mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-27 02:30:35 +00:00
Code Issues 1 Projects Releases Wiki Activity

Release 1.15.2

Browse source
This commit is contained in:
Tim-Philipp Müller 2019-02-26 11:47:29 +00:00
parent 1ef240e24d
commit 48b46e3f14
6 changed files with 790 additions and 102 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

563
ChangeLog
View file

@ -1,3 +1,566 @@
=== release 1.15.2 ===
2019-02-26 11:47:29 +0000 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-good.doap:
* meson.build:
Release 1.15.2
2019-02-26 11:47:29 +0000 Tim-Philipp Müller <tim@centricular.com>
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-audioparsers.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dtmf.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.xml:
* docs/plugins/inspect/plugin-gtk.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-imagefreeze.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-isomp4.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-lame.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mpg123.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multifile.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-qmlgl.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shapewipe.xml:
* docs/plugins/inspect/plugin-shout2.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-soup.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-taglib.xml:
* docs/plugins/inspect/plugin-twolame.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-vpx.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
Update docs
2019-02-26 11:47:25 +0000 Tim-Philipp Müller <tim@centricular.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/eo.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/fur.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/mt.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/zh_HK.po:
* po/zh_TW.po:
Update translations
2019-02-25 19:08:08 +1100 Matthew Waters <matthew@centricular.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2dec: also remove the colorimetry and chroma-site fields
If a different format is chosen, then these values are incorrect.
2019-02-22 16:02:12 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtpmanager/gstrtpsession.c:
rtpsession: Fix EOS forwarding
So far we assumed that if all sources are bye, this meant we needed to
send an EOS on the RTCP sink. The problem is that this case may happens
if we only had one internal source and it detected a collision.
So now we limit the EOS forwarding to when there is a send_rtp_sink pad
and that this pad has received EOS. We don'tcheck the recv_rtp_sink
since the code does not wait for the bye to be send before sending EOS
to the RTCP src pad.
2019-02-25 01:12:56 +1100 Jan Schmidt <jan@centricular.com>
* gst/wavparse/gstwavparse.c:
wavparse: Declare support for RF64
RF64 encode support was added to wavenc quite some time
ago, but not declared in wavparse. It seems wavparse can
decode it though, so add it to the sink pad.
The RF64 support was added in
https://bugzilla.gnome.org/show_bug.cgi?id=735627
2019-02-12 18:28:40 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
rtp: Add property to disable RTCP reports per internal rtpsource
This is useful when implementing custom retransmission mechanism like
RIST to prevent RTCP from being produces for the retransmitted SSRC.
This would also be used in general for various purpose when customizing
an RTP base pipeline.
2019-02-12 18:26:21 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtpmanager/rtpsession.c:
rtpsession: Emit on-new-sender-ssrc for RTX ssrc also
2019-01-15 18:04:09 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/elements/rtpjitterbuffer.c:
rtp jitterbuffer test: Test for queue filling
2019-01-11 17:53:43 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
rtpjitterbuffer: Limit size to 2^15 packets
If it goes over 2^15 packets, it will think it has rolled over
and start dropping all packets. So make sure the seqnum distance is not too big.
But let's not limit it to a number that is too small to avoid emptying it
needlessly if there is a spurious huge sequence number, let's allow at
least 10k packets in any case.
2019-02-11 11:33:32 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: There is no automatic reorder threshold
2019-01-30 10:47:49 -0300 Thibault Saunier <tsaunier@igalia.com>
* ext/pulse/pulsedeviceprovider.c:
pulse: Post DEVICE_CHANGED on modification
2018-11-26 13:48:56 -0300 Thibault Saunier <tsaunier@igalia.com>
* ext/pulse/pulsedeviceprovider.c:
* ext/pulse/pulsedeviceprovider.h:
pulse: Mark default devices as "default"
2019-02-08 16:10:25 +0000 Ilya Smelykh <ilya.smelykh@gmail.com>
* gst/flv/gstflvmux.c:
flvmux: Use 8kHz sample rate for alaw/mulaw audio
2019-02-07 09:54:31 +0000 Ilya Smelykh <ilya.smelykh@gmail.com>
* gst/flv/gstflvdemux.c:
flvdemux: set sample rate to 8KHz for G.711 audio
2019-02-08 13:59:19 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
* gst/isomp4/gstqtmux.c:
qtmux: Only write timecode trak for video
Recent changes in ccextractor were attaching timecode meta to the closed
caption track. We shouldn't write timecode information for the closed
caption trak.
2019-02-05 22:14:18 +0100 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
* configure.ac:
* ext/vpx/gstvpxdec.c:
* ext/vpx/meson.build:
vpx: Fix build against libvpx 1.8
The deprecated debug visualizer was removed.
2019-02-05 19:41:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* tests/check/elements/rtph264.c:
* tests/check/elements/rtph265.c:
* tests/check/elements/rtpulpfec.c:
misc: Fix warnings on Cerbero's mingw (gcc 4.7)
error: this decimal constant is unsigned only in ISO C90 [-Werror]
2019-02-06 14:43:18 +0530 Arun Raghavan <arunsr@codeaurora.org>
* ext/pulse/pulsesink.c:
pulsesink: Deal with not being able to convert a format to caps
It is possible that PulseAudio adds formats that are not yet supported
in pulsesink, and in those cases, we want to gracefully skip them rather
than cause an assert on a NULL caps.
2019-01-17 09:22:18 +0100 Edward Hervey <edward@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Remove trailing '\n' in debug
2019-02-05 15:27:49 +1100 Matthew Waters <matthew@centricular.com>
* ext/qt/gstqtgl.h:
qmlgl: Fix opengl header guard changes again
Reapply 3d708a5bfa8961cc37671bc3226976dfc9ba50ad in the correct place
after the iOS additions.
2019-02-02 02:29:10 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: use the correct segment seqnum
2019-02-02 02:26:47 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: use the correct segment seqnum
2019-02-02 02:24:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpsession.h:
rtpsession: use the correct segment seqnum
2019-01-26 10:35:31 -0300 Thibault Saunier <tsaunier@igalia.com>
* gst/flv/gstflvdemux.c:
flvdemux: Do not error out if the first added and chained pad is not linked
And let it the oportunity to get its other pad linked
Example:
```
$ gst-launch-1.0 uridecodebin uri=file:///home/thiblahute/gst-validate.save/gst-integration-testsuites/testsuites/../medias/defaults/flv/819290236.flv caps=audio/x-raw expose-all-streams=FALSE ! fakesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: Internal data stream error.
Additional debug info:
../subprojects/gst-plugins-good/gst/flv/gstflvdemux.c(2760): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0:
streaming stopped, reason not-linked (-1)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
```
2019-01-16 23:54:25 -0800 Christopher Snowhill <kode54@gmail.com>
* gst/matroska/webm-mux.c:
webmmux: allow resolutions above 4096
Modify the caps string to allow width and height greater than 4096.
There is no need to restrict it since the matroska format allows the
width and height values to be up to eight bytes long, and this also
applies to the webm subset of the format.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/550
2019-02-01 14:27:11 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* ext/qt/meson.build:
meson: qmlgl plugin iOS definitions
Tested with cross-ios-arm64 and cross-ios-x86, since those two are the
only archs shipped with the official Qt binaries.
2019-02-01 14:27:11 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* ext/qt/gstqtgl.h:
qt: Don't define GLsync inside gstglfuncs.h
This was originally added for fixing conflicting definitions between
Android and Qt, but times have changed and now this breaks the build
on iOS:
[...]/OpenGLES.framework/Headers/ES3/gl.h:1006:48: error: unknown type name 'gst_qt_GLsync'
GL_API void GL_APIENTRY glGetSynciv (GLsync sync, GLenum pname, GLsizei bufSize, GLsizei* length, GLint* values) OPENGLES_DEPRECATED(ios(7.0, 12.0), tvos(9.0, 12.0));
^
../ext/qt/gstqtgl.h:49:16: note: expanded from macro 'GLsync'
#define GLsync gst_qt_GLsync
^
6 errors generated.
Instead, we simply avoid defining GLsync ourselves if we're using Qt.
2019-02-01 14:27:11 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* ext/qt/meson.build:
meson: Fix indentation in qt plugin and add a FIXME comment
2019-01-26 21:02:27 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtp/gstrtph265depay.c:
rtph265depay; Fix handling of marker on aggregated packet
When multiple nals are aggrgated, the marker bit should be associated only
with the last NAL of the packet. Otherwise we may break rendering in with
AU alignment.
2019-01-26 21:01:08 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtp/gstrtph264depay.c:
rtph264depay: Fix handling or marker on STAP-A
Only forward the marker for the last NAL of the STAP-A. Otherwise each NAL
endup being assumed to be a full frame which may break rendering.
Fixes 557
2019-01-27 09:19:00 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* tests/check/elements/rtph265.c:
test: h265depay: Add todo for testing aggregate packets with marker
We are missing a sample to test this, but a fix has been made, so add a
todo.
