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authorSebastian Dröge <sebastian@centricular.com>2016-07-06 13:06:06 +0300
committerSebastian Dröge <sebastian@centricular.com>2016-07-06 13:06:06 +0300
commit08f993d090b9b5c761dcaaf63f6286c6f114c6d4 (patch)
tree97bc4bd0718aa96bcc57c91c849ad056465bb754 /ChangeLog
parent49c644ce25cb4cf96725707d9e12e47c7adad678 (diff)
downloadgstreamer-plugins-base-08f993d090b9b5c761dcaaf63f6286c6f114c6d4.tar.gz
Release 1.9.11.9.1
Diffstat (limited to 'ChangeLog')
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diff --git a/ChangeLog b/ChangeLog
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@@ -1,9 +1,2074 @@
+=== release 1.9.1 ===
+
+2016-07-06 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.9.1
+
+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-core.h:
+ * 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 Sebastian Dröge <slomo@coaxion.net>
+2016-03-24 12:19:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.8.0
+ * 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>
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