Commit 48f584e6 authored by Sebastian Dröge's avatar Sebastian Dröge

Release 1.7.90

parent 4e076fa2
=== release 1.7.90 ===
2016-03-01 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 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 Sebastian Dröge <slomo@coaxion.net>
2016-02-19 11:48:30 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
releasing 1.7.2
* 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>
......@@ -181,6 +548,35 @@
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:
......@@ -1022,6 +1418,16 @@
* 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:
......@@ -1426,6 +1832,12 @@
* 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:
......@@ -1468,6 +1880,14 @@
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:
......@@ -1701,6 +2121,14 @@
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:
......@@ -1738,6 +2166,86 @@
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:
......@@ -1942,6 +2450,17 @@
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:
......@@ -2740,6 +3259,15 @@
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:
......@@ -2760,6 +3288,18 @@
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:
......@@ -3119,6 +3659,14 @@
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:
......@@ -3219,6 +3767,15 @@
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:
......@@ -3544,6 +4101,13 @@
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:
......@@ -3686,6 +4250,19 @@
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:
......@@ -4392,6 +4969,12 @@
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:
......@@ -5052,6 +5635,17 @@
* 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:
......@@ -5075,6 +5669,13 @@
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:
......@@ -5718,6 +6319,14 @@
* 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:
......@@ -5795,6 +6404,22 @@
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: