mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Release 1.7.2
This commit is contained in:
Sebastian Dröge
parent
163a67abab
commit
97e108beba
40 changed files with 999 additions and 130 deletions
885
ChangeLog
885
ChangeLog
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@ -1,9 +1,888 @@
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=== release 1.7.1 ===
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=== release 1.7.2 ===
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2015-12-24 Sebastian Dröge <slomo@coaxion.net>
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2016-02-19 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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releasing 1.7.1
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releasing 1.7.2
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2016-02-19 10:31:05 +0200 Sebastian Dröge <sebastian@centricular.com>
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* po/af.po:
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* po/az.po:
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* po/bg.po:
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* po/ca.po:
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* po/cs.po:
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* po/da.po:
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* po/de.po:
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* po/el.po:
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* po/en_GB.po:
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* po/eo.po:
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* po/es.po:
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* po/eu.po:
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* po/fi.po:
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* po/fr.po:
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* po/gl.po:
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* po/hr.po:
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* po/hu.po:
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* po/id.po:
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* po/it.po:
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* po/ja.po:
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* po/lt.po:
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* po/lv.po:
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* po/nb.po:
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* po/nl.po:
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* po/or.po:
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* po/pl.po:
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* po/pt_BR.po:
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* po/ro.po:
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* po/ru.po:
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* po/sk.po:
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* po/sl.po:
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* po/sq.po:
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* po/sr.po:
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* po/sv.po:
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* po/tr.po:
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* po/uk.po:
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* po/vi.po:
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* po/zh_CN.po:
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po: Update translations
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2016-02-18 14:31:28 +0000 Julien Isorce <j.isorce@samsung.com>
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* pkgconfig/gstreamer-allocators-uninstalled.pc.in:
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* pkgconfig/gstreamer-app-uninstalled.pc.in:
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* pkgconfig/gstreamer-audio-uninstalled.pc.in:
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* pkgconfig/gstreamer-fft-uninstalled.pc.in:
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* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
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* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
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* pkgconfig/gstreamer-riff-uninstalled.pc.in:
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* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
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* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
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* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
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* pkgconfig/gstreamer-tag-uninstalled.pc.in:
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* pkgconfig/gstreamer-video-uninstalled.pc.in:
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uninstalled.pc: add support for non libtool build systems
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Currently the .la path is provided which requires to use libtool as
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mentioned in the GStreamer manual section-helloworld-compilerun.html.
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It is fine as long as the application is built using libtool.
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So currently it is not possible to compile a GStreamer application
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within gst-uninstalled with CMake or other build system different
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than autotools.
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This patch allows to do the following in gst-uninstalled env:
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gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
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gstreamer-video-1.0)
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Previously it required to prepend libtool --mode=link
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https://bugzilla.gnome.org/show_bug.cgi?id=720778
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2016-01-22 18:26:01 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
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* gst/typefind/gsttypefindfunctions.c:
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typefind: strengthen check for valid H.263 picture layer
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Avoids some false positives leading to miss identification:
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* Prevent picture start code emulation for the first 2 bytes read
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* Add check for valid "picture coding type" and "PB-frames mode" combination
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Additionally, change name on confusingly named TR var to what
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it is, the layer's PTYPE.
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https://bugzilla.gnome.org/show_bug.cgi?id=693263
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2015-11-23 15:06:02 +0900 Vineeth T M <vineeth.tm@samsung.com>
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* gst/playback/gstdecodebin2.c:
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decodebin: return incomplete topology if decode chains' cap could not be obtained
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When getting caps of the decode chain, in get_topology, the caps are being
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checked if fixed or not. But get_topology will be called when the decode is
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chain is being exposed and hence it will always be fixed. Hence removing the
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check for fixed caps. Removing gst_pad_get_current_caps for the chain->pad, as
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get_pad_caps will again call the same api.
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And get_topology can return NULL value if currently shutting down the
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pipeline, which on being passed to create message will result in assertion
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error. Check if topology is valid before using it
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https://bugzilla.gnome.org/show_bug.cgi?id=755918
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2016-02-05 10:10:40 +0100 Havard Graff <havard.graff@gmail.com>
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* gst-libs/gst/Makefile.am:
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rtp: build audio library before rtp
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Because audio-enumtypes.h needs to be available for
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gstrtpbaseaudiopayload.c
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https://bugzilla.gnome.org/show_bug.cgi?id=761949
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2016-02-15 21:28:33 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/playback/gstdecodebin2.c:
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decodebin: Fix documentation of the autoplug-query signal
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2016-01-26 13:54:46 +0100 Stian Selnes <stian@pexip.com>
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* gst-libs/gst/video/gstvideoencoder.c:
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* tests/check/libs/videoencoder.c:
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videoencoder: Fix leak when pre_push does not return OK
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https://bugzilla.gnome.org/show_bug.cgi?id=761951
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2016-02-11 19:47:04 +0100 Wim Taymans <wtaymans@redhat.com>
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* gst/audioresample/resample.c:
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resample: avoid overflows
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Avoid overflow in rate calculation. This can cause the resampler to
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start on the wrong phase after a rate change.
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Avoid overflow in cubic fraction calculation. This can cause noise when
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dealing with higher samplerates.
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2016-02-11 18:01:40 +0100 Wim Taymans <wtaymans@redhat.com>
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* gst/audioresample/resample_sse.h:
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resample: fix double interpolation sse code
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We were only reading 2 filter taps and we need to read 4 to do cubic
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interpolation.
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2016-02-10 12:48:15 +0100 Wim Taymans <wtaymans@redhat.com>
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* gst-libs/gst/audio/audio-converter.c:
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audio-converter: make a copy if we can't write in unpack
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If we don't have writable memory, make sure to make a copy of the input
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samples into a temporary (writable) buffer, even if we are dealing with
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a native intermediate format that we don't need to call the unpack
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function for.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=761655
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2016-02-05 19:15:16 -0300 Thiago Santos <thiagoss@osg.samsung.com>
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* tests/check/Makefile.am:
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tests: extend the AM_TESTS_ENVIRONMENT from check.mak
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To get the CK_DEFAULT_TIMEOUT defined for all tests.
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Also replaces a 120 timeout that was set.
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https://bugzilla.gnome.org/show_bug.cgi?id=761472
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2016-02-05 18:03:07 -0300 Thiago Santos <thiagoss@osg.samsung.com>
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* autogen.sh:
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* common:
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Automatic update of common submodule
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From 86e4663 to b64f03f
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2016-01-21 09:43:35 +0100 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>
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* ext/pango/gstbasetextoverlay.c:
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* ext/pango/gstbasetextoverlay.h:
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textoverlay: Expose rendering dimensions as properties.
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In order to detect graphical user input on the
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textoverlay, the resulting rendering properties
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need to be exposed to applications.
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Fixes delayx property declaration.
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https://bugzilla.gnome.org/show_bug.cgi?id=761251
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2016-01-20 15:37:44 +0100 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>
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* ext/pango/gstbasetextoverlay.c:
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textoverlay: Do not limit positioning to video area.
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The current position property is limited to X,Y positions
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in the range of [0, 1]. This patch allows full control
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over the overlay position, including partially outside
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of the video area.
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https://bugzilla.gnome.org/show_bug.cgi?id=761251
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2016-01-28 13:29:39 +0100 Sebastian Dröge <sebastian@centricular.com>
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* gst/audiorate/gstaudiorate.c:
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audiorate: Use gst_audio_format_fill_silence() instead of memset with 0 for generating silence
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For unsigned formats, silence is not all bits 0.
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2016-01-28 13:21:33 +0100 HoonHee Lee <hoonhee.lee@lge.com>
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* gst-libs/gst/audio/gstaudiodecoder.c:
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* gst-libs/gst/video/gstvideodecoder.c:
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audio/videodecoder: Minor cleanup of last commit
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https://bugzilla.gnome.org/show_bug.cgi?id=761218
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2016-01-28 18:06:44 +0900 HoonHee Lee <hoonhee.lee@lge.com>
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* gst-libs/gst/audio/gstaudiodecoder.c:
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* gst-libs/gst/video/gstvideodecoder.c:
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audio/videodecoder: use gst_pad_peer_query_caps to make output caps
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gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
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In that case, use gst_pad_peer_query_caps() with template caps as filter
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to have negotiated output caps properly before forwarding GAP event.
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https://bugzilla.gnome.org/show_bug.cgi?id=761218
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2016-01-26 19:23:04 +0100 Thibault Saunier <tsaunier@gnome.org>
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* gst/encoding/gstencodebin.c:
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encodebin: Allow streamheader update when profile.allow_dynamic_output == FALSE
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Some encoders can update the stream header through time (for example
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vp8 might do that) but it does not strictly changes the output format.
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2016-01-26 14:09:42 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
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* gst-libs/gst/video/video-format.h:
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video-format: fix GstVideoFormatInfo documentation warnings
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Add missing ':' to tile_ws and tile_hs fields documentation to avoid
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bad render of these two fields, mark reserved bytes as private to hide
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field and avoid gtkdoc warning and add parameters description to
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documented macro to avoid gtkdoc warnings.
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https://bugzilla.gnome.org/show_bug.cgi?id=761132
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2016-01-26 16:56:57 +0100 Wim Taymans <wtaymans@redhat.com>
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* gst-libs/gst/audio/audio-converter.c:
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* gst-libs/gst/audio/audio-converter.h:
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* win32/common/libgstaudio.def:
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audio-converter: add reset function
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2016-01-26 16:36:41 +0100 Wim Taymans <wtaymans@redhat.com>
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* gst-libs/gst/audio/audio-converter.c:
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audio-converter: handle NULL input
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Allow NULL as input to mean silence samples.
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2016-01-26 17:16:52 +0100 Wim Taymans <wtaymans@redhat.com>
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* gst-libs/gst/audio/audio-converter.c:
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audio-converter: improve _update_config
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Allow NULL config to keep the existing parameters.
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Fix the docs.
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2016-01-26 17:14:20 +0100 Wim Taymans <wtaymans@redhat.com>
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* gst-libs/gst/audio/audio-converter.c:
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* gst-libs/gst/audio/audio-converter.h:
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audio-converter: audio-converter: make some optimized functions
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Make optimized functions for generic and passthrough conversion.
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2016-01-26 16:34:35 +0100 Wim Taymans <wtaymans@redhat.com>
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* gst-libs/gst/audio/audio-quantize.c:
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* gst-libs/gst/audio/audio-quantize.h:
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audio-quantize: add _reset function
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Add a reset function that clears any history.
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2016-01-25 17:40:23 +0000 Tim-Philipp Müller <tim@centricular.com>
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* configure.ac:
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* m4/Makefile.am:
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* m4/freetype2.m4:
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* tests/examples/Makefile.am:
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build: remove nonsensical check for freetype
|
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The examples need Gtk+, nothing uses freetype directly.
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2016-01-25 16:22:17 +0000 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/elements/libvisual.c:
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tests: libvisual: make run faster
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Reduce resolution, which shouldn't make any difference
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to what's tested here. Makes test finish in less than
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half the time it took before (8s vs. 21s).
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2016-01-25 18:30:30 +0530 Arun Raghavan <git@arunraghavan.net>
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* ext/alsa/gstalsasink.c:
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alsa: Trivial doc update
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alsasink now does more than just raw audio.
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2016-01-21 18:30:40 +0200 Sebastian Dröge <sebastian@centricular.com>
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|
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* gst/playback/gstdecodebin2.c:
|
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decodebin: Correctly expose pads from elements that have directly exposable pads
|
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analyze_new_pad() can return a new decode chain, which might have a new
|
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GstDecodePad in the end. We should use those two for expose_pad() and not the
|
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original ones that were passed to analyze_new_pad().
|
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This fails when having a demuxer element that has raw pads immediately or
|
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if a decoder with raw caps is after an adaptive demuxer.
|
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https://bugzilla.gnome.org/show_bug.cgi?id=760949
|
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|
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2016-01-21 16:08:46 +0100 Wim Taymans <wtaymans@redhat.com>
|
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|
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* gst-libs/gst/audio/audio-converter.c:
|
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audio-converter: ensure correct alignment of samples
|
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Make sure that the data we allocate for our temporary buffers is
|
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properly aligned.
|
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=760938
|
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|
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2016-01-21 10:45:40 +0100 Wim Taymans <wtaymans@redhat.com>
|
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|
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* gst-libs/gst/video/video-color.c:
|
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* gst-libs/gst/video/video-color.h:
|
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video-color: add Adobe RGB primaries and transfer function
|
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|
||||
2016-01-20 10:19:34 +0100 Wim Taymans <wtaymans@redhat.com>
|
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|
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* gst-libs/gst/video/video-info.c:
|
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video-info: enfore RGB matrix for RGB formats
|
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In gst_video_info_to_caps(), make sure we end up with an RGB matrix for
|
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RGB formats and warn when the GstVideoInfo colorimetry is wrong.
|
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In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
|
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for RGB formats and warn about inconsistent caps.
|
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See https://bugzilla.gnome.org/show_bug.cgi?id=759624
|
||||
|
||||
2016-01-20 10:02:20 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/video/video-converter.c:
|
||||
video-converter: ignore matrix for RGB formats
|
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For RGB formats, the matrix in the colorimetry (conversion from YUV to
|
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RGB) is irrelevant and we should ignore it and assume the identity
|
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transform for everything we do.
|
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624
|
||||
|
||||
2016-01-19 23:26:57 +0100 Thibault Saunier <tsaunier@gnome.org>
|
||||
|
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* gst-libs/gst/video/gstvideoencoder.h:
|
||||
videoencoder: Deprecate GST_VIDEO_ENCODER_FLOW_DROPPED
|
||||
It was never actually supported or used
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760666
|
||||
|
||||
2016-01-19 23:22:35 +0100 Thibault Saunier <tsaunier@gnome.org>
|
||||
|
||||
* gst-libs/gst/video/gstvideoencoder.c:
|
||||
Revert "videoencoder: Release video frame when ->handle return ERROR or DROPPED"
|
||||
This reverts commit 63517d0ed348784cce4ab4b295c2c0f1b78baa81.
|
||||
It was wrong ref counting wise and we decided to deprecated DROPPED
|
||||
return value
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760666
|
||||
|
||||
2016-01-18 11:40:36 +0900 Vineeth TM <vineeth.tm@samsung.com>
|
||||
|
||||
* tests/check/elements/audioconvert.c:
|
||||
tests:audioconvert: Fix integer overflow build error
|
||||
value of 32768L << 16 and 1L << 31 is 2147483648
|
||||
but it exceeds the positive range of int which is 2147483647
|
||||
resulting in integer overflow error. Use G_GINT64_CONSTANT instead of L.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760769
|
||||
|
||||
2016-01-19 12:39:22 +0530 Arun Raghavan <git@arunraghavan.net>
|
||||
|
||||
* gst-libs/gst/app/gstappsrc.c:
|
||||
appsrc: Minor documentation cleanup
|
||||
|
||||
2016-01-14 23:14:27 +0000 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* tools/gst-play.c:
|
||||
tools: gst-play: allow setting of flags in serialized foo+bar format
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=751901
|
||||
|
||||
2015-07-02 17:58:00 +0200 Hugues Fruchet <hugues.fruchet@st.com>
|
||||
|
||||
* tools/gst-play.c:
|
||||
tools: gst-play: add command line options for verbose output and playbin flags
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=751901
|
||||
|
||||
2016-01-18 15:51:16 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* win32/common/libgstapp.def:
|
||||
win32: Update exports
|
||||
|
||||
2015-10-15 10:38:16 -0400 Evan Callaway <evan.callaway@ipconfigure.com>
|
||||
|
||||
* gst-libs/gst/app/gstappsink.c:
|
||||
* gst-libs/gst/app/gstappsink.h:
|
||||
Add WAIT_ON_EOS flag to gstappsink.
|
||||
If set, an appsink that receives an EOS will wait until all of its buffers have been processed before continuing.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=756187
|
||||
|
||||
2016-01-16 10:17:50 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/audio/gstaudioencoder.c:
|
||||
audioencoder: Add note to the documentation about various settings being reset before set_format()
|
||||
It's quite unexpected behaviour that various subclass settings are just
|
||||
reset before set_format(). Unfortunately changing this now has the risk
|
||||
of breaking existing code but we should reconsider this for 2.0.
|
||||
|
||||
2016-01-09 04:35:23 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
|
||||
|
||||
* gst/playback/gststreamsynchronizer.c:
|
||||
streamsynchronizer: Ignore flushing streams [..]
|
||||
[..] when resetting group start time. In GES, we are usually connected
|
||||
to the streamsynchronizer on one audio and one video pad.
|
||||
When seeking the timeline, both nlecompositions often output their flush_start
|
||||
before any of them has output its flush_stop.
|
||||
The current code, when receiving the first flush stop was using the
|
||||
running time of the start of the second composition, which could
|
||||
be pretty much anything, and means nothing at that point.
|
||||
This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
|
||||
both when setting flushing and when checking it.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=750013
|
||||
|
||||
2016-01-08 18:53:52 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst/playback/gstplaybin2.c:
|
||||
playbin: Only append non-raw and sysmem pad template caps to the autoplug-query result
|
||||
Otherwise a decoder supporting GL memory will think that all downstream can
|
||||
support GL memory because of seeing its own template caps.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=758212
|
||||
|
||||
2016-01-08 18:37:16 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst/playback/gstplaybin2.c:
|
||||
Revert "playbin: only add the template caps when the result is empty"
|
||||
This reverts commit 023af2d3b192f8ebf1bd4fe75a22a4adaedc1e05.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=758212
|
||||
|
||||
2016-01-15 13:35:22 +0000 Thibault Saunier <tsaunier@gnome.org>
|
||||
|
||||
* gst-libs/gst/video/gstvideoencoder.c:
|
||||
videoencoder: Release video frame when ->handle return ERROR or DROPPED
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760666
|
||||
|
||||
2016-01-15 09:50:29 +0100 Edward Hervey <edward@centricular.com>
|
||||
|
||||
* gst/playback/gstplaysink.c:
|
||||
playsink: Properly mark pending blocked pads
|
||||
When blocking input pads, we also need to properly set the appropriate
|
||||
pending flag.
|
||||
Without this, when switching stream types after initial configuration
|
||||
(like going from Audio+Video to Audio+Video+Sub) playsink would never
|
||||
wait for *all* input streams to be blocked (it would just wait for the
|
||||
new input pad (text in this case) to be blocked).
|
||||
Since the reconfiguration might introduce unlinking/relinking of elements,
|
||||
we need to ensure that *ALL* input streams are blocked.
|
||||
Failure to do so would result in having some input streams pushing data
|
||||
to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
|
||||
(returning GST_FLOW_NOT_LINKED).
|
||||
A later optimization could involve only blocking the input pads that
|
||||
might be involved in reconfiguration. But better be safe than sorry for
|
||||
now :)
|
||||
|
||||
2016-01-06 10:12:43 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
|
||||
|
||||
* tools/gst-device-monitor.c:
|
||||
gst-device-monitor: Use g_printerr instead of g_error
|
||||
g_error is meant to be used for programmer errors (causes an abort),
|
||||
not for expected runtime errors.
|
||||
|
||||
2016-01-13 16:32:25 -0300 Thiago Santos <thiagoss@osg.samsung.com>
|
||||
|
||||
* gst/playback/gstsubtitleoverlay.c:
|
||||
subtitleoverlay: replace gst_caps_can_intersect() with is_subset()
|
||||
Subset check verifies also that all required fields are present
|
||||
and is mostly commonly used when checking if an element accepts
|
||||
a certain caps
|
||||
|
||||
2016-01-12 11:31:50 -0300 Thiago Santos <thiagoss@osg.samsung.com>
|
||||
|
||||
* gst/playback/gstplaybin2.c:
|
||||
playbin: use subset check instead of intersect
|
||||
Elements usually require that all fields on their caps are present
|
||||
on the fixed caps they receive. Using intersection won't verify it,
|
||||
resort to using is_subset() checks.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760477
|
||||
|
||||
2016-01-12 15:56:36 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/audio/audio-channel-mixer.c:
|
||||
audio-channel-mixer: round before truncating
|
||||
Round the result before truncating for int channel mixing.
|
||||
|
||||
2016-01-12 15:27:16 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
audio-converter: Avoid conversion when possible
|
||||
When the input and output formats are the same and in a possible
|
||||
intermediate format, avoid unpack and pack.
|
||||
Never do passthrough channel mixing.
|
||||
Only do dithering and noise shaping in S32 format
|
||||
|
||||
2016-01-12 11:43:20 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/audio/audio-channel-mixer.c:
|
||||
audio-channel-mixer: add more formats
|
||||
Add support for float and int16 mixing
|
||||
Remove in-place processing, this simplifies things as we won't be using it.
|
||||
Don't do clipping for float audio formats
|
||||
|
||||
2016-01-12 11:37:17 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
audio-converter: improve processing loop
|
||||
Process as many samples as we can from the input and return the number
|
||||
of processed samples from the chain. This simplifies some code.
|
||||
Fix the IN_WRITABLE handling, don't overwrite the flags.
|
||||
|
||||
2016-01-11 18:24:48 -0300 Thiago Santos <thiagoss@osg.samsung.com>
|
||||
|
||||
* gst/playback/gstsubtitleoverlay.c:
|
||||
subtitleoverlay: replace accept-caps with caps query
|
||||
Those accept caps are actually checking if downstream supports
|
||||
some particular caps to check if it need to negotiate a different
|
||||
format. Checking only the next element with accept-caps is not enough
|
||||
to guarantee that it is supported.
|
||||
Using a caps query makes it obtain the supported caps for downstream
|
||||
as a whole instead of only the next element.
|
||||
|
||||
2016-01-08 21:27:16 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* win32/common/libgstaudio.def:
|
||||
audio: Update exported symbols list
|
||||
|
||||
2016-01-08 15:05:38 -0300 Thiago Santos <thiagoss@osg.samsung.com>
|
||||
|
||||
* gst/videorate/gstvideorate.c:
|
||||
videorate: replace accept-caps with a caps query
|
||||
accept-caps is only a shallow check, it needs to know
|
||||
whether downstream as a whole accepts the framerate
|
||||
|
||||
2016-01-08 16:08:47 +0000 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* docs/libs/gst-plugins-base-libs-sections.txt:
|
||||
docs: fix up for GstAudioChannelMix rename as well
|
||||
|
||||
2016-01-08 17:34:50 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
* gst-libs/gst/audio/audio-converter.h:
|
||||
* gst/audioconvert/gstaudioconvert.c:
|
||||
audio-converter: small API tweaks
|
||||
Pass flags in _converter_new() so that we can configure ourselves
|
||||
differently depending on some options.
|
||||
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
|
||||
|
||||
2016-01-08 17:28:31 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
* gst-libs/gst/audio/audio-converter.h:
|
||||
audio-converter: prepare API for rate changes
|
||||
Use the update function to update the sample rates along with the config
|
||||
once we implement resampling.
|
||||
|
||||
2016-01-08 17:17:44 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
* gst-libs/gst/audio/audio-converter.h:
|
||||
* gst/audioconvert/gstaudioconvert.c:
|
||||
audio-convert: simplify API
|
||||
Simplify the API, we don't need the consumed and produced output
|
||||
arguments. The caller needs to use the _get_in_frames/get_out_frames API
|
||||
to check how much input is needed and how much output will be produced.
|
||||
|
||||
2016-01-08 17:50:21 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/audio/gstaudioutilsprivate.h:
|
||||
* gst-libs/gst/video/gstvideoutilsprivate.h:
|
||||
audio/video: Use G_GNUC_INTERNAL for internal functions
|
||||
|
||||
2016-01-08 16:22:25 +0100 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst-libs/gst/audio/Makefile.am:
|
||||
* gst-libs/gst/audio/audio-channel-mix.c:
|
||||
* gst-libs/gst/audio/audio-channel-mix.h:
|
||||
* gst-libs/gst/audio/audio-channel-mixer.c:
|
||||
* gst-libs/gst/audio/audio-channel-mixer.h:
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
* gst-libs/gst/audio/audio.h:
|
||||
* win32/common/libgstaudio.def:
|
||||
audio: GstAudioChannelMix -> GstAudioChannelMixer
|
||||
Rename the GstAudioChannelMix object to GstAudioChannelMixer because it
|
||||
looks better and to avoid a conflict with a library in -bad.
|
||||
|
||||
2016-01-07 15:24:25 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst/playback/gstplaybin2.c:
|
||||
playbin: Use the caps query instead of accept-caps to detect if a sink accepts caps
|
||||
accept-caps is only for one element, caps query is recursive. Fixes playback
|
||||
with totem and other situations.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760234
|
||||
|
||||
2016-01-06 15:49:59 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
|
||||
|
||||
* gst-libs/gst/video/gstvideopool.c:
|
||||
videopool: store videoinfo after choosing the biggest buffer size
|
||||
Otherwise, pool could be negotiated with a size which will be different
|
||||
from the one used in allocation which is the GstVideoInfo.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760222
|
||||
|
||||
2016-01-06 12:14:39 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
|
||||
|
||||
* gst/videotestsrc/gstvideotestsrc.c:
|
||||
videotestsrc: add missing break in set_property switch case
|
||||
To avoid future issue when adding new properties.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760204
|
||||
|
||||
2016-01-06 01:04:31 +0000 Koop Mast <kwm@FreeBSD.org>
|
||||
|
||||
* tests/check/elements/audioconvert.c:
|
||||
tests: audioconvert: fix test compilation with clang
|
||||
With clang 3.7.1 on FreeBSD:
|
||||
elements/audioconvert.c:650:12: error: shifting a negative signed value is
|
||||
undefined [-Werror,-Wshift-negative-value]
|
||||
(-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
|
||||
~~~ ^
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=760134
|
||||
|
||||
2016-01-06 01:06:10 +0000 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* tests/check/libs/audiodecoder.c:
|
||||
* tests/check/libs/audioencoder.c:
|
||||
* tests/check/libs/rtp.c:
|
||||
* tests/check/libs/rtpbasepayload.c:
|
||||
tests: fix indentation of various unit tests
|
||||
|
||||
2016-01-05 22:52:34 +0000 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
||||
* docs/libs/gst-plugins-base-libs-sections.txt:
|
||||
docs: add new audio API
|
||||
|
||||
2016-01-03 17:21:18 +0000 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* gst-libs/gst/sdp/gstmikey.h:
|
||||
* gst-libs/gst/video/video-overlay-composition.h:
|
||||
docs: remove dummy function declarations with G_INLINE_FUNCTION for gtk-doc
|
||||
gtk-doc can handle static inline functions just fine these days,
|
||||
there's no need for this stuff any more.
|
||||
|
||||
2016-01-03 10:33:53 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/riff/riff-ids.h:
|
||||
riff: Add missing closing parenthesis to GST_RIFF_WAVE_FORMAT_ANTEX_ADPCME
|
||||
Apparently this #define is unused.
|
||||
|
||||
2016-01-02 23:29:22 +0100 Stefan Sauer <ensonic@users.sf.net>
|
||||
|
||||
* gst-libs/gst/riff/riff-ids.h:
|
||||
riff-ids: remove trailing whitespace
|
||||
|
||||
2016-01-02 23:27:44 +0100 Stefan Sauer <ensonic@users.sf.net>
|
||||
|
||||
* gst-libs/gst/riff/riff-ids.h:
|
||||
riff-ids: fix two swapped ids
|
||||
For these fourcc ids the name and value is swapped. This was causing a warning
|
||||
when registering the avi ids.
|
||||
|
||||
2015-12-31 20:43:28 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/Makefile.am:
|
||||
sdp: Also reorder SUBDIRS to try even harder to build the RTP library first
|
||||
|
||||
2015-12-31 20:41:38 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/Makefile.am:
|
||||
sdp: The SDP library depends on the RTP library now and is not independent anymore
|
||||
Fix up the build dependencies.
|
||||
|
||||
2015-10-07 18:50:18 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
||||
|
||||
* docs/libs/gst-plugins-base-libs-sections.txt:
|
||||
* gst-libs/gst/sdp/Makefile.am:
|
||||
* gst-libs/gst/sdp/gstmikey.c:
|
||||
* gst-libs/gst/sdp/gstmikey.h:
|
||||
* gst-libs/gst/sdp/gstsdpmessage.c:
|
||||
* gst-libs/gst/sdp/gstsdpmessage.h:
|
||||
* tests/check/libs/sdp.c:
|
||||
* win32/common/libgstsdp.def:
|
||||
sdp: add helper fuctions from/to sdp from/to caps
|
||||
<gstsdpmessage.h>
|
||||
GstCaps* gst_sdp_media_get_caps_from_media (const GstSDPMedia *media, gint pt);
|
||||
GstSDPResult gst_sdp_media_set_media_from_caps (const GstCaps* caps, GstSDPMedia *media);
|
||||
gchar * gst_sdp_make_keymgmt (const gchar *uri, const gchar *base64);
|
||||
GstSDPResult gst_sdp_message_attributes_to_caps (GstSDPMessage *msg, GstCaps *caps);
|
||||
GstSDPResult gst_sdp_media_attributes_to_caps (GstSDPMedia *media, GstCaps *caps);
|
||||
<gstmikey.h>
|
||||
GstMIKEYMessage * gst_mikey_message_new_from_caps (GstCaps *caps);
|
||||
gchar * gst_mikey_message_base64_encode (GstMIKEYMessage* msg);
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=745880
|
||||
|
||||
2015-12-29 18:14:54 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst/audioconvert/gstaudioconvert.c:
|
||||
audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
|
||||
In this specific case it wouldn't cause problems as we only ever access the
|
||||
first array element, but let's make explicit what is happening here.
|
||||
CID 1346530 and 1346529
|
||||
|
||||
2015-12-29 17:56:21 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/pbutils/encoding-profile.c:
|
||||
encoding-profile: Check for FALSE'ness directly, not by comparing with FALSE
|
||||
|
||||
2015-12-29 17:54:44 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/pbutils/encoding-profile.c:
|
||||
encoding-profile: Don't use preset_name string after free
|
||||
When we run the loop for another time and do not have a preset name, we would
|
||||
try to print the preset name of a previous iteration that is already freed.
|
||||
Also move some other variables into the block where they are actually used
|
||||
to prevent similar mistakes in the future.
|
||||
CID 1346536
|
||||
|
||||
2015-12-29 14:40:04 +0100 Stefan Sauer <ensonic@users.sf.net>
|
||||
|
||||
* tests/check/elements/audioconvert.c:
|
||||
audioconvert: add a test for gap handling
|
||||
|
||||
2015-12-29 14:23:59 +0100 Stefan Sauer <ensonic@users.sf.net>
|
||||
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
* tests/check/elements/audioconvert.c:
|
||||
audioconvert: fix passthrough operation
|
||||
We did not take the sample size into account. Rearrange the tests to have more
|
||||
conversion test and an extra test case for passthrough operations.
|
||||
Fixes #759890
|
||||
|
||||
2015-12-29 11:29:31 +0000 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* tools/gst-device-monitor.c:
|
||||
tools: gst-device-monitor: print uint properties in both decimal and hex
|
||||
Some values are easier to read and make sense of in hex.