2019-01-26 20:42:40 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* tests/check/elements/rtph264.c:
test: rtph264depay: Check handling of STAP-A marker
Related to #557
2019-01-31 15:23:43 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* tests/check/meson.build:
meson: orc-test is not required
This is especially never available on iOS.
2019-01-30 19:44:01 +0900 Seungha Yang <seungha.yang@navercorp.com>
* meson.build:
* tests/check/meson.build:
meson: Add support orc fallback
Allow fallback to orc subproject if any.
Additionally 'dependencies' keyword is removed from find_library,
because it's invalid keyword for find_library.
2019-01-17 21:06:54 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* ext/gdk_pixbuf/gstgdkpixbufdec.c:
gdkpixbufdec: always output a TIME segment
It makes no sense for a decoder to output a BYTES segment, and
many elements one would plug downstream of a video decoder assume
the segments they receive are in TIME format, for example this fixes:
gst-validate-1.0 filesrc location=opacity01.svg ! gdkpixbufdec ! \
videobalance ! videoconvert ! fakesink
In that case, videobalance was emitting an assertion when trying to
call gst_object_sync_values()
2019-01-29 12:12:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* ext/qt/meson.build:
meson: Add macOS definitions for qmlgl plugin
Tested with Cerbero.
2019-01-29 12:12:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* ext/qt/meson.build:
meson: Fix building of qmlgl plugin on Android
Needs gnustl for C++ STL support, which is the GNU STL on Android API
19 and older, and is a wrapper for the llvm-libc++ STL on newer APIs.
QtGui C++ templates use GL functions, so GLESv2 is needed at link time
2019-01-24 16:21:12 -0500 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/interleave/deinterleave.c:
deinterleave: Allow switching between 1 channel configs
regardless of whether they're positioned, since positioning
with a 1 channel stream doesn't change anything.
2019-01-22 11:45:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* configure.ac:
configure.ac: Fix Qt Android integration
The Qt Android integration is now signalled with HAVE_QT_ANDROID
See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/86
2018-12-18 14:46:25 -0500 Xavier Claessens <xavier.claessens@collabora.com>
* ext/soup/meson.build:
Meson: fallback to libsoup subproject
2019-01-22 12:52:25 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: detect opengl api from -base .pc files correctly
There was a mismatch between the .pc files generated by
autotools and by meson that would lead to meson not detecting
that opengl api is available even though it is, if -base was
built with autotools. The mismatch has now been rectified in
-base, so we need to update for that.
This is mostly for consistency, this problem didn't seem
to affect anything in -good.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/871
2019-01-22 09:51:33 +0000 Tim-Philipp Müller <tim@centricular.com>
* sys/ximage/meson.build:
meson: ximage: check for XShmAttach()
Fixes FIXME.
2019-01-22 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson_options.txt:
* sys/ximage/meson.build:
meson: add options for ximagesrc xshm, xfixes, xdamage checks
And rename x11 option to ximagesrc.
Fixes #553
2019-01-21 11:53:53 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
* ext/qt/README.md:
qmlgl: add README.md with information on building for non-linux platforms with qmake
2019-01-19 15:46:41 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
* ext/qt/meson.build:
qmlgl: meson: fix theoretical support for building for android
The android code path is slightly different than the EGLFS one,
so I added previously a HAVE_QT_ANDROID define for use with qmake.
Here I also add it in meson, although I expect nobody will ever use
meson to build this, as it's complicated.
2019-01-19 15:37:45 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
* ext/qt/qtplugin.pro:
qmlgl: qmake: add support for MacOS target
2019-01-19 15:21:43 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
* ext/qt/qtplugin.pro:
qmlgl: qmake: remove cerbero's include dir from the include path
pkg-config should do it's job here, this is unnecessary and implies using cerbero
2019-01-19 15:19:26 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
* ext/qt/gstqtgl.h:
* ext/qt/qtplugin.pro:
qmlgl: qualify Qt includes with their module and remove module include dir from the .pro file
it is perfectly legal to use the <module/class> style of includes with Qt
and it avoids the need for having the module's include dir in the include path
2019-01-19 15:10:09 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
* ext/qt/qtplugin.pro:
qmlgl: qmake: don't link against QtWidgets, it's not used
2019-01-19 15:07:44 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
* ext/qt/gstqtglutility.cc:
* ext/qt/qtplugin.pro:
qmlgl: qmake: fix building for android
2019-01-19 02:39:32 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* ext/qt/meson.build:
meson: Generate pkg-config file for qmlgl plugin
2019-01-17 16:26:56 +0100 Victor Toso <me@victortoso.com>
* tests/check/elements/rtp-payloading.c:
tests: rtp-payloading avoid -Wmaybe-uninitialized
More false positives as both of them are initialized in the line
before they are used, wrapped with fail_unless() check.
2019-01-17 16:19:40 +0100 Victor Toso <me@victortoso.com>
* tests/check/elements/matroskamux.c:
tests: matroskamux avoid -Wmaybe-uninitialized
False positive for the three variables but some warnings like:
../tests/check/elements/matroskamux.c:875:10:
warning: 'chapters_offset' may be used uninitialized in this function [-Wmaybe-uninitialized]
*index = chapters_offset;
~~~~~~~^~~~~~~~~~~~~~~~~
The above is false positive as there is a gboolean to check if it was
initialized or not (found_chapters_declaration).
2018-05-28 14:39:53 +0530 Arun Raghavan <arunsr@codeaurora.org>
* ext/pulse/pulseutil.c:
pulse: Fix format info to caps conversion for PCM
2019-01-18 12:27:34 +0530 Arun Raghavan <arunsr@codeaurora.org>
* ext/pulse/pulsesink.c:
* ext/pulse/pulsesrc.c:
* ext/pulse/pulsesrc.h:
* ext/pulse/pulseutil.c:
* ext/pulse/pulseutil.h:
Revert "pulsesrc: Move to extended stream API"
This reverts commit 4d67d1bd16bcf25acf89d8acd952badcd5b9a657.
Using the extended API for the capture path depends on a fix in
PulseAudio
(https://gitlab.freedesktop.org/pulseaudio/pulseaudio/merge_requests/49).
Until then, let's go back to the standard API.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/552
2019-01-18 14:41:14 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* ext/qt/meson.build:
meson: Search for qmake-qt5 before qmake
The canonical name for the binary is qmake-qt5, and qmake is the
generic name that can also be a qt4 qmake.
2019-01-17 15:30:25 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
* sys/v4l2/gstv4l2.c:
v4l2: mark caps from probe as MAY_BE_LEAKED
2019-01-15 18:06:11 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
* sys/v4l2/gstv4l2transform.c:
v4l2transform: fix cdata caps leaks
The cdata structure was freed but not its caps.
It was already done in gst_v4l2_video_dec_subclass_init() and
gst_v4l2_video_enc_subclass_init().
=== release 1.15.1 ===
2019-01-17 01:59:28 +0000 Tim-Philipp Müller <tim@centricular.com>

307
NEWS
View file

@ -15,7 +15,7 @@ the git master branch and which will eventually result in 1.16.
See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
version of this document.
_Last updated: Monday 14 January 2019, 13:00 UTC (log)_
_Last updated: Monday 25 January 2019, 15:00 UTC (log)_
Introduction
@ -24,8 +24,8 @@ The GStreamer team is proud to announce a new major feature release in
the stable 1.x API series of your favourite cross-platform multimedia
framework!
As always, this release is again packed with new features, bug fixes and
other improvements.
As always, this release is again packed with many new features, bug
fixes and other improvements.
Highlights
@ -40,7 +40,7 @@ Highlights
- Support for Closed Captions and other Ancillary Data in video
- Spport for planar (non-interleaved) raw audio
- Support for planar (non-interleaved) raw audio
- GstVideoAggregator, compositor and OpenGL mixer elements are now in
-base
@ -98,14 +98,17 @@ Noteworthy new API
to process the media in a live pipeline before it reaches the sink.
This is on top of the systemic latency that is normally reported by
the latency query. This defaults to 20ms and should make pipelines
such as “v4lsrc ! xvimagesink” not claim that all frames are late in
the QoS events. Ideally, this should replace max_lateness for most
applications.
such as v4l2src ! xvimagesink not claim that all frames are late in
the QoS events. Ideally, this should replace the "max-lateness"
property for most applications.
- RTCP Extended Reports (XR) parsing according to RFC 3611:
Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time,
Delay since the last Receiver (DLRR), Statistics Summary, and VoIP
Metrics reports.
Metrics reports. This only provides the ability to parse such
packets, generation of XR packets is not supported yet and XR
packets are not automatically parsed by rtpbin / rtpsession but must
be actively handled by the application.
- a new mode for interlaced video was added where each buffer carries
a single field of interlaced video, with buffer flags indicating
@ -146,9 +149,10 @@ or planar arrangement in memory would look like
|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory
chunks or separated by some padding.
GStreamer has always had signalling for non-interleaved audio, but it
was never actually properly implemented in any elements. audioconvert
would advertise support for it, but wasn’t actually able to handle it.
GStreamer has always had signalling for non-interleaved audio since
version 1.0, but it was never actually properly implemented in any
elements. audioconvert would advertise support for it, but wasn’t
actually able to handle it correctly.
With this release we now have full support for non-interleaved audio as
well, which means more efficient integration with external APIs that
@ -177,18 +181,18 @@ The video support library has gained support for detecting and
extracting Ancillary Data from videos as per the SMPTE S291M
specification, including:
- a VBI (Video Blanking Interval) parser that can detect and extract
Ancillary Data from Vertical Blanking Interval lines of component
signals. This is currently supported for videos in v210 and UYVY
format.
- a VBI (Vertical Blanking Interval) parser that can detect and
extract Ancillary Data from Vertical Blanking Interval lines of
component signals. This is currently supported for videos in v210
and UYVY format.
- a new GstMeta for closed captions: GstVideoCaptionMeta. This
supports the two types of closed captions, CEA-608 and CEA-708,
along with the four different ways they can be transported (other
systems are a superset of those).
- a VBI (Video Blanking Interval) encoder for writing ancillary data
to the Vertical Blanking Interval lines of component signals.
- a VBI (Vertical Blanking Interval) encoder for writing ancillary
data to the Vertical Blanking Interval lines of component signals.
The new closedcaption plugin in gst-plugins-bad then makes use of all
this new infrastructure and provides the following elements:
@ -222,6 +226,9 @@ support:
- playbin and playbin3 learned how to autoplug CEA 608/708 CC overlay
elements
- the externally maintained ajavideosrc element for AJA capture cards
has support for extracting closed captions
The rsclosedcaption plugin in the Rust plugins collection includes a
MacCaption (MCC) file parser and encoder.
@ -239,7 +246,7 @@ New Elements
- gloverlaycompositor: New OpenGL-based compositor element that
flattens any overlays from GstVideoOverlayCompositionMetas into the
video stream.
video stream. This element is also always part of glimagesink.
- glalpha: New element that adds an alpha channel to a video stream.
The values of the alpha channel can either be set to a constant or
@ -248,7 +255,7 @@ New Elements
done in floating point so results may not be identical to the output
of the existing alpha element.
- rtpfunnel funnels together rtp-streams into a single session. Use
- rtpfunnel funnels together RTP streams into a single session. Use
cases include multiplexing and bundle. webrtcbin uses it to
implement BUNDLE support.
@ -264,10 +271,12 @@ New Elements
WPE
- Two new OpenCV-based elements: cameracalibrate and cameraundistort
who can communicate to figure out distortion correction parameters
that can communicate to figure out distortion correction parameters
for a camera and correct for the distortion.
- new sctp plugin based on usrsctp with sctpenc and sctpdec elements
- New sctp plugin based on usrsctp with sctpenc and sctpdec elements.
These elements are used inside webrtcbin for implementing data
channels.
New element features and additions
@ -348,12 +357,12 @@ New element features and additions
- rtspsrc now allows applications to send RTSP SET_PARAMETER and
GET_PARAMETER requests using action signals.
- rtspsrc also has a small (100ms) configurable teardown delay by
default to try and make sure an RTSP TEARDOWN request gets sent out
when the source element shuts down. This will block the downward
PAUSED to READY state change for a short time, but can be unset
where it’s a problem. Some servers only allow a limited number of
concurren clients, so if no proper TEARDOWN is sent clients may have
- rtspsrc has a small (100ms) configurable teardown delay by default
to try and make sure an RTSP TEARDOWN request gets sent out when the
source element shuts down. This will block the downward PAUSED to
READY state change for a short time, but can be disabled where it’s
a problem. Some servers only allow a limited number of concurrent
clients, so if no proper TEARDOWN is sent new clients may have
problems connecting to the server for a while.
- souphttpsrc behaves better with low bitrate streams now. Before it
@ -364,6 +373,7 @@ New element features and additions
- filesink: do internal buffering to avoid performance regression with
small writes since we bypass libc buffering by using writev()
instead of fwrite()
- identity: add "eos-after" property and fix "error-after" property
when the element is reused
@ -405,9 +415,9 @@ New element features and additions
relays (TURN servers).
- The removesilence element has received various new features and
properties, such as a
"threshold"1 property, detecting silence only after minimum silence time/buffers, a“silent”property to control bus message notifications as well as a“squash”`
property.
properties, such as a "threshold" property, detecting silence only
after minimum silence time/buffers, a "silent" property to control
bus message notifications as well as a "squash" property.
- AOMedia AV1 decoder gained support for 10/12bit decoding whilst the
AV1 encoder supports more image formats and subsamplings now and
@ -430,15 +440,15 @@ Plugin and library moves
- The stereo element was moved from -bad into the existing audiofx
plugin in -good. If you get duplicate type registration warnings
when upgrading, check that you don’t have a stale gststereo plugin
lying about somewhere.
when upgrading, check that you don’t have a stale stereoplugin lying
about somewhere.
GstVideoAggregator, compositor, and OpenGL mixer elements moved from -bad to -base
GstVideoAggregator is a new base class for raw video mixers and muxers
and is based on [GstAggregator][aggregator]. It provides defined-latency
mixing of raw video inputs and ensures that the pipeline won’t stall
even if one of the input streams stops producing data.
and is based on GstAggregator. It provides defined-latency mixing of raw
video inputs and ensures that the pipeline won’t stall even if one of
the input streams stops producing data.
As part of the move to stabilise the API there were some last-minute API
changes and clean-ups, but those should mostly affect internal elements.
@ -456,14 +466,15 @@ would expected in most scenarios.
The compositor element has gained support for per-pad blending mode
operators (SOURCE, OVER, ADD) which determines what operator to use for
blending this pad over the previous ones. This can be used to implement
crossfading.
crossfading and the available operators can be extended in the future as
needed.
A number of OpenGL-based video mixer elements (glvideomixer, glmixerbin,
glvideomixerelement, glstereomix, glmosaic) which are built on top of
GstVideoAggregator have also been moved from -bad to -base now. These
elements have been merged into the existing OpenGL plugin, so if you get
duplicate type registration warnings when upgrading, check that you
don’t have a stale gstopenglmixers plugin lying about somewhere.
don’t have a stale openglmixers plugin lying about somewhere.
Plugin removals
@ -477,11 +488,11 @@ The following plugins have been removed from gst-plugins-bad:
plugin.
- The acmmp3dec and acmenc plugins for Windows have been removed. ACM
is an ancient legacy API and there was no point in keeping them
around for a licensed mp3 decoder now that mp3 patents have expired
and we have a decoder in -good. We also didn’t ship these in our
cerbero-built Windows packages, so it’s unlikely that they’ll be
missed.
is an ancient legacy API and there was no point in keeping the
plugins around for a licensed MP3 decoder now that the MP3 patents
have expired and we have a decoder in -good. We also didn’t ship
these in our cerbero-built Windows packages, so it’s unlikely that
they’ll be missed.
Miscellaneous API additions
@ -506,7 +517,8 @@ Miscellaneous API additions
one might need to put such elements into READY state to test if the
hardware is present in the system for example.
- protection: Add a new definition for unspecified system protection
- protection: Add a new definition for unspecified system protection,
GST_PROTECTION_UNSPECIFIED_SYSTEM_ID
- take functions for various mini objects that didn’t have them yet:
gst_query_take(), gst_message_take(), gst_tag_list_take(),
@ -522,11 +534,18 @@ Miscellaneous API additions
gst_clear_mini_object(), gst_clear_object()
- miniobject: new API gst_mini_object_add_parent() and
gst_mini_object_remove_parent()to set parent pointers on mini objects to ensure correct writability: Every container of miniobjects now needs to store itself as parent in the child object, and remove itself again later. A mini object is then only writable if there is at most one parent, that parent is writable itself, and the reference count of the mini object is 1.GstBuffer(for memories),GstBufferList(for buffers),GstSample(for caps, buffer, bufferlist), andGstVideoOverlayComposition`
were updated accordingly. Without this it was possible to have
e.g. a buffer list with a refcount of 2 used in two places at once
that both modify the same buffer with refcount 1 at the same time
wrongly thinking it is writable even though it’s really not.
gst_mini_object_remove_parent() to set parent pointers on mini
objects to ensure correct writability: Every container of
miniobjects now needs to store itself as parent in the child object,
and remove itself again later. A mini object is then only writable
if there is at most one parent, that parent is writable itself, and
the reference count of the mini object is 1. GstBuffer (for
memories), GstBufferList (for buffers), GstSample (for caps, buffer,
bufferlist), and GstVideoOverlayComposition were updated
accordingly. Without this it was possible to have e.g. a buffer list
with a refcount of 2 used in two places at once that both modify the
same buffer with refcount 1 at the same time wrongly thinking it is
writable even though it’s really not.
