+=== 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 Sebastian Dröge <slomo@coaxion.net>
+2015-09-18 19:20:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.5.91
+ * 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>
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:
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:
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:
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:
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:
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:
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:
* 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:
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:
* 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:
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:
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:
{ 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:
* 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:
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:
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:
* 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:
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:
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:
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:
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:
2014-10-30 11:43:52 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/video/Makefile.am:
- * gst-libs/gst/video/resampler.c:
- * gst-libs/gst/video/resampler.h:
* 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/Makefile.am:
* gst-libs/gst/video/video-converter.c:
* gst-libs/gst/video/video-converter.h:
- * gst-libs/gst/video/video-convertor.c:
- * gst-libs/gst/video/video-convertor.h:
* gst-libs/gst/video/video.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.c:
* gst/videoconvert/videoconvert.h:
* tests/check/Makefile.am:
* win32/common/libgstvideo.def:
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:
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:
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:
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:
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:
* 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:
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:
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:
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:
* 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:
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:
* po/zh_CN.po:
po: update for new translatable strings
+2012-04-06 14:52:12 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ 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