|
||||
https://bugzilla.gnome.org//show_bug.cgi?id=759780
|
||||
|
||||
2015-11-12 14:01:03 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
|
||||
|
||||
* gst-libs/gst/video/video-blend.c:
|
||||
videoblend: special case 1x1 src dims on increment computation
|
||||
Fix crash with 1x1 overlay pixmap
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=757290
|
||||
|
||||
2015-12-28 12:28:26 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst/typefind/gsttypefindfunctions.c:
|
||||
typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
|
||||
We would otherwise read beyond the array bounds and crash every now and then.
|
||||
This was introduced with 5640ba17c8db80976b7718904e4024dcfe9ee1a0.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=759910
|
||||
|
||||
2015-12-27 19:41:43 +0100 Stefan Sauer <ensonic@users.sf.net>
|
||||
|
||||
* tests/check/elements/audioconvert.c:
|
||||
tests: remove commented code from audioconvert test
|
||||
This is just what we have in gst_check_buffer_data().
|
||||
|
||||
2015-12-27 19:25:20 +0100 Stefan Sauer <ensonic@users.sf.net>
|
||||
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
audio-converter: code cleanup
|
||||
Rename samples to num_samples, since we also have samples in chain, but that is
|
||||
the data pointer. Always use gzize for num_samples. Make the log output a bit
|
||||
more homogenous.
|
||||
|
||||
2015-12-26 11:34:47 +0000 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* tools/gst-device-monitor.c:
|
||||
tools: gst-device-monitor: print non-string device properties too
|
||||
|
||||
2015-12-26 09:43:56 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/audio/audio-channel-mix.c:
|
||||
* gst-libs/gst/audio/audio-converter.c:
|
||||
* gst-libs/gst/audio/audio-quantize.c:
|
||||
audio: Fix some documentation warnings
|
||||
Remove/rename function parameters and skip some functions that can't
|
||||
be used by bindings as they are now.
|
||||
|
||||
2015-12-26 09:43:51 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/video/gstvideoaffinetransformationmeta.c:
|
||||
videoaffinetransformmeta: Add (transfer none) annotation for return value
|
||||
|
||||
2015-12-25 11:34:10 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst/playback/gstplaysink.c:
|
||||
playsink: Don't leak audio/video filters due to floating references weirdness
|
||||
The filters' floating references are sinked during set_property() already,
|
||||
which means that GstBin takes a new reference when adding the filter to it.
|
||||
Get rid of the additional reference after adding the filter to the bin.
|
||||
|
||||
2015-12-25 10:36:44 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst/playback/gstplaysink.c:
|
||||
playsink: Allow reuse of audio/video filters by unparenting them from their bins
|
||||
And also recreate the chains if the filter is changing.
|
||||
|
||||
2015-12-25 10:28:02 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst/playback/gstplaysink.c:
|
||||
playsink: Don't leak audio/video filters when using non-raw media
|
||||
|
||||
2015-12-24 15:27:43 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* configure.ac:
|
||||
Back to development
|
||||
|
||||
2015-12-24 13:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* gst-libs/gst/pbutils/Makefile.am:
|
||||
pbutils: Link to libgstbase for bytewriter and adapter
|
||||
|
||||
=== release 1.7.1 ===
|
||||
|
||||
2015-12-24 13:59:15 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* ChangeLog:
|
||||
* NEWS:
|
||||
* RELEASE:
|
||||
* configure.ac:
|
||||
* docs/plugins/inspect/plugin-adder.xml:
|
||||
* docs/plugins/inspect/plugin-alsa.xml:
|
||||
* docs/plugins/inspect/plugin-app.xml:
|
||||
* docs/plugins/inspect/plugin-audioconvert.xml:
|
||||
* docs/plugins/inspect/plugin-audiorate.xml:
|
||||
* docs/plugins/inspect/plugin-audioresample.xml:
|
||||
* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
||||
* docs/plugins/inspect/plugin-cdparanoia.xml:
|
||||
* docs/plugins/inspect/plugin-encoding.xml:
|
||||
* docs/plugins/inspect/plugin-gio.xml:
|
||||
* docs/plugins/inspect/plugin-libvisual.xml:
|
||||
* docs/plugins/inspect/plugin-ogg.xml:
|
||||
* docs/plugins/inspect/plugin-pango.xml:
|
||||
* docs/plugins/inspect/plugin-playback.xml:
|
||||
* docs/plugins/inspect/plugin-subparse.xml:
|
||||
* docs/plugins/inspect/plugin-tcp.xml:
|
||||
* docs/plugins/inspect/plugin-theora.xml:
|
||||
* docs/plugins/inspect/plugin-typefindfunctions.xml:
|
||||
* docs/plugins/inspect/plugin-videoconvert.xml:
|
||||
* docs/plugins/inspect/plugin-videorate.xml:
|
||||
* docs/plugins/inspect/plugin-videoscale.xml:
|
||||
* docs/plugins/inspect/plugin-videotestsrc.xml:
|
||||
* docs/plugins/inspect/plugin-volume.xml:
|
||||
* docs/plugins/inspect/plugin-vorbis.xml:
|
||||
* docs/plugins/inspect/plugin-ximagesink.xml:
|
||||
* docs/plugins/inspect/plugin-xvimagesink.xml:
|
||||
* gst-plugins-base.doap:
|
||||
* win32/common/_stdint.h:
|
||||
* win32/common/audio-enumtypes.c:
|
||||
* win32/common/audio-enumtypes.h:
|
||||
* win32/common/config.h:
|
||||
* win32/common/pbutils-enumtypes.c:
|
||||
* win32/common/pbutils-enumtypes.h:
|
||||
Release 1.7.1
|
||||
|
||||
2015-12-24 13:10:08 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* po/af.po:
|
||||
* po/az.po:
|
||||
* po/bg.po:
|
||||
* po/ca.po:
|
||||
* po/cs.po:
|
||||
* po/da.po:
|
||||
* po/de.po:
|
||||
* po/el.po:
|
||||
* po/en_GB.po:
|
||||
* po/eo.po:
|
||||
* po/es.po:
|
||||
* po/eu.po:
|
||||
* po/fi.po:
|
||||
* po/fr.po:
|
||||
* po/gl.po:
|
||||
* po/hr.po:
|
||||
* po/hu.po:
|
||||
* po/id.po:
|
||||
* po/it.po:
|
||||
* po/ja.po:
|
||||
* po/lt.po:
|
||||
* po/lv.po:
|
||||
* po/nb.po:
|
||||
* po/nl.po:
|
||||
* po/or.po:
|
||||
* po/pl.po:
|
||||
* po/pt_BR.po:
|
||||
* po/ro.po:
|
||||
* po/ru.po:
|
||||
* po/sk.po:
|
||||
* po/sl.po:
|
||||
* po/sq.po:
|
||||
* po/sr.po:
|
||||
* po/sv.po:
|
||||
* po/tr.po:
|
||||
* po/uk.po:
|
||||
* po/vi.po:
|
||||
* po/zh_CN.po:
|
||||
Update .po files
|
||||
|
||||
2015-12-24 12:22:04 +0100 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
|
|
|
|||
2
NEWS
2
NEWS
|
|
@ -1,2 +1,2 @@
|
|||
This is GStreamer 1.7.1
|
||||
This is GStreamer 1.7.2
|
||||
|
||||
|
|
|
|||
106
RELEASE
106
RELEASE
|
|
@ -1,8 +1,7 @@
|
|||
|
||||
Release notes for GStreamer Base Plugins 1.7.1
|
||||
Release notes for GStreamer Base Plugins 1.7.2
|
||||
|
||||
|
||||
The GStreamer team is pleased to announce the first release of the unstable
|
||||
The GStreamer team is pleased to announce the second release of the unstable
|
||||
1.7 release series. The 1.7 release series is adding new features on top of
|
||||
the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release
|
||||
series of the GStreamer multimedia framework. The unstable 1.7 release series
|
||||
|
|
@ -14,7 +13,6 @@ Binaries for Android, iOS, Mac OS X and Windows will be provided separately
|
|||
during the unstable 1.7 release series.