- poll: add API to watch for POLLPRI and stop treating POLLPRI as a
read. This is useful to wait for video4linux events which are
@ -596,7 +615,7 @@ GstPlayer
Miscellaneous changes
- As a result of moving to different FFmpeg APIs, encoder and decoder
- As a result of moving to newer FFmpeg APIs, encoder and decoder
elements exposed by the GStreamer FFmpeg wrapper plugin (gst-libav)
may have seen possibly incompatible changes to property names and/or
types, and not all properties exposed might be functional. We are
@ -649,48 +668,48 @@ Tracing framework and debugging improvements
object. This is currently limited to pads for GstElements and
events for the pads. The output may look like this:
(gdb) gst-print pad.object.parent
GstMatroskaDemux (matroskademux0) {
SinkPad (sink, pull) {
(gdb) gst-print pad.object.parent
GstMatroskaDemux (matroskademux0) {
SinkPad (sink, pull) {
}
SrcPad (video_0, push) {
events:
stream-start:
stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/001:1274058367
caps: video/x-theora
width: 1920
height: 800
pixel-aspect-ratio: 1/1
framerate: 24/1
streamheader: < 0x5555557c7d30 [GstBuffer], 0x5555557c7e40 [GstBuffer], 0x7fffe00141d0 [GstBuffer] >
segment: time
rate: 1
tag: global
container-format: Matroska
}
SrcPad (audio_0, push) {
events:
stream-start:
stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/002:1551204875
caps: audio/mpeg
mpegversion: 4
framed: true
stream-format: raw
codec_data: 0x7fffe0014500 [GstBuffer]
level: 2
base-profile: lc
profile: lc
channels: 2
rate: 44100
segment: time
rate: 1
tag: global
container-format: Matroska
tag: stream
audio-codec: MPEG-4 AAC audio
language-code: en
}
}
SrcPad (video_0, push) {
events:
stream-start:
stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/001:1274058367
caps: video/x-theora
width: 1920
height: 800
pixel-aspect-ratio: 1/1
framerate: 24/1
streamheader: < 0x5555557c7d30 [GstBuffer], 0x5555557c7e40 [GstBuffer], 0x7fffe00141d0 [GstBuffer] >
segment: time
rate: 1
tag: global
container-format: Matroska
}
SrcPad (audio_0, push) {
events:
stream-start:
stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/002:1551204875
caps: audio/mpeg
mpegversion: 4
framed: true
stream-format: raw
codec_data: 0x7fffe0014500 [GstBuffer]
level: 2
base-profile: lc
profile: lc
channels: 2
rate: 44100
segment: time
rate: 1
tag: global
container-format: Matroska
tag: stream
audio-codec: MPEG-4 AAC audio
language-code: en
}
}
- gst_structure_to_string() now serialises the actual value of
pointers when serialising GstStructures instead of claiming they’re
@ -704,10 +723,10 @@ Tools
- gst-inspect-1.0 has coloured output now and will automatically use a
pager if the output does not fit on a page. This only works in a
unix environment and if the output is not piped. If you don’t like
the colours you can disable them by setting the
GST_INSPECT_NO_COLORS=1 environment variable or passing the
--no-colors command line option.
UNIX environment and if the output is not piped, and on Windows 10
build 16257 or newer. If you don’t like the colours you can disable
them by setting the GST_INSPECT_NO_COLORS=1 environment variable or
passing the --no-color command line option.
GStreamer RTSP server
@ -736,6 +755,29 @@ GStreamer VAAPI
- this section will be filled in in due course
GStreamer OMX
- Add support of NV16 format to video encoders input.
- Video decoders now handle the ALLOCATION query to tell upstream
about the number of buffers they require. Video encoders will also
use this query to adjust their number of allocated buffers
preventing starvation when using dynamic buffer mode.
- The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE
and can now be used to track a widder variety of interactions
between OMX and GStreamer.
- Video encoders will now detect frame rate only changes and will
inform OMX about it rather than doing a full format reset.
- Various Zynq UltraScale+ specific improvements:
- Video encoders are now able to import dmabuf from upstream.
- Support for HEVC range extension profiles and more AVC profiles.
- We can now request video encoders to generate an IDR using the
force key unit event.
GStreamer Editing Services and NLE
- this section will be filled in in due course
@ -858,7 +900,7 @@ Added
value per tag. The old ::iter_tag_list() function was renamed to
::iter_generic() and still provides access to each value for a tag
- Bus::iter() and Bus::iter_timed() iterators around the corresponding
::pop*() functions
::pop\*() functions
- serde serialization of Value can also handle Buffer now
@ -995,11 +1037,78 @@ Build and Dependencies
- New sctp plugin based on usrsctp (for WebRTC data channels)
Cerbero
Cerbero is a meta build system used to build GStreamer plus dependencies
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
Cerbero has seen a number of improvements:
- Cerbero has been ported to Python 3 and requires Python 3.5 or newer
now
- Source tarballs are now protected by checksums in the recipes to
guard against download errors and malicious takeover of projects or
websites. In addition, downloads are only allowed via secure
transports now and plain HTTP, FTP and git:// transports are not
allowed anymore.
