44:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/videooverlay.c:
* gst-libs/gst/interfaces/videooverlay.h:
- * gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
* gst-plugins-base.spec.in:
interfaces: rename GstXOverlay interface to GstVideoOverlay
* gst-libs/gst/video/video.c:
video: improve debug
+2011-08-04 09:40:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-04 09:36:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ common
+ configure.ac
+ gst/colorspace/colorspace.c
+ gst/colorspace/colorspace.h
+ gst/colorspace/gstcolorspace.c
+
2011-08-03 14:14:55 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* gst/encoding/gstencodebin.c:
2011-07-07 21:24:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* Android.mk:
- * android/ffmpegcolorspace.mk:
* android/videoconvert.mk:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* ext/ogg/gstoggmux.c:
tests: remove tests from ancient times
They're just noise.
+2011-06-05 00:54:19 -0700 David Schleef <ds@schleef.org>
+
+ * ext/opus/Makefile.am:
+ * ext/opus/gstopus.c:
+ * ext/opus/gstopusdec.c:
+ * ext/opus/gstopusdec.h:
+ * ext/opus/gstopusenc.c:
+ * ext/opus/gstopusenc.h:
+ opus: duplicate from CELT
+ Copy the celt plugin and convert it to Opus. Mostly works.
+
2011-07-07 11:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
* configure.ac:
* gst/colorspace/Makefile.am:
- * gst/colorspace/colorspace.c:
- * gst/colorspace/colorspace.h:
- * gst/colorspace/colorspace.vcproj:
- * gst/colorspace/gstcolorspace.c:
- * gst/colorspace/gstcolorspace.h:
- * gst/colorspace/gstcolorspaceorc-dist.c:
- * gst/colorspace/gstcolorspaceorc-dist.h:
- * gst/colorspace/gstcolorspaceorc.orc:
* gst/videoconvert/Makefile.am:
* gst/videoconvert/gstvideoconvert.c:
* gst/videoconvert/gstvideoconvert.h:
* tests/check/elements/decodebin2.c:
* tests/check/elements/playbin-compressed.c:
* tests/check/elements/playbin.c:
- * tests/check/elements/playbin2-compressed.c:
* tests/check/elements/playbin2.c:
tests: fix up unit tests for playbin2/decodebin2 renames and updates
Even if they don't work yet.
* configure.ac:
* gst-libs/gst/audio/.gitignore:
* gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/testchannels.c:
* tests/examples/Makefile.am:
* tests/examples/audio/.gitignore:
* tests/examples/audio/Makefile.am:
2010-12-13 09:58:53 +0200 Stefan Kost <ensonic@users.sf.net>
- * docs/design-audiosinks.txt:
* docs/design/design-audiosinks.txt:
docs: move design doc to design folder
* docs/libs/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/descriptions.c:
- * gst-libs/gst/pbutils/gstdiscoverer-private.h:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/pbutils.h:
* gst/typefind/Makefile.am:
* gst/typefind/gstaacutil.c:
- * gst/typefind/gstaacutil.h:
* gst/typefind/gsttypefindfunctions.c:
* win32/common/libgstpbutils.def:
pbutils: add codec-specific utility functions for AAC
* configure.ac:
* tests/examples/Makefile.am:
- * tests/examples/playback/.gitignore:
- * tests/examples/playback/Makefile.am:
- * tests/examples/playback/decodetest.c:
- * tests/examples/playback/test.c:
- * tests/examples/playback/test2.c:
- * tests/examples/playback/test3.c:
- * tests/examples/playback/test4.c:
- * tests/examples/playback/test5.c:
- * tests/examples/playback/test6.c:
- * tests/examples/playback/test7.c:
* tests/icles/Makefile.am:
* tests/icles/playback/.gitignore:
* tests/icles/playback/Makefile.am:
* configure.ac:
* gst/playback/.gitignore:
* gst/playback/Makefile.am:
- * gst/playback/decodetest.c:
- * gst/playback/test.c:
- * gst/playback/test2.c:
- * gst/playback/test3.c:
- * gst/playback/test4.c:
- * gst/playback/test5.c:
- * gst/playback/test6.c:
- * gst/playback/test7.c:
* tests/examples/Makefile.am:
* tests/examples/playback/.gitignore:
* tests/examples/playback/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/tag/lang.c:
* gst/ffmpegcolorspace/Makefile.am:
- * gst/ffmpegcolorspace/gstffmpeg.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraparse.c:
- * ext/theora/theora.c:
- * ext/theora/theoradec.c:
- * ext/theora/theoraenc.c:
- * ext/theora/theoraparse.c:
theora: Rename source files to have the same name as the headers
2010-01-14 10:07:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisparse.c:
* ext/vorbis/gstvorbistag.c:
- * ext/vorbis/vorbis.c:
- * ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisenc.c:
- * ext/vorbis/vorbisparse.c:
- * ext/vorbis/vorbistag.c:
vorbis: Rename source files to have the same name as the headers
2010-01-14 10:05:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2010-01-07 15:26:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/tag/lang-tables.c:
* gst-libs/gst/tag/lang-tables.dat:
* gst-libs/gst/tag/lang.c:
tag: fix up disting of lang-tables.c more correctly
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp-marshal.list:
* gst-libs/gst/rtsp/gstrtspextension.c:
- * gst-libs/gst/rtsp/rtsp-marshal.list:
* gst-libs/gst/video/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
* ext/vorbis/gstvorbistag.h:
* ext/vorbis/vorbis.c:
* ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisdec.h:
* ext/vorbis/vorbisenc.c:
- * ext/vorbis/vorbisenc.h:
* ext/vorbis/vorbisparse.c:
- * ext/vorbis/vorbisparse.h:
* ext/vorbis/vorbistag.c:
- * ext/vorbis/vorbistag.h:
vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audioresample/speex_resampler_int.c:
* gst/audioresample/speex_resampler_wrapper.h:
* gst/speexresample/Makefile.am:
- * gst/speexresample/README:
- * gst/speexresample/arch.h:
- * gst/speexresample/fixed_arm4.h:
- * gst/speexresample/fixed_arm5e.h:
- * gst/speexresample/fixed_bfin.h:
- * gst/speexresample/fixed_debug.h:
- * gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
- * gst/speexresample/resample_sse.h:
- * gst/speexresample/speex_resampler.h:
- * gst/speexresample/speex_resampler_double.c:
- * gst/speexresample/speex_resampler_float.c:
- * gst/speexresample/speex_resampler_int.c:
- * gst/speexresample/speex_resampler_wrapper.h:
* gst/typefind/gsttypefindfunctions.c:
* tests/check/Makefile.am:
* tests/check/elements/audioresample.c:
Original commit message from CVS:
releasing 0.10.0
+2001-12-17 18:37:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ building up speed
+ Original commit message from CVS:
+ building up speed
+
projects / platform / upstream / gst-plugins-base.git / blobdiff