|
||||
|
||||
|
||||
|
||||
This module contains a set of reference plugins, base classes for other
|
||||
plugins, and helper libraries. It also includes essential elements such
|
||||
as audio and video format converters, and higher-level components like playbin,
|
||||
|
|
@ -61,55 +59,42 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
|
|||
|
||||
Bugs fixed in this release
|
||||
|
||||
* 681447 : video overlay composition: fix video blending over transparent frame
|
||||
* 705579 : Playbin prevents plugins requesting a GstContext to work properly
|
||||
* 726117 : typefinding: issue in MPEG-TS detection logic for streams with Null Pids
|
||||
* 726472 : rtpbasepayload: Implement video SDP attributes
|
||||
* 727970 : videorate: remove dead code
|
||||
* 730926 : tags: add GST_TAG_PRIVATE_DATA and expose ID3 private frame ( " PRIV " ) data
|
||||
* 731791 : videometa: add GstVideoAffineTransformationMeta
|
||||
* 738687 : midi: add alsamidisrc, an ALSA MIDI sequencer source
|
||||
* 749596 : rtsp-over-http authentication failure
|
||||
* 751470 : encodebin: Fix special case.
|
||||
* 752651 : decodebin: segfault on setting to NULL
|
||||
* 753852 : gstreamer: base: Fix memory leaks when context parse fails.
|
||||
* 754054 : videorate: remove unnecessary break statement
|
||||
* 754196 : audiodecoder-test: port to using GstHarness
|
||||
* 754223 : audioencoder-tests: port to use GstHarness
|
||||
* 754450 : audiotestsrc: remove frequency and channel number limit
|
||||
* 755260 : decodebin: Fix a race condition accessing the decode_chain field.
|
||||
* 755301 : audioconvert: Integer- > Float conversion creates values slightly smaller than -1.0
|
||||
* 755440 : gst-play: Add keyboard shortcut '0' to seek to beginning
|
||||
* 755482 : videotestsrc: Force alpha downstream if foreground color contains alpha
|
||||
* 756804 : playsink: text_sink dynamic reconnection is not working
|
||||
* 757008 : tests: typefindfunctions: Fix error leak
|
||||
* 757068 : audio{filter,convert,resample}: Clip input buffers to the segment before handling them
|
||||
* 757351 : audioconvert: Latest audioconvert outputs noise
|
||||
* 757480 : Use GST_STIME_FORMAT and GST_STIME_ARGS with GstClockTimeDiff
|
||||
* 757926 : pbutils:encoding-target: Fix string memory leak
|
||||
* 757927 : tests:video: Fix overlay rectangle and buffer leak
|
||||
* 757928 : audio-quantize: Fix dither_buffer memory leak
|
||||
* 758235 : rtspconnection: add support for parsing custom headers
|
||||
* 758744 : allocators: Add logging category for GstFdMemory
|
||||
* 758911 : audiobasesink/src: send latency message on setcaps
|
||||
* 758922 : rtspconnection should optionally make HTTP requests with abs_path instead of absoluteURI
|
||||
* 759126 : appsrc: issues with duration query handling
|
||||
* 759329 : convertframe: Support video crop when convert frame
|
||||
* 759356 : encodebin: Implement an encoding profile serialization format
|
||||
* 742875 : [API] new audiovisualizer base class
|
||||
* 758754 : oggdemux: failing to play an Opus sample file
|
||||
* 745880 : sdp: SDP < - > GstCaps helper functions
|
||||
* 751901 : gst-play: verbose & playbin flags options support
|
||||
* 755918 : decodebin: Refactor code to remove assertion errors
|
||||
* 756187 : appsink: Always blocks on EOS until buffers are consumed since 1.6, should be configurable
|
||||
* 758212 : playbin adds the template caps on autoplug-query
|
||||
* 759729 : audiofxbad: Name collision with new GstAudioChannelMix API from libgstaudio
|
||||
* 759855 : build: pbutils needs to link to libgstbase for bytewriter and adapter
|
||||
* 759890 : audioconvert: creates choppy audio
|
||||
* 760134 : audioconvert test: doesn't build with clang
|
||||
* 760204 : videotestsrc: add missing break in set_property switch case
|
||||
* 760234 : playbin: Assumes recursive accept-caps query, breaks totem
|
||||
* 760408 : #750013 (streamsynchronizer patches) broke some use cases in GES
|
||||
* 760477 : playbin: caps intersection autoplugs too early and stream stops
|
||||
* 760769 : tests:audioconvert: Build error when running make check
|
||||
* 760938 : audioconvert: crash when executing orc unpack function
|
||||
* 760949 : decodebin: Correctly expose pads from elements that have directly exposable pads
|
||||
* 761132 : video-format: fix GstVideoFormatInfo documentation warnings
|
||||
* 761218 : audio/videodecoder: Use gst_pad_peer_query_caps() instead of using gst_pad_get_allowed_caps() to make negotiated output caps before forwarding GAP event
|
||||
* 761251 : textoverlay: Expose text rendering dimensions to applications and remove absolute positioning limit
|
||||
* 761949 : gst-libs/gst/Makefile.am: build audio before rtp
|
||||
* 761951 : videoencoder: Fix leak when pre_push does not return OK
|
||||
* 762085 : gst-base 1.7 update created background buzzing noise with audioconvert
|
||||
* 762239 : matroskademux: Assertions about unmappable memory when demuxing wavpack streams
|
||||
* 693263 : typefinding: MPEG-2 video ES detected as H.263
|
||||
|
||||
==== Download ====
|
||||
|
||||
You can find source releases of gst-plugins-base in the download
|
||||
directory: http://gstreamer.freedesktop.org/src/gst-plugins-base/
|
||||
directory: https://gstreamer.freedesktop.org/src/gst-plugins-base/
|
||||
|
||||
The git repository and details how to clone it can be found at
|
||||
http://cgit.freedesktop.org/gstreamer/gst-plugins-base/
|
||||
|
||||
==== Homepage ====
|
||||
|
||||
The project's website is http://gstreamer.freedesktop.org/
|
||||
The project's website is https://gstreamer.freedesktop.org/
|
||||
|
||||
==== Support and Bugs ====
|
||||
|
||||
|
|
@ -134,50 +119,27 @@ subscribe to the gstreamer-devel list.