- There is now a new fetch-bootstrap command which downloads sources
required for bootstrapping, with an optional --build-tools-only
argument to match the bootstrap --build-tools-only command.
- The bootstrap, build, package and bundle-source commands gained a
new --offline switch that ensures that only sources from the cache
are used and never downloaded via the network. This is useful in
combination with the fetch and fetch-bootstrap commands that acquire
sources ahead of time before any build steps are executed. This
allows more control over the sources used and when sources are
updated, and is particularly useful for build environments that
don’t have network access.
- bootstrap --assume-yes will automatically say ‘yes’ to any
interactive prompts during the bootstrap stage, such as those from
apt-get or yum.
- bootstrap --system-only will only bootstrap the system without build
tools.
- Manifest support: The build manifest can be used in continuous
integration (CI) systems to fixate the Git revision of certain
projects so that all builds of a pipeline are on the same reference.
This is used in GStreamer’s gitlab CI for example. It can also be
used in order to re-produce a specific build. To set a manifest, you
can set manifest = 'my_manifest.xml' in your configuration file, or
use the --manifest command line option. The command line option will
take precendence over anything specific in the configuration file.
- The new build-deps command can be used to build only the
dependencies of a recipe, without the recipe itself.
- new --list-variants command to list available variants
- variants can now be set on the command line via the -v option as a
comma-separated list. This overrides any variants set in any
configuration files.
- new qt5, intelmsdk and nvidia variants for enabling Qt5 and hardware
codec support. See the Enabling Optional Features with Variants
section in the Cerbero documentation for more details how to enable
and use these variants.
- A new -t / --timestamp command line switch makes commands print
timestamps
Platform-specific changes and improvements
Android
- toolchain: update compiler to clang and NDKr18. NDK r18 removed the
armv5 target and only has Android platforms that target at least
armv7 so the armv5 target is not useful anymore.
- The way that GIO modules are named has changed due to upstream GLib
natively adding support for loading static GIO modules. This means
that any GStreamer application using gnutls for SSL/TLS on the
@ -1010,9 +1119,15 @@ Android
library. Look at this commit for the necessary change in the
examples.
- various build issues on Android have been fixed.
macOS and iOS
- macOS binaries should be fully relocatable now
- various build issues on iOS have been fixed.
- the minimum required iOS version is now 9.0. The difference in
adoption between 8.0 and 9.0 is 0.1% and the bump to 9.0 fixes some
build issues.
- The way that GIO modules are named has changed due to upstream GLib
natively adding support for loading static GIO modules. This means

2
RELEASE
View file

@ -1,4 +1,4 @@
This is GStreamer gst-plugins-good 1.15.1.
This is GStreamer gst-plugins-good 1.15.2.
GStreamer 1.15 is the development branch leading up to the next major
stable version which will be 1.16.

8
configure.ac
View file

@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/pre
AC_INIT([GStreamer Good Plug-ins],[1.15.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AC_INIT([GStreamer Good Plug-ins],[1.15.2],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AG_GST_INIT
@ -46,11 +46,11 @@ AC_DEFINE_UNQUOTED(GST_API_VERSION, "$GST_API_VERSION",
[GStreamer API Version])
AG_GST_LIBTOOL_PREPARE
AS_LIBTOOL(GST, 1501, 0, 1501)
AS_LIBTOOL(GST, 1502, 0, 1502)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.15.1
GSTPB_REQ=1.15.1
GST_REQ=1.15.2
GSTPB_REQ=1.15.2
dnl *** autotools stuff ****

10
gst-plugins-good.doap
View file

@ -32,6 +32,16 @@ the plug-in code, LGPL or LGPL-compatible for the supporting library).
</GitRepository>
</repository>
<release>
<Version>
<revision>1.15.2</revision>
<branch>master</branch>
<name></name>
<created>2019-02-26</created>
<file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.15.2.tar.xz" />
</Version>
</release>
<release>
<Version>
<revision>1.15.1</revision>

2
meson.build
View file

@ -1,5 +1,5 @@
project('gst-plugins-good', 'c',
version : '1.15.1',
version : '1.15.2',
meson_version : '>= 0.47',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])