|
|||
|
||||
Contributors to this release
|
||||
|
||||
* Andreas Frisch
|
||||
* Antonio Ospite
|
||||
* Arnaud Vrac
|
||||
* Csaba Toth
|
||||
* Arun Raghavan
|
||||
* Aurélien Zanelli
|
||||
* Edward Hervey
|
||||
* Eunhae Choi
|
||||
* Evan Callaway
|
||||
* Guillaume Desmottes
|
||||
* Havard Graff
|
||||
* Jan Schmidt
|
||||
* Joan Pau Beltran
|
||||
* HoonHee Lee
|
||||
* Hugues Fruchet
|
||||
* Hyunjun Ko
|
||||
* Julien Isorce
|
||||
* Kazunori Kobayashi
|
||||
* Koop Mast
|
||||
* Luis de Bethencourt
|
||||
* Lubosz Sarnecki
|
||||
* Mathieu Duponchelle
|
||||
* Matthew Waters
|
||||
* Michael Olbrich
|
||||
* Miguel París Díaz
|
||||
* Nicolas Dufresne
|
||||
* Nirbheek Chauhan
|
||||
* Ognyan Tonchev
|
||||
* Pankaj Darak
|
||||
* Pavel Bludov
|
||||
* Perry Hung
|
||||
* Philippe Normand
|
||||
* Rajat Verma
|
||||
* Ravi Kiran K N
|
||||
* Reynaldo H. Verdejo Pinochet
|
||||
* Sebastian Dröge
|
||||
* Sebastian Rasmussen
|
||||
* Song Bing
|
||||
* Stefan Sauer
|
||||
* Stian Selnes
|
||||
* Thiago Santos
|
||||
* Thibault Saunier
|
||||
* Thomas Bluemel
|
||||
* Tim-Philipp Müller
|
||||
* Vincent Penquerc'h
|
||||
* Vineeth T M
|
||||
* Vineeth TM
|
||||
* Vivia Nikolaidou
|
||||
* William Manley
|
||||
* Wim Taymans
|
||||
* Xavier Claessens
|
||||
* eunhae choi
|
||||
|
||||
|
|
@ -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/prerelease
|
||||
AC_INIT([GStreamer Base Plug-ins],[1.7.1.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
|
||||
AC_INIT([GStreamer Base Plug-ins],[1.7.2],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
|
||||
|
||||
AG_GST_INIT
|
||||
|
||||
|
|
@ -56,10 +56,10 @@ dnl 1.2.5 => 205
|
|||
dnl 1.10.9 (who knows) => 1009
|
||||
dnl
|
||||
dnl sets GST_LT_LDFLAGS
|
||||
AS_LIBTOOL(GST, 701, 0, 701)
|
||||
AS_LIBTOOL(GST, 702, 0, 702)
|
||||
|
||||
dnl *** required versions of GStreamer stuff ***
|
||||
GST_REQ=1.7.1.1
|
||||
GST_REQ=1.7.2
|
||||
|
||||
dnl *** autotools stuff ****
|
||||
|
||||
|
|
|
|||
|
|
@ -1348,6 +1348,16 @@
|
|||
<DEFAULT>0</DEFAULT>
|
||||
</ARG>
|
||||
|
||||
<ARG>
|
||||
<NAME>GstAppSink::wait-on-eos</NAME>
|
||||
<TYPE>gboolean</TYPE>
|
||||
<RANGE></RANGE>
|
||||
<FLAGS>rw</FLAGS>
|
||||
<NICK>Wait on EOS</NICK>
|
||||
<BLURB>Wait for all buffers to be processed after receiving an EOS.</BLURB>
|
||||
<DEFAULT>TRUE</DEFAULT>
|
||||
</ARG>
|
||||
|
||||
<ARG>
|
||||
<NAME>GstVolume::mute</NAME>
|
||||
<TYPE>gboolean</TYPE>
|
||||
|
|
|
|||
|
|
@ -126,6 +126,7 @@ GObject
|
|||
PangoFcFontMap
|
||||
PangoCairoFcFontMap
|
||||
GInterface
|
||||
GDatagramBased
|
||||
GFile
|
||||
GInitable
|
||||
GTypePlugin
|
||||
|
|
|
|||
|
|
@ -1,4 +1,5 @@
|
|||
GSocket GInitable
|
||||
GSocket GInitable GDatagramBased
|
||||
GTypeModule GTypePlugin
|
||||
GstAdder GstChildProxy
|
||||
GstAppSink GstURIHandler
|
||||
|
|
|
|||
|
|
@ -1,3 +1,4 @@
|
|||
GDatagramBased GObject
|
||||
GFile GObject
|
||||
GInitable GObject
|
||||
GstChildProxy GObject
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adds multiple streams</description>
|
||||
<filename>../../gst/adder/.libs/libgstadder.so</filename>
|
||||
<basename>libgstadder.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>ALSA plugin library</description>
|
||||
<filename>../../ext/alsa/.libs/libgstalsa.so</filename>
|
||||
<basename>libgstalsa.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Elements used to communicate with applications</description>
|
||||
<filename>../../gst/app/.libs/libgstapp.so</filename>
|
||||
<basename>libgstapp.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Convert audio to different formats</description>
|
||||
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
|
||||
<basename>libgstaudioconvert.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adjusts audio frames</description>
|
||||
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
|
||||
<basename>libgstaudiorate.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Resamples audio</description>
|
||||
<filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
|
||||
<basename>libgstaudioresample.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Creates audio test signals of given frequency and volume</description>
|
||||
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
|
||||
<basename>libgstaudiotestsrc.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Read audio from CD in paranoid mode</description>
|
||||
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
|
||||
<basename>libgstcdparanoia.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>various encoding-related elements</description>
|
||||
<filename>../../gst/encoding/.libs/libgstencodebin.so</filename>
|
||||
<basename>libgstencodebin.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>GIO elements</description>
|
||||
<filename>../../gst/gio/.libs/libgstgio.so</filename>
|
||||
<basename>libgstgio.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>libvisual visualization plugins</description>
|
||||
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
|
||||
<basename>libgstlibvisual.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
File diff suppressed because one or more lines are too long
|
|
@ -3,7 +3,7 @@
|
|||
<description>Pango-based text rendering and overlay</description>
|
||||
<filename>../../ext/pango/.libs/libgstpango.so</filename>
|
||||
<basename>libgstpango.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>various playback elements</description>
|
||||
<filename>../../gst/playback/.libs/libgstplayback.so</filename>
|
||||
<basename>libgstplayback.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Subtitle parsing</description>
|
||||
<filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
|
||||
<basename>libgstsubparse.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>transfer data over the network via TCP</description>
|
||||
<filename>../../gst/tcp/.libs/libgsttcp.so</filename>
|
||||
<basename>libgsttcp.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Theora plugin library</description>
|
||||
<filename>../../ext/theora/.libs/libgsttheora.so</filename>
|
||||
<basename>libgsttheora.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>default typefind functions</description>
|
||||
<filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename>
|
||||
<basename>libgsttypefindfunctions.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Colorspace conversion</description>
|
||||
<filename>../../gst/videoconvert/.libs/libgstvideoconvert.so</filename>
|
||||
<basename>libgstvideoconvert.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adjusts video frames</description>
|
||||
<filename>../../gst/videorate/.libs/libgstvideorate.so</filename>
|
||||
<basename>libgstvideorate.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Resizes video</description>
|
||||
<filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename>
|
||||
<basename>libgstvideoscale.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Creates a test video stream</description>
|
||||
<filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename>
|
||||
<basename>libgstvideotestsrc.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>plugin for controlling audio volume</description>
|
||||
<filename>../../gst/volume/.libs/libgstvolume.so</filename>
|
||||
<basename>libgstvolume.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Vorbis plugin library</description>
|
||||
<filename>../../ext/vorbis/.libs/libgstvorbis.so</filename>
|
||||
<basename>libgstvorbis.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>X11 video output element based on standard Xlib calls</description>
|
||||
<filename>../../sys/ximage/.libs/libgstximagesink.so</filename>
|
||||
<basename>libgstximagesink.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>XFree86 video output plugin using Xv extension</description>
|
||||
<filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename>
|
||||
<basename>libgstxvimagesink.so</basename>
|
||||
<version>1.7.1</version>
|
||||
<version>1.7.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -34,6 +34,16 @@ A wide range of video and audio decoders, encoders, and filters are included.
|
|||
</GitRepository>
|
||||
</repository>
|
||||
|
||||
<release>
|
||||
<Version>
|
||||
<revision>1.7.2</revision>
|
||||
<branch>master</branch>
|
||||
<name></name>
|
||||
<created>2016-02-19</created>
|
||||
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.7.2.tar.xz" />
|
||||
</Version>
|
||||
</release>
|
||||
|
||||
<release>
|
||||
<Version>
|
||||
<revision>1.7.1</revision>
|
||||
|
|
|
|||
|
|
@ -1,8 +1,8 @@
|
|||
#ifndef _GST_PLUGINS_BASE__STDINT_H
|
||||
#define _GST_PLUGINS_BASE__STDINT_H 1
|
||||
#ifndef _GENERATED_STDINT_H
|
||||
#define _GENERATED_STDINT_H "gst-plugins-base 1.7.1"
|
||||
/* generated using gnu compiler gcc-5 (Debian 5.3.1-4) 5.3.1 20151219 */
|
||||
#define _GENERATED_STDINT_H "gst-plugins-base 1.7.2"
|
||||
/* generated using gnu compiler gcc-6 (Debian 6-20160205-1) 6.0.0 20160205 (experimental) [trunk revision 233183] */
|
||||
#define _STDINT_HAVE_STDINT_H 1
|
||||
#include <stdint.h>
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -6,7 +6,7 @@
|
|||
#include "audio.h"
|
||||
#include "audio-format.h"
|
||||
#include "audio-channels.h"
|
||||
#include "audio-channel-mix.h"
|
||||
#include "audio-channel-mixer.h"
|
||||
#include "audio-converter.h"
|
||||
#include "audio-info.h"
|
||||
#include "audio-quantize.h"
|
||||
|
|
@ -196,25 +196,26 @@ gst_audio_channel_position_get_type (void)
|
|||
return g_define_type_id__volatile;
|
||||
}
|
||||
|
||||
/* enumerations from "audio-channel-mix.h" */
|
||||
/* enumerations from "audio-channel-mixer.h" */
|
||||
GType
|
||||
gst_audio_channel_mix_flags_get_type (void)
|
||||
gst_audio_channel_mixer_flags_get_type (void)
|
||||
{
|
||||
static volatile gsize g_define_type_id__volatile = 0;
|
||||
if (g_once_init_enter (&g_define_type_id__volatile)) {
|
||||
static const GFlagsValue values[] = {
|
||||
{GST_AUDIO_CHANNEL_MIX_FLAGS_NONE, "GST_AUDIO_CHANNEL_MIX_FLAGS_NONE",
|
||||
{GST_AUDIO_CHANNEL_MIXER_FLAGS_NONE, "GST_AUDIO_CHANNEL_MIXER_FLAGS_NONE",
|
||||
"none"},
|
||||
{GST_AUDIO_CHANNEL_MIX_FLAGS_NON_INTERLEAVED,
|
||||
"GST_AUDIO_CHANNEL_MIX_FLAGS_NON_INTERLEAVED", "non-interleaved"},
|
||||
{GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_IN,
|
||||
"GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_IN", "unpositioned-in"},
|
||||
{GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_OUT,
|
||||
"GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_OUT", "unpositioned-out"},
|
||||
{GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED,
|
||||
"GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED", "non-interleaved"},
|
||||
{GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN,
|
||||
"GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN", "unpositioned-in"},
|
||||
{GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT,
|
||||
"GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT",
|
||||
"unpositioned-out"},
|
||||
{0, NULL, NULL}
|
||||
};
|
||||
GType g_define_type_id =
|
||||
g_flags_register_static ("GstAudioChannelMixFlags", values);
|
||||
g_flags_register_static ("GstAudioChannelMixerFlags", values);
|
||||
g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
|
||||
}
|
||||
return g_define_type_id__volatile;
|
||||
|
|
@ -228,8 +229,10 @@ gst_audio_converter_flags_get_type (void)
|
|||
if (g_once_init_enter (&g_define_type_id__volatile)) {
|
||||
static const GFlagsValue values[] = {
|
||||
{GST_AUDIO_CONVERTER_FLAG_NONE, "GST_AUDIO_CONVERTER_FLAG_NONE", "none"},
|
||||
{GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE,
|
||||
"GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE", "source-writable"},
|
||||
{GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE,
|
||||
"GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE", "in-writable"},
|
||||
{GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE,
|
||||
"GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE", "variable-rate"},
|
||||
{0, NULL, NULL}
|
||||
};
|
||||
GType g_define_type_id =
|
||||
|
|
|
|||
|
|
@ -20,9 +20,9 @@ GType gst_audio_pack_flags_get_type (void);
|
|||
GType gst_audio_channel_position_get_type (void);
|
||||
#define GST_TYPE_AUDIO_CHANNEL_POSITION (gst_audio_channel_position_get_type())
|
||||
|
||||
/* enumerations from "audio-channel-mix.h" */
|
||||
GType gst_audio_channel_mix_flags_get_type (void);
|
||||
#define GST_TYPE_AUDIO_CHANNEL_MIX_FLAGS (gst_audio_channel_mix_flags_get_type())
|
||||
/* enumerations from "audio-channel-mixer.h" */
|
||||
GType gst_audio_channel_mixer_flags_get_type (void);
|
||||
#define GST_TYPE_AUDIO_CHANNEL_MIXER_FLAGS (gst_audio_channel_mixer_flags_get_type())
|
||||
|
||||
/* enumerations from "audio-converter.h" */
|
||||
GType gst_audio_converter_flags_get_type (void);
|
||||
|
|
|
|||
|
|
@ -90,7 +90,7 @@
|
|||
#define GST_PACKAGE_ORIGIN "Unknown package origin"
|
||||
|
||||
/* GStreamer package release date/time for plugins as YYYY-MM-DD */
|
||||
#define GST_PACKAGE_RELEASE_DATETIME "2015-12-24"
|
||||
#define GST_PACKAGE_RELEASE_DATETIME "2016-02-19"
|
||||
|
||||
/* Define if static plugins should be built */
|
||||
#undef GST_PLUGIN_BUILD_STATIC
|
||||
|
|
@ -323,8 +323,7 @@
|
|||
/* gettext locale dir */
|
||||
#define LOCALEDIR PREFIX "\\share\\locale"
|
||||
|
||||
/* Define to the sub-directory in which libtool stores uninstalled libraries.
|
||||
*/
|
||||
/* Define to the sub-directory where libtool stores uninstalled libraries. */
|
||||
#undef LT_OBJDIR
|
||||
|
||||
/* Name of package */
|
||||
|
|
@ -337,7 +336,7 @@
|
|||
#define PACKAGE_NAME "GStreamer Base Plug-ins"
|
||||
|
||||
/* Define to the full name and version of this package. */
|
||||
#define PACKAGE_STRING "GStreamer Base Plug-ins 1.7.1"
|
||||
#define PACKAGE_STRING "GStreamer Base Plug-ins 1.7.2"
|
||||
|
||||
/* Define to the one symbol short name of this package. */
|
||||
#define PACKAGE_TARNAME "gst-plugins-base"
|
||||
|
|
@ -346,7 +345,7 @@
|
|||
#undef PACKAGE_URL
|
||||
|
||||
/* Define to the version of this package. */
|
||||
#define PACKAGE_VERSION "1.7.1"
|
||||
#define PACKAGE_VERSION "1.7.2"
|
||||
|
||||
/* directory where plugins are located */
|
||||
#ifdef _DEBUG
|
||||
|
|
@ -380,7 +379,7 @@
|
|||
#undef USE_TREMOLO
|
||||
|
||||
/* Version number of package */
|
||||
#define VERSION "1.7.1"
|
||||
#define VERSION "1.7.2"
|
||||
|
||||
/* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most
|
||||
significant byte first (like Motorola and SPARC, unlike Intel). */
|
||||
|
|
|
|||
|
|
@ -205,6 +205,7 @@ gst_video_transfer_function_get_type (void)
|
|||
{GST_VIDEO_TRANSFER_LOG316, "GST_VIDEO_TRANSFER_LOG316", "log316"},
|
||||
{GST_VIDEO_TRANSFER_BT2020_12, "GST_VIDEO_TRANSFER_BT2020_12",
|
||||
"bt2020-12"},
|
||||
{GST_VIDEO_TRANSFER_ADOBERGB, "GST_VIDEO_TRANSFER_ADOBERGB", "adobergb"},
|
||||
{0, NULL, NULL}
|
||||
};
|
||||
GType g_define_type_id =
|
||||
|
|
@ -236,6 +237,8 @@ gst_video_color_primaries_get_type (void)
|
|||
"film"},
|
||||
{GST_VIDEO_COLOR_PRIMARIES_BT2020, "GST_VIDEO_COLOR_PRIMARIES_BT2020",
|
||||
"bt2020"},
|
||||
{GST_VIDEO_COLOR_PRIMARIES_ADOBERGB, "GST_VIDEO_COLOR_PRIMARIES_ADOBERGB",
|
||||
"adobergb"},
|
||||
{0, NULL, NULL}
|
||||
};
|
||||
GType g_define_type_id =
|
||||
|
|
|
|||
Loading…
Reference in a new issue